Request 6067 (accepted)

Overview

Request 6067 (accepted)

New stable branch

Created by zaitor 4 months ago
In state accepted
Package maintainer: zaitor

Submit gstreamer-plugins-ugly-codecs

Submit package home:zait...sentials / gstreamer...y-codecs to package Essentials / gstreamer-plugins-ugly-codecs

gstreamer-plugins-ugly-codecs.changes Changed

gstreamer-plugins-ugly-codecs.spec Changed

gst-plugins-ugly-1.22.7.tar.xz/ext/amrnb Deleted

gst-plugins-ugly-1.22.7.tar.xz/ext/amrnb/GstAmrnbEnc.prs Deleted

gst-plugins-ugly-1.22.7.tar.xz/ext/amrnb/README Deleted

gst-plugins-ugly-1.22.7.tar.xz/ext/amrnb/amrnb.c Deleted

@@ -1,44 +0,0 @@
-/* GStreamer Adaptive Multi-Rate Narrow-Band (AMR-NB) plugin
- * Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include "amrnbdec.h"
-#include "amrnbenc.h"
-
-static gboolean
-plugin_init (GstPlugin * plugin)
-{
-  gboolean ret = FALSE;
-
-  ret |= GST_ELEMENT_REGISTER (amrnbdec, plugin);
-  ret |= GST_ELEMENT_REGISTER (amrnbenc, plugin);
-
-  return ret;
-}
-
-
-GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
-    GST_VERSION_MINOR,
-    amrnb,
-    "Adaptive Multi-Rate Narrow-Band",
-    plugin_init, VERSION, GST_LICENSE_UNKNOWN, GST_PACKAGE_NAME,
-    GST_PACKAGE_ORIGIN);

gst-plugins-ugly-1.22.7.tar.xz/ext/amrnb/amrnbdec.c Deleted

@@ -1,309 +0,0 @@
-/* GStreamer Adaptive Multi-Rate Narrow-Band (AMR-NB) plugin
- * Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-
-/**
- * SECTION:element-amrnbdec
- * @title: amrnbdec
- * @see_also: #GstAmrnbEnc, #GstAmrParse
- *
- * AMR narrowband decoder based on the
- * opencore codec implementation(http://sourceforge.net/projects/opencore-amr).
- *
- * ## Example launch line
- * |
- * gst-launch-1.0 filesrc location=abc.amr ! amrparse ! amrnbdec ! audioconvert ! audioresample ! autoaudiosink
- * |
- *
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include "amrnbdec.h"
-
-static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
-    GST_PAD_SINK,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/AMR, " "rate = (int) 8000, " "channels = (int) 1")
-    );
-
-static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
-    GST_PAD_SRC,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/x-raw, format = (string) " GST_AUDIO_NE (S16) ", "
-        "layout = (string) interleaved, "
-        "rate = (int) 8000," "channels = (int) 1")
-    );
-
-GST_DEBUG_CATEGORY_STATIC (gst_amrnbdec_debug);
-#define GST_CAT_DEFAULT gst_amrnbdec_debug
-
-static const gint block_size_if116 = { 12, 13, 15, 17, 19, 20, 26, 31, 5,
-  0, 0, 0, 0, 0, 0, 0
-};
-
-static const gint block_size_if216 = { 12, 13, 15, 17, 18, 20, 25, 30, 5,
-  0, 0, 0, 0, 0, 0, 0
-};
-
-static GType
-gst_amrnb_variant_get_type (void)
-{
-  static GType gst_amrnb_variant_type = 0;
-  static const GEnumValue gst_amrnb_variant = {
-    {GST_AMRNB_VARIANT_IF1, "IF1", "IF1"},
-    {GST_AMRNB_VARIANT_IF2, "IF2", "IF2"},
-    {0, NULL, NULL},
-  };
-  if (!gst_amrnb_variant_type) {
-    gst_amrnb_variant_type =
-        g_enum_register_static ("GstAmrnbVariant", gst_amrnb_variant);
-  }
-  return gst_amrnb_variant_type;
-}
-
-#define GST_AMRNB_VARIANT_TYPE (gst_amrnb_variant_get_type())
-
-#define VARIANT_DEFAULT GST_AMRNB_VARIANT_IF1
-enum
-{
-  PROP_0,
-  PROP_VARIANT
-};
-
-static void gst_amrnbdec_set_property (GObject * object, guint prop_id,
-    const GValue * value, GParamSpec * pspec);
-static void gst_amrnbdec_get_property (GObject * object, guint prop_id,
-    GValue * value, GParamSpec * pspec);
-
-static gboolean gst_amrnbdec_start (GstAudioDecoder * dec);
-static gboolean gst_amrnbdec_stop (GstAudioDecoder * dec);
-static gboolean gst_amrnbdec_set_format (GstAudioDecoder * dec, GstCaps * caps);
-static GstFlowReturn gst_amrnbdec_parse (GstAudioDecoder * dec,
-    GstAdapter * adapter, gint * offset, gint * length);
-static GstFlowReturn gst_amrnbdec_handle_frame (GstAudioDecoder * dec,
-    GstBuffer * buffer);
-
-#define gst_amrnbdec_parent_class parent_class
-G_DEFINE_TYPE (GstAmrnbDec, gst_amrnbdec, GST_TYPE_AUDIO_DECODER);
-GST_ELEMENT_REGISTER_DEFINE (amrnbdec, "amrnbdec", GST_RANK_PRIMARY,
-    GST_TYPE_AMRNBDEC);
-
-static void
-gst_amrnbdec_class_init (GstAmrnbDecClass * klass)
-{
-  GObjectClass *object_class = G_OBJECT_CLASS (klass);
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-  GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
-
-  object_class->set_property = gst_amrnbdec_set_property;
-  object_class->get_property = gst_amrnbdec_get_property;
-
-  gst_element_class_add_static_pad_template (element_class, &sink_template);
-  gst_element_class_add_static_pad_template (element_class, &src_template);
-
-  gst_element_class_set_static_metadata (element_class, "AMR-NB audio decoder",
-      "Codec/Decoder/Audio",
-      "Adaptive Multi-Rate Narrow-Band audio decoder",
-      "GStreamer maintainers <gstreamer-devel@lists.freedesktop.org>");
-
-  base_class->start = GST_DEBUG_FUNCPTR (gst_amrnbdec_start);
-  base_class->stop = GST_DEBUG_FUNCPTR (gst_amrnbdec_stop);
-  base_class->set_format = GST_DEBUG_FUNCPTR (gst_amrnbdec_set_format);
-  base_class->parse = GST_DEBUG_FUNCPTR (gst_amrnbdec_parse);
-  base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_amrnbdec_handle_frame);
-
-  g_object_class_install_property (object_class, PROP_VARIANT,
-      g_param_spec_enum ("variant", "Variant",
-          "The decoder variant", GST_AMRNB_VARIANT_TYPE,
-          VARIANT_DEFAULT,
-          G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
-
-  GST_DEBUG_CATEGORY_INIT (gst_amrnbdec_debug, "amrnbdec", 0,
-      "AMR-NB audio decoder");
-
-  gst_type_mark_as_plugin_api (GST_AMRNB_VARIANT_TYPE, 0);
-}
-
-static void
-gst_amrnbdec_init (GstAmrnbDec * amrnbdec)
-{
-  gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (amrnbdec), TRUE);
-  gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
-      (amrnbdec), TRUE);
-  GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (amrnbdec));
-}
-
-static gboolean
-gst_amrnbdec_start (GstAudioDecoder * dec)
-{
-  GstAmrnbDec *amrnbdec = GST_AMRNBDEC (dec);
-
-  GST_DEBUG_OBJECT (dec, "start");
-  if (!(amrnbdec->handle = Decoder_Interface_init ()))
-    return FALSE;
-
-  amrnbdec->rate = 0;
-  amrnbdec->channels = 0;
-
-  return TRUE;
-}
-
-static gboolean
-gst_amrnbdec_stop (GstAudioDecoder * dec)
-{
-  GstAmrnbDec *amrnbdec = GST_AMRNBDEC (dec);
-
-  GST_DEBUG_OBJECT (dec, "stop");
-  Decoder_Interface_exit (amrnbdec->handle);
-
-  return TRUE;
-}
-
-static void
-gst_amrnbdec_set_property (GObject * object, guint prop_id,
-    const GValue * value, GParamSpec * pspec)
-{
-  GstAmrnbDec *self = GST_AMRNBDEC (object);
-
-  switch (prop_id) {
-    case PROP_VARIANT:
-      self->variant = g_value_get_enum (value);
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-  return;
-}
-
-static void
-gst_amrnbdec_get_property (GObject * object, guint prop_id,
-    GValue * value, GParamSpec * pspec)
-{
-  GstAmrnbDec *self = GST_AMRNBDEC (object);
-
-  switch (prop_id) {
-    case PROP_VARIANT:
-      g_value_set_enum (value, self->variant);
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-  return;
-}
-
-static gboolean
-gst_amrnbdec_set_format (GstAudioDecoder * dec, GstCaps * caps)
-{
-  GstStructure *structure;
-  GstAmrnbDec *amrnbdec;
-  GstAudioInfo info;
-
-  amrnbdec = GST_AMRNBDEC (dec);
-
-  structure = gst_caps_get_structure (caps, 0);
-
-  /* get channel count */
-  gst_structure_get_int (structure, "channels", &amrnbdec->channels);
-  gst_structure_get_int (structure, "rate", &amrnbdec->rate);
-
-  /* create reverse caps */
-  gst_audio_info_init (&info);
-  gst_audio_info_set_format (&info,
-      GST_AUDIO_FORMAT_S16, amrnbdec->rate, amrnbdec->channels, NULL);
-
-  return gst_audio_decoder_set_output_format (dec, &info);
-}
-
-static GstFlowReturn
-gst_amrnbdec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
-    gint * offset, gint * length)
-{
-  GstAmrnbDec *amrnbdec = GST_AMRNBDEC (dec);
-  guint8 head1;
-  guint size;
-  gboolean sync, eos;
-  gint block, mode;
-
-  size = gst_adapter_available (adapter);
-  if (size < 1)
-    return GST_FLOW_ERROR;
-
-  gst_audio_decoder_get_parse_state (dec, &sync, &eos);
-
-  /* need to peek data to get the size */
-  gst_adapter_copy (adapter, head, 0, 1);
-
-  /* get size */
-  switch (amrnbdec->variant) {
-    case GST_AMRNB_VARIANT_IF1:
-      mode = (head0 >> 3) & 0x0F;
-      block = block_size_if1mode + 1;
-      break;
-    case GST_AMRNB_VARIANT_IF2:
-      mode = head0 & 0x0F;
-      block = block_size_if2mode + 1;
-      break;
-    default:
-      g_assert_not_reached ();
-      return GST_FLOW_ERROR;
-      break;
-  }
-
-  GST_DEBUG_OBJECT (amrnbdec, "mode %d, block %d", mode, block);
-
-  if (block > size)
-    return GST_FLOW_EOS;
-
-  *offset = 0;
-  *length = block;
-
-  return GST_FLOW_OK;
-}
-
-static GstFlowReturn
-gst_amrnbdec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
-{
-  GstAmrnbDec *amrnbdec;
-  GstMapInfo inmap, outmap;
-  GstBuffer *out;
-
-  amrnbdec = GST_AMRNBDEC (dec);
-
-  /* no fancy flushing */
-  if (!buffer || !gst_buffer_get_size (buffer))
-    return GST_FLOW_OK;
-
-  gst_buffer_map (buffer, &inmap, GST_MAP_READ);
-
-  /* get output */
-  out = gst_buffer_new_and_alloc (160 * 2);
-  /* decode */
-  gst_buffer_map (out, &outmap, GST_MAP_WRITE);
-  Decoder_Interface_Decode (amrnbdec->handle, inmap.data,
-      (gint16 *) outmap.data, 0);
-  gst_buffer_unmap (out, &outmap);
-
-  gst_buffer_unmap (buffer, &inmap);
-
-  return gst_audio_decoder_finish_frame (dec, out, 1);
-}

gst-plugins-ugly-1.22.7.tar.xz/ext/amrnb/amrnbdec.h Deleted

@@ -1,71 +0,0 @@
-/* GStreamer Adaptive Multi-Rate Narrow-Band (AMR-NB) plugin
- * Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-
-#ifndef __GST_AMRNBDEC_H__
-#define __GST_AMRNBDEC_H__
-
-#include <gst/gst.h>
-#include <gst/audio/gstaudiodecoder.h>
-
-#include <opencore-amrnb/interf_dec.h>
-
-G_BEGIN_DECLS
-
-#define GST_TYPE_AMRNBDEC \
-  (gst_amrnbdec_get_type())
-#define GST_AMRNBDEC(obj) \
-  (G_TYPE_CHECK_INSTANCE_CAST((obj), GST_TYPE_AMRNBDEC, GstAmrnbDec))
-#define GST_AMRNBDEC_CLASS(klass) \
-  (G_TYPE_CHECK_CLASS_CAST((klass), GST_TYPE_AMRNBDEC, GstAmrnbDecClass))
-#define GST_IS_AMRNBDEC(obj) \
-  (G_TYPE_CHECK_INSTANCE_TYPE((obj), GST_TYPE_AMRNBDEC))
-#define GST_IS_AMRNBDEC_CLASS(klass) \
-  (G_TYPE_CHECK_CLASS_TYPE((klass), GST_TYPE_AMRNBDEC))
-
-typedef struct _GstAmrnbDec GstAmrnbDec;
-typedef struct _GstAmrnbDecClass GstAmrnbDecClass;
-
-typedef enum
-{
-  GST_AMRNB_VARIANT_IF1,
-  GST_AMRNB_VARIANT_IF2
-} GstAmrnbVariant;
-
-struct _GstAmrnbDec {
-  GstAudioDecoder element;
-
-  GstAmrnbVariant variant;
-
-  /* library handle */
-  void *handle;
-
-  /* output settings */
-  gint channels, rate;
-};
-
-struct _GstAmrnbDecClass {
-  GstAudioDecoderClass parent_class;
-};
-
-GType gst_amrnbdec_get_type (void);
-GST_ELEMENT_REGISTER_DECLARE (amrnbdec);
-
-G_END_DECLS
-
-#endif /* __GST_AMRNBDEC_H__ */

gst-plugins-ugly-1.22.7.tar.xz/ext/amrnb/amrnbenc.c Deleted

@@ -1,295 +0,0 @@
-/* GStreamer Adaptive Multi-Rate Narrow-Band (AMR-NB) plugin
- * Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-
-/**
- * SECTION:element-amrnbenc
- * @title: amrnbenc
- * @see_also: #GstAmrnbDec, #GstAmrnbParse
- *
- * AMR narrowband encoder based on the
- * opencore codec implementation(http://sourceforge.net/projects/opencore-amr).
- *
- * ## Example launch line
- * |
- * gst-launch-1.0 filesrc location=abc.wav ! wavparse ! audioconvert ! audioresample ! amrnbenc ! filesink location=abc.amr
- * |
- * Please note that the above stream misses the header, that is needed to play
- * the stream.
- *
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include "amrnbenc.h"
-
-static GType
-gst_amrnbenc_bandmode_get_type (void)
-{
-  static GType gst_amrnbenc_bandmode_type = 0;
-  static const GEnumValue gst_amrnbenc_bandmode = {
-    {MR475, "MR475", "MR475"},
-    {MR515, "MR515", "MR515"},
-    {MR59, "MR59", "MR59"},
-    {MR67, "MR67", "MR67"},
-    {MR74, "MR74", "MR74"},
-    {MR795, "MR795", "MR795"},
-    {MR102, "MR102", "MR102"},
-    {MR122, "MR122", "MR122"},
-    {MRDTX, "MRDTX", "MRDTX"},
-    {0, NULL, NULL},
-  };
-  if (!gst_amrnbenc_bandmode_type) {
-    gst_amrnbenc_bandmode_type =
-        g_enum_register_static ("GstAmrnbEncBandMode", gst_amrnbenc_bandmode);
-  }
-  return gst_amrnbenc_bandmode_type;
-}
-
-#define GST_AMRNBENC_BANDMODE_TYPE (gst_amrnbenc_bandmode_get_type())
-
-#define BANDMODE_DEFAULT MR122
-enum
-{
-  PROP_0,
-  PROP_BANDMODE
-};
-
-static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
-    GST_PAD_SINK,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/x-raw, format = (string) " GST_AUDIO_NE (S16) ", "
-        "layout = (string) interleaved, "
-        "rate = (int) 8000," "channels = (int) 1")
-    );
-
-static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
-    GST_PAD_SRC,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/AMR, " "rate = (int) 8000, " "channels = (int) 1")
-    );
-
-GST_DEBUG_CATEGORY_STATIC (gst_amrnbenc_debug);
-#define GST_CAT_DEFAULT gst_amrnbenc_debug
-
-static gboolean gst_amrnbenc_start (GstAudioEncoder * enc);
-static gboolean gst_amrnbenc_stop (GstAudioEncoder * enc);
-static gboolean gst_amrnbenc_set_format (GstAudioEncoder * enc,
-    GstAudioInfo * info);
-static GstFlowReturn gst_amrnbenc_handle_frame (GstAudioEncoder * enc,
-    GstBuffer * in_buf);
-
-#define gst_amrnbenc_parent_class parent_class
-G_DEFINE_TYPE (GstAmrnbEnc, gst_amrnbenc, GST_TYPE_AUDIO_ENCODER);
-GST_ELEMENT_REGISTER_DEFINE (amrnbenc, "amrnbenc", GST_RANK_SECONDARY,
-    GST_TYPE_AMRNBENC);
-
-static void
-gst_amrnbenc_set_property (GObject * object, guint prop_id,
-    const GValue * value, GParamSpec * pspec)
-{
-  GstAmrnbEnc *self = GST_AMRNBENC (object);
-
-  switch (prop_id) {
-    case PROP_BANDMODE:
-      self->bandmode = g_value_get_enum (value);
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-  return;
-}
-
-static void
-gst_amrnbenc_get_property (GObject * object, guint prop_id,
-    GValue * value, GParamSpec * pspec)
-{
-  GstAmrnbEnc *self = GST_AMRNBENC (object);
-
-  switch (prop_id) {
-    case PROP_BANDMODE:
-      g_value_set_enum (value, self->bandmode);
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-  return;
-}
-
-static void
-gst_amrnbenc_class_init (GstAmrnbEncClass * klass)
-{
-  GObjectClass *object_class = G_OBJECT_CLASS (klass);
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-  GstAudioEncoderClass *base_class = GST_AUDIO_ENCODER_CLASS (klass);
-
-  object_class->set_property = gst_amrnbenc_set_property;
-  object_class->get_property = gst_amrnbenc_get_property;
-
-  base_class->start = GST_DEBUG_FUNCPTR (gst_amrnbenc_start);
-  base_class->stop = GST_DEBUG_FUNCPTR (gst_amrnbenc_stop);
-  base_class->set_format = GST_DEBUG_FUNCPTR (gst_amrnbenc_set_format);
-  base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_amrnbenc_handle_frame);
-
-  g_object_class_install_property (object_class, PROP_BANDMODE,
-      g_param_spec_enum ("band-mode", "Band Mode",
-          "Encoding Band Mode (Kbps)", GST_AMRNBENC_BANDMODE_TYPE,
-          BANDMODE_DEFAULT,
-          G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
-
-  gst_element_class_add_static_pad_template (element_class, &sink_template);
-  gst_element_class_add_static_pad_template (element_class, &src_template);
-
-  gst_element_class_set_static_metadata (element_class, "AMR-NB audio encoder",
-      "Codec/Encoder/Audio",
-      "Adaptive Multi-Rate Narrow-Band audio encoder",
-      "Wim Taymans <wim.taymans@gmail.com>");
-
-  GST_DEBUG_CATEGORY_INIT (gst_amrnbenc_debug, "amrnbenc", 0,
-      "AMR-NB audio encoder");
-
-  gst_type_mark_as_plugin_api (GST_AMRNBENC_BANDMODE_TYPE, 0);
-}
-
-static void
-gst_amrnbenc_init (GstAmrnbEnc * amrnbenc)
-{
-  GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (amrnbenc));
-}
-
-static gboolean
-gst_amrnbenc_start (GstAudioEncoder * enc)
-{
-  GstAmrnbEnc *amrnbenc = GST_AMRNBENC (enc);
-
-  GST_DEBUG_OBJECT (amrnbenc, "start");
-
-  if (!(amrnbenc->handle = Encoder_Interface_init (0)))
-    return FALSE;
-
-  return TRUE;
-}
-
-static gboolean
-gst_amrnbenc_stop (GstAudioEncoder * enc)
-{
-  GstAmrnbEnc *amrnbenc = GST_AMRNBENC (enc);
-
-  GST_DEBUG_OBJECT (amrnbenc, "stop");
-
-  Encoder_Interface_exit (amrnbenc->handle);
-
-  return TRUE;
-}
-
-static gboolean
-gst_amrnbenc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
-{
-  GstAmrnbEnc *amrnbenc;
-  GstCaps *copy;
-
-  amrnbenc = GST_AMRNBENC (enc);
-
-  /* parameters already parsed for us */
-  amrnbenc->rate = GST_AUDIO_INFO_RATE (info);
-  amrnbenc->channels = GST_AUDIO_INFO_CHANNELS (info);
-
-  /* we do not really accept other input, but anyway ... */
-  /* this is not wrong but will sound bad */
-  if (amrnbenc->channels != 1) {
-    g_warning ("amrnbdec is only optimized for mono channels");
-  }
-  if (amrnbenc->rate != 8000) {
-    g_warning ("amrnbdec is only optimized for 8000 Hz samplerate");
-  }
-
-  /* create reverse caps */
-  copy = gst_caps_new_simple ("audio/AMR",
-      "channels", G_TYPE_INT, amrnbenc->channels,
-      "rate", G_TYPE_INT, amrnbenc->rate, NULL);
-
-  gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (amrnbenc), copy);
-  gst_caps_unref (copy);
-
-  /* report needs to base class: hand one frame at a time */
-  gst_audio_encoder_set_frame_samples_min (enc, 160);
-  gst_audio_encoder_set_frame_samples_max (enc, 160);
-  gst_audio_encoder_set_frame_max (enc, 1);
-
-  return TRUE;
-}
-
-static GstFlowReturn
-gst_amrnbenc_handle_frame (GstAudioEncoder * enc, GstBuffer * buffer)
-{
-  GstAmrnbEnc *amrnbenc;
-  GstFlowReturn ret;
-  GstBuffer *out;
-  GstMapInfo in_map, out_map;
-  gsize out_size;
-
-  amrnbenc = GST_AMRNBENC (enc);
-
-  g_return_val_if_fail (amrnbenc->handle, GST_FLOW_FLUSHING);
-
-  /* we don't deal with squeezing remnants, so simply discard those */
-  if (G_UNLIKELY (buffer == NULL)) {
-    GST_DEBUG_OBJECT (amrnbenc, "no data");
-    return GST_FLOW_OK;
-  }
-
-  gst_buffer_map (buffer, &in_map, GST_MAP_READ);
-
-  if (G_UNLIKELY (in_map.size < 320)) {
-    gst_buffer_unmap (buffer, &in_map);
-    GST_DEBUG_OBJECT (amrnbenc, "discarding trailing data of %" G_GSIZE_FORMAT
-        " bytes", in_map.size);
-    return gst_audio_encoder_finish_frame (enc, NULL, -1);
-  }
-
-  /* get output, max size is 32 */
-  out = gst_buffer_new_and_alloc (32);
-  /* AMR encoder actually writes into the source data buffers it gets */
-  /* should be able to handle that with what we are given */
-
-  gst_buffer_map (out, &out_map, GST_MAP_WRITE);
-  /* encode */
-  out_size =
-      Encoder_Interface_Encode (amrnbenc->handle, amrnbenc->bandmode,
-      (short *) in_map.data, out_map.data, 0);
-  gst_buffer_unmap (out, &out_map);
-  gst_buffer_resize (out, 0, out_size);
-  gst_buffer_unmap (buffer, &in_map);
-
-  GST_LOG_OBJECT (amrnbenc, "output data size %" G_GSIZE_FORMAT, out_size);
-
-  if (out_size) {
-    ret = gst_audio_encoder_finish_frame (enc, out, 160);
-  } else {
-    /* should not happen (without dtx or so at least) */
-    GST_WARNING_OBJECT (amrnbenc, "no encoded data; discarding input");
-    gst_buffer_unref (out);
-    ret = gst_audio_encoder_finish_frame (enc, NULL, -1);
-  }
-
-  return ret;
-}

gst-plugins-ugly-1.22.7.tar.xz/ext/amrnb/amrnbenc.h Deleted

@@ -1,67 +0,0 @@
-/* GStreamer Adaptive Multi-Rate Narrow-Band (AMR-NB) plugin
- * Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-
-#ifndef __GST_AMRNBENC_H__
-#define __GST_AMRNBENC_H__
-
-#include <gst/gst.h>
-#include <gst/audio/gstaudioencoder.h>
-
-#include <opencore-amrnb/interf_enc.h>
-
-G_BEGIN_DECLS
-
-#define GST_TYPE_AMRNBENC \
-  (gst_amrnbenc_get_type())
-#define GST_AMRNBENC(obj) \
-  (G_TYPE_CHECK_INSTANCE_CAST((obj), GST_TYPE_AMRNBENC, GstAmrnbEnc))
-#define GST_AMRNBENC_CLASS(klass) \
-  (G_TYPE_CHECK_CLASS_CAST((klass), GST_TYPE_AMRNBENC, GstAmrnbEncClass))
-#define GST_IS_AMRNBENC(obj) \
-  (G_TYPE_CHECK_INSTANCE_TYPE((obj), GST_TYPE_AMRNBENC))
-#define GST_IS_AMRNBENC_CLASS(klass) \
-  (G_TYPE_CHECK_CLASS_TYPE((klass), GST_TYPE_AMRNBENC))
-
-typedef struct _GstAmrnbEnc GstAmrnbEnc;
-typedef struct _GstAmrnbEncClass GstAmrnbEncClass;
-
-struct _GstAmrnbEnc {
-  GstAudioEncoder element;
-
-  /* library handle */
-  void *handle;
-
-  /* input settings */
-  gint channels, rate;
-  gint duration;
-
-  /* property */
-  enum Mode bandmode;
-};
-
-struct _GstAmrnbEncClass {
-  GstAudioEncoderClass parent_class;
-};
-
-GType gst_amrnbenc_get_type (void);
-GST_ELEMENT_REGISTER_DECLARE (amrnbenc);
-
-G_END_DECLS
-
-#endif /* __GST_AMRNBENC_H__ */

gst-plugins-ugly-1.22.7.tar.xz/ext/amrnb/meson.build Deleted

gst-plugins-ugly-1.22.7.tar.xz/ext/amrwbdec Deleted

gst-plugins-ugly-1.22.7.tar.xz/ext/amrwbdec/README Deleted

gst-plugins-ugly-1.22.7.tar.xz/ext/amrwbdec/amrwb.c Deleted

@@ -1,38 +0,0 @@
-/* GStreamer Adaptive Multi-Rate Wide-Band (AMR-WB) Decoder plugin
- * Copyright (C) 2006 Edgard Lima <edgard.lima@gmail.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include "amrwbdec.h"
-
-static gboolean
-plugin_init (GstPlugin * plugin)
-{
-  return GST_ELEMENT_REGISTER (amrwbdec, plugin);
-}
-
-
-GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
-    GST_VERSION_MINOR,
-    amrwbdec,
-    "Adaptive Multi-Rate Wide-Band Decoder",
-    plugin_init, VERSION, GST_LICENSE_UNKNOWN, GST_PACKAGE_NAME,
-    GST_PACKAGE_ORIGIN);

gst-plugins-ugly-1.22.7.tar.xz/ext/amrwbdec/amrwbdec.c Deleted

@@ -1,234 +0,0 @@
-/* GStreamer Adaptive Multi-Rate Narrow-Band (AMR-NB) plugin
- * Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-
-/**
- * SECTION:element-amrwbdec
- * @title: amrwbdec
- * @see_also: #GstAmrwbEnc
- *
- * AMR wideband decoder based on the
- * opencore codec implementation(http://sourceforge.net/projects/opencore-amr).
- *
- * ## Example launch line
- * |
- * gst-launch-1.0 filesrc location=abc.amr ! amrparse ! amrwbdec ! audioconvert ! audioresample ! autoaudiosink
- * |
- *
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include <gst/audio/audio.h>
-
-#include "amrwbdec.h"
-
-static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
-    GST_PAD_SINK,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/AMR-WB, "
-        "rate = (int) 16000, " "channels = (int) 1")
-    );
-
-static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
-    GST_PAD_SRC,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/x-raw, "
-        "format = (string) " GST_AUDIO_NE (S16) ", "
-        "layout = (string) interleaved, "
-        "rate = (int) 16000, " "channels = (int) 1")
-    );
-
-GST_DEBUG_CATEGORY_STATIC (gst_amrwbdec_debug);
-#define GST_CAT_DEFAULT gst_amrwbdec_debug
-
-#define L_FRAME16k      320     /* Frame size at 16kHz  */
-
-static const unsigned char block_size16 =
-    { 18, 24, 33, 37, 41, 47, 51, 59, 61,
-  6, 0, 0, 0, 0, 1, 1
-};
-
-static gboolean gst_amrwbdec_start (GstAudioDecoder * dec);
-static gboolean gst_amrwbdec_stop (GstAudioDecoder * dec);
-static gboolean gst_amrwbdec_set_format (GstAudioDecoder * dec, GstCaps * caps);
-static GstFlowReturn gst_amrwbdec_parse (GstAudioDecoder * dec,
-    GstAdapter * adapter, gint * offset, gint * length);
-static GstFlowReturn gst_amrwbdec_handle_frame (GstAudioDecoder * dec,
-    GstBuffer * buffer);
-
-#define gst_amrwbdec_parent_class parent_class
-G_DEFINE_TYPE (GstAmrwbDec, gst_amrwbdec, GST_TYPE_AUDIO_DECODER);
-GST_ELEMENT_REGISTER_DEFINE (amrwbdec, "amrwbdec",
-    GST_RANK_PRIMARY, GST_TYPE_AMRWBDEC);
-
-static void
-gst_amrwbdec_class_init (GstAmrwbDecClass * klass)
-{
-  GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-
-  gst_element_class_add_static_pad_template (element_class, &sink_template);
-  gst_element_class_add_static_pad_template (element_class, &src_template);
-
-  gst_element_class_set_static_metadata (element_class, "AMR-WB audio decoder",
-      "Codec/Decoder/Audio",
-      "Adaptive Multi-Rate Wideband audio decoder",
-      "Renato Araujo <renato.filho@indt.org.br>");
-
-  base_class->start = GST_DEBUG_FUNCPTR (gst_amrwbdec_start);
-  base_class->stop = GST_DEBUG_FUNCPTR (gst_amrwbdec_stop);
-  base_class->set_format = GST_DEBUG_FUNCPTR (gst_amrwbdec_set_format);
-  base_class->parse = GST_DEBUG_FUNCPTR (gst_amrwbdec_parse);
-  base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_amrwbdec_handle_frame);
-
-  GST_DEBUG_CATEGORY_INIT (gst_amrwbdec_debug, "amrwbdec", 0,
-      "AMR-WB audio decoder");
-}
-
-static void
-gst_amrwbdec_init (GstAmrwbDec * amrwbdec)
-{
-  gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (amrwbdec), TRUE);
-  gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
-      (amrwbdec), TRUE);
-  GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (amrwbdec));
-}
-
-static gboolean
-gst_amrwbdec_start (GstAudioDecoder * dec)
-{
-  GstAmrwbDec *amrwbdec = GST_AMRWBDEC (dec);
-
-  GST_DEBUG_OBJECT (dec, "start");
-  if (!(amrwbdec->handle = D_IF_init ()))
-    return FALSE;
-
-  amrwbdec->rate = 0;
-  amrwbdec->channels = 0;
-
-  return TRUE;
-}
-
-static gboolean
-gst_amrwbdec_stop (GstAudioDecoder * dec)
-{
-  GstAmrwbDec *amrwbdec = GST_AMRWBDEC (dec);
-
-  GST_DEBUG_OBJECT (dec, "stop");
-  D_IF_exit (amrwbdec->handle);
-
-  return TRUE;
-}
-
-static gboolean
-gst_amrwbdec_set_format (GstAudioDecoder * dec, GstCaps * caps)
-{
-  GstStructure *structure;
-  GstAmrwbDec *amrwbdec;
-  GstAudioInfo info;
-
-  amrwbdec = GST_AMRWBDEC (dec);
-
-  structure = gst_caps_get_structure (caps, 0);
-
-  /* get channel count */
-  gst_structure_get_int (structure, "channels", &amrwbdec->channels);
-  gst_structure_get_int (structure, "rate", &amrwbdec->rate);
-
-  /* create reverse caps */
-  gst_audio_info_init (&info);
-  gst_audio_info_set_format (&info,
-      GST_AUDIO_FORMAT_S16, amrwbdec->rate, amrwbdec->channels, NULL);
-
-  gst_audio_decoder_set_output_format (dec, &info);
-
-  return TRUE;
-}
-
-static GstFlowReturn
-gst_amrwbdec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
-    gint * offset, gint * length)
-{
-  GstAmrwbDec *amrwbdec = GST_AMRWBDEC (dec);
-  guint8 header1;
-  guint size;
-  gboolean sync, eos;
-  gint block, mode;
-
-  size = gst_adapter_available (adapter);
-  if (size < 1)
-    return GST_FLOW_ERROR;
-
-  gst_audio_decoder_get_parse_state (dec, &sync, &eos);
-
-  /* need to peek data to get the size */
-  gst_adapter_copy (adapter, header, 0, 1);
-  mode = (header0 >> 3) & 0x0F;
-  block = block_sizemode;
-
-  GST_DEBUG_OBJECT (amrwbdec, "mode %d, block %d", mode, block);
-
-  if (block) {
-    if (block > size)
-      return GST_FLOW_EOS;
-    *offset = 0;
-    *length = block;
-  } else {
-    /* no frame yet, skip one byte */
-    GST_LOG_OBJECT (amrwbdec, "skipping byte");
-    *offset = 1;
-    return GST_FLOW_EOS;
-  }
-
-  return GST_FLOW_OK;
-}
-
-static GstFlowReturn
-gst_amrwbdec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
-{
-  GstAmrwbDec *amrwbdec;
-  GstBuffer *out;
-  GstMapInfo inmap, outmap;
-
-  amrwbdec = GST_AMRWBDEC (dec);
-
-  /* no fancy flushing */
-  if (!buffer || !gst_buffer_get_size (buffer))
-    return GST_FLOW_OK;
-
-  /* the library seems to write into the source data, hence the copy. */
-  /* should be no problem */
-  gst_buffer_map (buffer, &inmap, GST_MAP_READ);
-
-  /* get output */
-  out = gst_buffer_new_and_alloc (sizeof (gint16) * L_FRAME16k);
-  gst_buffer_map (out, &outmap, GST_MAP_WRITE);
-
-  /* decode */
-  D_IF_decode (amrwbdec->handle, (unsigned char *) inmap.data,
-      (short int *) outmap.data, _good_frame);
-
-  gst_buffer_unmap (out, &outmap);
-  gst_buffer_unmap (buffer, &inmap);
-
-  /* send out */
-  return gst_audio_decoder_finish_frame (dec, out, 1);
-}

gst-plugins-ugly-1.22.7.tar.xz/ext/amrwbdec/amrwbdec.h Deleted

@@ -1,69 +0,0 @@
-/* GStreamer Adaptive Multi-Rate Wide-Band (AMR-WB) plugin
- * Copyright (C) 2006 Edgard Lima <edgard.lima@gmail.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-
-#ifndef __GST_AMRWBDEC_H__
-#define __GST_AMRWBDEC_H__
-
-#include <gst/gst.h>
-#include <gst/audio/gstaudiodecoder.h>
-
-#include <opencore-amrwb/dec_if.h>
-#include <opencore-amrwb/if_rom.h>
-
-G_BEGIN_DECLS
-
-#define GST_TYPE_AMRWBDEC			\
-  (gst_amrwbdec_get_type())
-#define GST_AMRWBDEC(obj)						\
-  (G_TYPE_CHECK_INSTANCE_CAST((obj), GST_TYPE_AMRWBDEC, GstAmrwbDec))
-#define GST_AMRWBDEC_CLASS(klass)					\
-  (G_TYPE_CHECK_CLASS_CAST((klass), GST_TYPE_AMRWBDEC, GstAmrwbDecClass))
-#define GST_IS_AMRWBDEC(obj)					\
-  (G_TYPE_CHECK_INSTANCE_TYPE((obj), GST_TYPE_AMRWBDEC))
-#define GST_IS_AMRWBDEC_CLASS(klass)			\
-  (G_TYPE_CHECK_CLASS_TYPE((klass), GST_TYPE_AMRWBDEC))
-
-typedef struct _GstAmrwbDec GstAmrwbDec;
-typedef struct _GstAmrwbDecClass GstAmrwbDecClass;
-
-/**
- * GstAmrwbDec:
- *
- * Opaque data structure.
- */
-struct _GstAmrwbDec {
-  GstAudioDecoder element;
-
-  /* library handle */
-  void *handle;
-
-  /* output settings */
-  gint channels, rate;
-};
-
-struct _GstAmrwbDecClass {
-  GstAudioDecoderClass parent_class;
-};
-
-GType gst_amrwbdec_get_type (void);
-GST_ELEMENT_REGISTER_DECLARE (amrwbdec);
-
-G_END_DECLS
-
-#endif /* __GST_AMRWBDEC_H__ */

gst-plugins-ugly-1.22.7.tar.xz/ext/amrwbdec/meson.build Deleted

gst-plugins-ugly-1.22.7.tar.xz/gst/realmedia/asmrules.c Deleted

@@ -1,712 +0,0 @@
-/* GStreamer
- * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-
-#include <string.h>
-#include <stdlib.h>
-
-#include "asmrules.h"
-
-#define MAX_RULE_LENGTH	2048
-
-/* define to enable some more debug */
-#undef DEBUG
-
-static GstASMNode *
-gst_asm_node_new (void)
-{
-  GstASMNode *node;
-
-  node = g_new0 (GstASMNode, 1);
-  node->type = GST_ASM_NODE_UNKNOWN;
-
-  return node;
-}
-
-static void
-gst_asm_node_free (GstASMNode * node)
-{
-  if (node->left)
-    gst_asm_node_free (node->left);
-  if (node->right)
-    gst_asm_node_free (node->right);
-  if (node->type == GST_ASM_NODE_VARIABLE && node->data.varname)
-    g_free (node->data.varname);
-  g_free (node);
-}
-
-static gfloat
-gst_asm_operator_eval (GstASMOp optype, gfloat left, gfloat right)
-{
-  gfloat result = 0.0;
-
-  switch (optype) {
-    case GST_ASM_OP_GREATER:
-      result = (gfloat) (left > right);
-      break;
-    case GST_ASM_OP_LESS:
-      result = (gfloat) (left < right);
-      break;
-    case GST_ASM_OP_GREATEREQUAL:
-      result = (gfloat) (left >= right);
-      break;
-    case GST_ASM_OP_LESSEQUAL:
-      result = (gfloat) (left <= right);
-      break;
-    case GST_ASM_OP_EQUAL:
-      result = (gfloat) (left == right);
-      break;
-    case GST_ASM_OP_NOTEQUAL:
-      result = (gfloat) (left != right);
-      break;
-    case GST_ASM_OP_AND:
-      result = (gfloat) (left && right);
-      break;
-    case GST_ASM_OP_OR:
-      result = (gfloat) (left || right);
-      break;
-    default:
-      break;
-  }
-  return result;
-}
-
-static gfloat
-gst_asm_node_evaluate (GstASMNode * node, GHashTable * vars)
-{
-  gfloat result = 0.0;
-
-  if (node == NULL)
-    return 0.0;
-
-  switch (node->type) {
-    case GST_ASM_NODE_VARIABLE:
-    {
-      gchar *val;
-
-      val = g_hash_table_lookup (vars, node->data.varname);
-      if (val)
-        result = (gfloat) atof (val);
-      break;
-    }
-    case GST_ASM_NODE_INTEGER:
-      result = (gfloat) node->data.intval;
-      break;
-    case GST_ASM_NODE_FLOAT:
-      result = node->data.floatval;
-      break;
-    case GST_ASM_NODE_OPERATOR:
-    {
-      gfloat left, right;
-
-      left = gst_asm_node_evaluate (node->left, vars);
-      right = gst_asm_node_evaluate (node->right, vars);
-
-      result = gst_asm_operator_eval (node->data.optype, left, right);
-      break;
-    }
-    default:
-      break;
-  }
-  return result;
-}
-
-#define IS_SPACE(p) (((p) == ' ') || ((p) == '\n') || \
-                     ((p) == '\r') || ((p) == '\t'))
-#define IS_RULE_DELIM(p) (((p) == ',') || ((p) == ';') || ((p) == ')'))
-#define IS_OPERATOR(p) (((p) == '>') || ((p) == '<') || \
-                        ((p) == '=') || ((p) == '!') || \
-                        ((p) == '&') || ((p) == '|'))
-#define IS_NUMBER(p) ((((p) >= '0') && ((p) <= '9')) || ((p) == '.'))
-#define IS_CHAR(p) (!IS_OPERATOR(ch) && !IS_RULE_DELIM(ch) && (ch != '\0'))
-
-#define IS_OP_TOKEN(t) (((t) == GST_ASM_TOKEN_AND) || ((t) == GST_ASM_TOKEN_OR))
-#define IS_COND_TOKEN(t) (((t) == GST_ASM_TOKEN_LESS) || ((t) == GST_ASM_TOKEN_LESSEQUAL) || \
-		((t) == GST_ASM_TOKEN_GREATER) || ((t) == GST_ASM_TOKEN_GREATEREQUAL) || \
-		((t) == GST_ASM_TOKEN_EQUAL) || ((t) == GST_ASM_TOKEN_NOTEQUAL))
-
-typedef struct
-{
-  const gchar *buffer;
-  gint pos;
-  gchar ch;
-
-  GstASMToken token;
-  gchar valMAX_RULE_LENGTH;
-} GstASMScan;
-
-#define NEXT_CHAR(scan) ((scan)->ch = (scan)->buffer(scan)->pos++)
-#define THIS_CHAR(scan) ((scan)->ch)
-
-static GstASMScan *
-gst_asm_scan_new (const gchar * buffer)
-{
-  GstASMScan *scan;
-
-  scan = g_new0 (GstASMScan, 1);
-  scan->buffer = buffer;
-  NEXT_CHAR (scan);
-
-  return scan;
-}
-
-static void
-gst_asm_scan_free (GstASMScan * scan)
-{
-  g_free (scan);
-}
-
-static void
-gst_asm_scan_string (GstASMScan * scan, gchar delim)
-{
-  gchar ch;
-  gint i = 0;
-
-  ch = THIS_CHAR (scan);
-  while ((ch != delim) && (ch != '\0')) {
-    if (i < MAX_RULE_LENGTH - 1)
-      scan->vali++ = ch;
-    ch = NEXT_CHAR (scan);
-    if (ch == '\\')
-      ch = NEXT_CHAR (scan);
-  }
-  scan->vali = '\0';
-
-  if (ch == delim)
-    NEXT_CHAR (scan);
-
-  scan->token = GST_ASM_TOKEN_STRING;
-}
-
-static void
-gst_asm_scan_number (GstASMScan * scan)
-{
-  gchar ch;
-  gint i = 0;
-  gboolean have_float = FALSE;
-
-  ch = THIS_CHAR (scan);
-  /* real strips all spaces that are not inside quotes for numbers */
-  while ((IS_NUMBER (ch) || IS_SPACE (ch))) {
-    if (i < (MAX_RULE_LENGTH - 1) && !IS_SPACE (ch))
-      scan->vali++ = ch;
-    if (ch == '.')
-      have_float = TRUE;
-    ch = NEXT_CHAR (scan);
-  }
-  scan->vali = '\0';
-
-  if (have_float)
-    scan->token = GST_ASM_TOKEN_FLOAT;
-  else
-    scan->token = GST_ASM_TOKEN_INT;
-}
-
-static void
-gst_asm_scan_identifier (GstASMScan * scan)
-{
-  gchar ch;
-  gint i = 0;
-
-  ch = THIS_CHAR (scan);
-  /* real strips all spaces that are not inside quotes for identifiers */
-  while ((IS_CHAR (ch) || IS_SPACE (ch))) {
-    if (i < (MAX_RULE_LENGTH - 1) && !IS_SPACE (ch))
-      scan->vali++ = ch;
-    ch = NEXT_CHAR (scan);
-  }
-  scan->vali = '\0';
-
-  scan->token = GST_ASM_TOKEN_IDENTIFIER;
-}
-
-static void
-gst_asm_scan_print_token (GstASMScan * scan)
-{
-#ifdef DEBUG
-  switch (scan->token) {
-    case GST_ASM_TOKEN_NONE:
-      g_print ("none\n");
-      break;
-    case GST_ASM_TOKEN_EOF:
-      g_print ("EOF\n");
-      break;
-
-    case GST_ASM_TOKEN_INT:
-      g_print ("INT %d\n", atoi (scan->val));
-      break;
-    case GST_ASM_TOKEN_FLOAT:
-      g_print ("FLOAT %f\n", atof (scan->val));
-      break;
-    case GST_ASM_TOKEN_IDENTIFIER:
-      g_print ("ID %s\n", scan->val);
-      break;
-    case GST_ASM_TOKEN_STRING:
-      g_print ("STRING %s\n", scan->val);
-      break;
-
-    case GST_ASM_TOKEN_HASH:
-      g_print ("HASH\n");
-      break;
-    case GST_ASM_TOKEN_SEMICOLON:
-      g_print ("SEMICOLON\n");
-      break;
-    case GST_ASM_TOKEN_COMMA:
-      g_print ("COMMA\n");
-      break;
-    case GST_ASM_TOKEN_EQUAL:
-      g_print ("==\n");
-      break;
-    case GST_ASM_TOKEN_NOTEQUAL:
-      g_print ("!=\n");
-      break;
-    case GST_ASM_TOKEN_AND:
-      g_print ("&&\n");
-      break;
-    case GST_ASM_TOKEN_OR:
-      g_print ("||\n");
-      break;
-    case GST_ASM_TOKEN_LESS:
-      g_print ("<\n");
-      break;
-    case GST_ASM_TOKEN_LESSEQUAL:
-      g_print ("<=\n");
-      break;
-    case GST_ASM_TOKEN_GREATER:
-      g_print (">\n");
-      break;
-    case GST_ASM_TOKEN_GREATEREQUAL:
-      g_print (">=\n");
-      break;
-    case GST_ASM_TOKEN_DOLLAR:
-      g_print ("$\n");
-      break;
-    case GST_ASM_TOKEN_LPAREN:
-      g_print ("(\n");
-      break;
-    case GST_ASM_TOKEN_RPAREN:
-      g_print (")\n");
-      break;
-    default:
-      break;
-  }
-#endif
-}
-
-static GstASMToken
-gst_asm_scan_next_token (GstASMScan * scan)
-{
-  gchar ch;
-
-  ch = THIS_CHAR (scan);
-
-  /* skip spaces */
-  while (IS_SPACE (ch))
-    ch = NEXT_CHAR (scan);
-
-  /* remove \ which is common in front of " */
-  while (ch == '\\')
-    ch = NEXT_CHAR (scan);
-
-  switch (ch) {
-    case '#':
-      scan->token = GST_ASM_TOKEN_HASH;
-      NEXT_CHAR (scan);
-      break;
-    case ';':
-      scan->token = GST_ASM_TOKEN_SEMICOLON;
-      NEXT_CHAR (scan);
-      break;
-    case ',':
-      scan->token = GST_ASM_TOKEN_COMMA;
-      NEXT_CHAR (scan);
-      break;
-    case '=':
-      scan->token = GST_ASM_TOKEN_EQUAL;
-      if (NEXT_CHAR (scan) == '=')
-        NEXT_CHAR (scan);
-      break;
-    case '!':
-      if (NEXT_CHAR (scan) == '=') {
-        scan->token = GST_ASM_TOKEN_NOTEQUAL;
-        NEXT_CHAR (scan);
-      }
-      break;
-    case '&':
-      scan->token = GST_ASM_TOKEN_AND;
-      if (NEXT_CHAR (scan) == '&')
-        NEXT_CHAR (scan);
-      break;
-    case '|':
-      scan->token = GST_ASM_TOKEN_OR;
-      if (NEXT_CHAR (scan) == '|')
-        NEXT_CHAR (scan);
-      break;
-    case '<':
-      scan->token = GST_ASM_TOKEN_LESS;
-      if (NEXT_CHAR (scan) == '=') {
-        scan->token = GST_ASM_TOKEN_LESSEQUAL;
-        NEXT_CHAR (scan);
-      }
-      break;
-    case '>':
-      scan->token = GST_ASM_TOKEN_GREATER;
-      if (NEXT_CHAR (scan) == '=') {
-        scan->token = GST_ASM_TOKEN_GREATEREQUAL;
-        NEXT_CHAR (scan);
-      }
-      break;
-    case '$':
-      scan->token = GST_ASM_TOKEN_DOLLAR;
-      NEXT_CHAR (scan);
-      break;
-    case '(':
-      scan->token = GST_ASM_TOKEN_LPAREN;
-      NEXT_CHAR (scan);
-      break;
-    case ')':
-      scan->token = GST_ASM_TOKEN_RPAREN;
-      NEXT_CHAR (scan);
-      break;
-    case '"':
-      NEXT_CHAR (scan);
-      gst_asm_scan_string (scan, '"');
-      break;
-    case '\'':
-      NEXT_CHAR (scan);
-      gst_asm_scan_string (scan, '\'');
-      break;
-    case '0':
-    case '1':
-    case '2':
-    case '3':
-    case '4':
-    case '5':
-    case '6':
-    case '7':
-    case '8':
-    case '9':
-      gst_asm_scan_number (scan);
-      break;
-    case '\0':
-      scan->token = GST_ASM_TOKEN_EOF;
-      break;
-    default:
-      gst_asm_scan_identifier (scan);
-      break;
-  }
-  gst_asm_scan_print_token (scan);
-  return scan->token;
-}
-
-static GstASMRule *
-gst_asm_rule_new (void)
-{
-  GstASMRule *rule;
-
-  rule = g_new (GstASMRule, 1);
-  rule->root = NULL;
-  rule->props = g_hash_table_new_full (g_str_hash, g_str_equal, g_free, g_free);
-
-  return rule;
-}
-
-static void
-gst_asm_rule_free (GstASMRule * rule)
-{
-  g_hash_table_destroy (rule->props);
-  if (rule->root)
-    gst_asm_node_free (rule->root);
-  g_free (rule);
-}
-
-static void
-gst_asm_rule_add_property (GstASMRule * rule, gchar * key, gchar * val)
-{
-  g_hash_table_insert (rule->props, key, val);
-}
-
-static GstASMNode *gst_asm_scan_parse_condition (GstASMScan * scan);
-
-static GstASMNode *
-gst_asm_scan_parse_operand (GstASMScan * scan)
-{
-  GstASMNode *node;
-
-  switch (scan->token) {
-    case GST_ASM_TOKEN_DOLLAR:
-      gst_asm_scan_next_token (scan);
-
-      if (scan->token != GST_ASM_TOKEN_IDENTIFIER)
-        g_warning ("identifier expected");
-
-      node = gst_asm_node_new ();
-      node->type = GST_ASM_NODE_VARIABLE;
-      node->data.varname = g_strdup (scan->val);
-      break;
-    case GST_ASM_TOKEN_INT:
-      node = gst_asm_node_new ();
-      node->type = GST_ASM_NODE_INTEGER;
-      node->data.intval = (gfloat) atof (scan->val);
-      break;
-    case GST_ASM_TOKEN_FLOAT:
-      node = gst_asm_node_new ();
-      node->type = GST_ASM_NODE_FLOAT;
-      node->data.floatval = atoi (scan->val);
-      break;
-    case GST_ASM_TOKEN_LPAREN:
-      gst_asm_scan_next_token (scan);
-      node = gst_asm_scan_parse_condition (scan);
-      if (scan->token != GST_ASM_TOKEN_RPAREN)
-        g_warning (") expected");
-      break;
-    default:
-      g_warning ("$ <number> or ) expected");
-      node = NULL;
-      break;
-  }
-  gst_asm_scan_next_token (scan);
-
-  return node;
-}
-
-static GstASMNode *
-gst_asm_scan_parse_expression (GstASMScan * scan)
-{
-  GstASMNode *node, *left;
-
-  node = gst_asm_scan_parse_operand (scan);
-
-  while (IS_COND_TOKEN (scan->token)) {
-    left = node;
-
-    node = gst_asm_node_new ();
-    node->type = GST_ASM_NODE_OPERATOR;
-    node->data.optype = (GstASMOp) scan->token;
-
-    gst_asm_scan_next_token (scan);
-
-    node->right = gst_asm_scan_parse_operand (scan);
-    node->left = left;
-  }
-  return node;
-}
-
-static GstASMNode *
-gst_asm_scan_parse_condition (GstASMScan * scan)
-{
-  GstASMNode *node, *left;
-
-  node = gst_asm_scan_parse_expression (scan);
-
-  while (IS_OP_TOKEN (scan->token)) {
-    left = node;
-
-    node = gst_asm_node_new ();
-    node->type = GST_ASM_NODE_OPERATOR;
-    node->data.optype = (GstASMOp) scan->token;
-
-    gst_asm_scan_next_token (scan);
-
-    node->right = gst_asm_scan_parse_expression (scan);
-    node->left = left;
-  }
-  return node;
-}
-
-static void
-gst_asm_scan_parse_property (GstASMRule * rule, GstASMScan * scan)
-{
-  gchar *key, *val;
-
-  if (scan->token != GST_ASM_TOKEN_IDENTIFIER) {
-    g_warning ("identifier expected");
-    return;
-  }
-  key = g_strdup (scan->val);
-
-  gst_asm_scan_next_token (scan);
-  if (scan->token != GST_ASM_TOKEN_EQUAL) {
-    g_warning ("= expected");
-    g_free (key);
-    return;
-  }
-  gst_asm_scan_next_token (scan);
-  val = g_strdup (scan->val);
-
-  gst_asm_rule_add_property (rule, key, val);
-  gst_asm_scan_next_token (scan);
-}
-
-static GstASMRule *
-gst_asm_scan_parse_rule (GstASMScan * scan)
-{
-  GstASMRule *rule;
-
-  rule = gst_asm_rule_new ();
-
-  if (scan->token == GST_ASM_TOKEN_HASH) {
-    gst_asm_scan_next_token (scan);
-    rule->root = gst_asm_scan_parse_condition (scan);
-    if (scan->token == GST_ASM_TOKEN_COMMA)
-      gst_asm_scan_next_token (scan);
-  }
-
-  if (scan->token != GST_ASM_TOKEN_SEMICOLON) {
-    gst_asm_scan_parse_property (rule, scan);
-    while (scan->token == GST_ASM_TOKEN_COMMA) {
-      gst_asm_scan_next_token (scan);
-      gst_asm_scan_parse_property (rule, scan);
-    }
-    gst_asm_scan_next_token (scan);
-  }
-  return rule;
-}
-
-static gboolean
-gst_asm_rule_evaluate (GstASMRule * rule, GHashTable * vars)
-{
-  gboolean res;
-
-  if (rule->root) {
-    res = (gboolean) gst_asm_node_evaluate (rule->root, vars);
-  } else
-    res = TRUE;
-
-  return res;
-}
-
-GstASMRuleBook *
-gst_asm_rule_book_new (const gchar * rulebook)
-{
-  GstASMRuleBook *book;
-  GstASMRule *rule = NULL;
-  GstASMScan *scan;
-  GstASMToken token;
-
-  book = g_new0 (GstASMRuleBook, 1);
-  book->rulebook = rulebook;
-
-  scan = gst_asm_scan_new (book->rulebook);
-  gst_asm_scan_next_token (scan);
-
-  do {
-    rule = gst_asm_scan_parse_rule (scan);
-    if (rule) {
-      book->rules = g_list_append (book->rules, rule);
-      book->n_rules++;
-    }
-    token = scan->token;
-  } while (token != GST_ASM_TOKEN_EOF);
-
-  gst_asm_scan_free (scan);
-
-  return book;
-}
-
-void
-gst_asm_rule_book_free (GstASMRuleBook * book)
-{
-  GList *walk;
-
-  for (walk = book->rules; walk; walk = g_list_next (walk)) {
-    GstASMRule *rule = (GstASMRule *) walk->data;
-
-    gst_asm_rule_free (rule);
-  }
-  g_list_free (book->rules);
-  g_free (book);
-}
-
-gint
-gst_asm_rule_book_match (GstASMRuleBook * book, GHashTable * vars,
-    gint * rulematches)
-{
-  GList *walk;
-  gint i, n = 0;
-
-  for (walk = book->rules, i = 0; walk; walk = g_list_next (walk), i++) {
-    GstASMRule *rule = (GstASMRule *) walk->data;
-
-    if (gst_asm_rule_evaluate (rule, vars)) {
-      rulematchesn++ = i;
-    }
-  }
-  return n;
-}
-
-#ifdef TEST
-gint
-main (gint argc, gchar * argv)
-{
-  GstASMRuleBook *book;
-  gint rulematchMAX_RULEMATCHES;
-  GHashTable *vars;
-  gint i, n;
-
-  static const gchar rules1 =
-      "#($Bandwidth < 67959),TimestampDelivery=T,DropByN=T,"
-      "priority=9;#($Bandwidth >= 67959) && ($Bandwidth < 167959),"
-      "AverageBandwidth=67959,Priority=9;#($Bandwidth >= 67959) && ($Bandwidth"
-      " < 167959),AverageBandwidth=0,Priority=5,OnDepend=\\\"1\\\";#($Bandwidth >= 167959)"
-      " && ($Bandwidth < 267959),AverageBandwidth=167959,Priority=9;#($Bandwidth >= 167959)"
-      " && ($Bandwidth < 267959),AverageBandwidth=0,Priority=5,OnDepend=\\\"3\\\";"
-      "#($Bandwidth >= 267959),AverageBandwidth=267959,Priority=9;#($Bandwidth >= 267959)"
-      ",AverageBandwidth=0,Priority=5,OnDepend=\\\"5\\\";";
-  static const gchar rules2 =
-      "AverageBandwidth=32041,Priority=5;AverageBandwidth=0,"
-      "Priority=5,OnDepend=\\\"0\\\", OffDepend=\\\"0\\\";";
-  static const gchar rules3 =
-      "#(($Bandwidth >= 27500) && ($OldPNMPlayer)),AverageBandwidth=27500,priority=9,PNMKeyframeRule=T;#(($Bandwidth >= 27500) && ($OldPNMPlayer)),AverageBandwidth=0,priority=5,PNMNonKeyframeRule=T;#(($Bandwidth < 27500) && ($OldPNMPlayer)),TimestampDelivery=T,DropByN=T,priority=9,PNMThinningRule=T;#($Bandwidth < 13899),TimestampDelivery=T,DropByN=T,priority=9;#($Bandwidth >= 13899) && ($Bandwidth < 19000),AverageBandwidth=13899,Priority=9;#($Bandwidth >= 13899) && ($Bandwidth < 19000),AverageBandwidth=0,Priority=5,OnDepend=\\\"4\\\";#($Bandwidth >= 19000) && ($Bandwidth < 27500),AverageBandwidth=19000,Priority=9;#($Bandwidth >= 19000) && ($Bandwidth < 27500),AverageBandwidth=0,Priority=5,OnDepend=\\\"6\\\";#($Bandwidth >= 27500) && ($Bandwidth < 132958),AverageBandwidth=27500,Priority=9;#($Bandwidth >= 27500) && ($Bandwidth < 132958),AverageBandwidth=0,Priority=5,OnDepend=\\\"8\\\";#($Bandwidth >= 132958) && ($Bandwidth < 187958),AverageBandwidth=132958,Priority=9;#($Bandwidth >= 132958) && ($Bandwidth < 187958),AverageBandwidth=0,Priority=5,OnDepend=\\\"10\\\";#($Bandwidth >= 187958),AverageBandwidth=187958,Priority=9;#($Bandwidth >= 187958),AverageBandwidth=0,Priority=5,OnDepend=\\\"12\\\";";
-
-  vars = g_hash_table_new (g_str_hash, g_str_equal);
-  g_hash_table_insert (vars, (gchar *) "Bandwidth", (gchar *) "300000");
-
-  book = gst_asm_rule_book_new (rules1);
-  n = gst_asm_rule_book_match (book, vars, rulematch);
-  gst_asm_rule_book_free (book);
-
-  g_print ("%d rules matched\n", n);
-  for (i = 0; i < n; i++) {
-    g_print ("rule %d matched\n", rulematchi);
-  }
-
-  book = gst_asm_rule_book_new (rules2);
-  n = gst_asm_rule_book_match (book, vars, rulematch);
-  gst_asm_rule_book_free (book);
-
-  g_print ("%d rules matched\n", n);
-  for (i = 0; i < n; i++) {
-    g_print ("rule %d matched\n", rulematchi);
-  }
-
-  book = gst_asm_rule_book_new (rules3);
-  n = gst_asm_rule_book_match (book, vars, rulematch);
-  gst_asm_rule_book_free (book);
-
-
-  g_print ("%d rules matched\n", n);
-  for (i = 0; i < n; i++) {
-    g_print ("rule %d matched\n", rulematchi);
-  }
-
-  g_hash_table_destroy (vars);
-
-  return 0;
-}
-#endif

gst-plugins-ugly-1.22.7.tar.xz/gst/realmedia/asmrules.h Deleted

@@ -1,115 +0,0 @@
-/* GStreamer
- * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-
-#ifndef __GST_ASM_RULES_H__
-#define __GST_ASM_RULES_H__
-
-#include <gst/gst.h>
-
-G_BEGIN_DECLS
-
-#define MAX_RULEMATCHES 16
-
-typedef struct _GstASMNode GstASMNode;
-typedef struct _GstASMRule GstASMRule;
-typedef struct _GstASMRuleBook GstASMRuleBook;
-
-typedef enum {
-  GST_ASM_TOKEN_NONE,
-  GST_ASM_TOKEN_EOF,
-
-  GST_ASM_TOKEN_INT,
-  GST_ASM_TOKEN_FLOAT,
-  GST_ASM_TOKEN_IDENTIFIER,
-  GST_ASM_TOKEN_STRING,
-  
-  GST_ASM_TOKEN_HASH,
-  GST_ASM_TOKEN_SEMICOLON,
-  GST_ASM_TOKEN_COMMA,
-  GST_ASM_TOKEN_DOLLAR,
-
-  GST_ASM_TOKEN_LPAREN,
-  GST_ASM_TOKEN_RPAREN,
-
-  GST_ASM_TOKEN_GREATER,
-  GST_ASM_TOKEN_LESS,
-  GST_ASM_TOKEN_GREATEREQUAL,
-  GST_ASM_TOKEN_LESSEQUAL,
-  GST_ASM_TOKEN_EQUAL,
-  GST_ASM_TOKEN_NOTEQUAL,
-
-  GST_ASM_TOKEN_AND,
-  GST_ASM_TOKEN_OR
-} GstASMToken;
-
-typedef enum {
-  GST_ASM_NODE_UNKNOWN,
-  GST_ASM_NODE_VARIABLE,
-  GST_ASM_NODE_INTEGER,
-  GST_ASM_NODE_FLOAT,
-  GST_ASM_NODE_OPERATOR
-} GstASMNodeType;
-
-typedef enum {
-  GST_ASM_OP_GREATER      = GST_ASM_TOKEN_GREATER,
-  GST_ASM_OP_LESS         = GST_ASM_TOKEN_LESS,
-  GST_ASM_OP_GREATEREQUAL = GST_ASM_TOKEN_GREATEREQUAL,
-  GST_ASM_OP_LESSEQUAL    = GST_ASM_TOKEN_LESSEQUAL,
-  GST_ASM_OP_EQUAL        = GST_ASM_TOKEN_EQUAL,
-  GST_ASM_OP_NOTEQUAL     = GST_ASM_TOKEN_NOTEQUAL,
-
-  GST_ASM_OP_AND          = GST_ASM_TOKEN_AND,
-  GST_ASM_OP_OR           = GST_ASM_TOKEN_OR
-} GstASMOp;
-
-struct _GstASMNode {
-  GstASMNodeType  type;
-
-  union {
-    gchar   *varname;
-    gint     intval;
-    gfloat   floatval;
-    GstASMOp optype;
-  } data;
-
-  GstASMNode     *left;
-  GstASMNode     *right;
-};
-
-struct _GstASMRule {
-  GstASMNode *root;
-  GHashTable *props;
-};
-
-struct _GstASMRuleBook {
-  const gchar *rulebook;
-
-  guint        n_rules;
-  GList       *rules;
-};
-
-G_END_DECLS
-
-GstASMRuleBook*   gst_asm_rule_book_new     (const gchar *rulebook);
-void              gst_asm_rule_book_free    (GstASMRuleBook *book);
-
-gint              gst_asm_rule_book_match   (GstASMRuleBook *book, GHashTable *vars, 
-		                             gint *rulematches);
-
-#endif /* __GST_ASM_RULES_H__ */

gst-plugins-ugly-1.22.7.tar.xz/gst/realmedia/gstrdtbuffer.c Deleted

@@ -1,477 +0,0 @@
-/* GStreamer
- * Copyright (C) <2008> Wim Taymans <wim.taymans@gmail.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-#include <string.h>
-
-#include "gstrdtbuffer.h"
-
-gboolean
-gst_rdt_buffer_validate_data (guint8 * data, guint len)
-{
-  return TRUE;
-}
-
-gboolean
-gst_rdt_buffer_validate (GstBuffer * buffer)
-{
-  return TRUE;
-}
-
-guint
-gst_rdt_buffer_get_packet_count (GstBuffer * buffer)
-{
-  GstRDTPacket packet;
-  guint count;
-
-  g_return_val_if_fail (GST_IS_BUFFER (buffer), 0);
-
-  count = 0;
-  if (gst_rdt_buffer_get_first_packet (buffer, &packet)) {
-    do {
-      count++;
-    } while (gst_rdt_packet_move_to_next (&packet));
-  }
-  return count;
-}
-
-static gboolean
-read_packet_header (GstRDTPacket * packet)
-{
-  GstMapInfo map;
-  guint8 *data;
-  gsize size;
-  guint offset;
-  guint length;
-  guint length_offset;
-
-  g_return_val_if_fail (packet != NULL, FALSE);
-  g_return_val_if_fail (GST_IS_BUFFER (packet->buffer), FALSE);
-
-  gst_buffer_map (packet->buffer, &map, GST_MAP_READ);
-  data = map.data;
-  size = map.size;
-
-  offset = packet->offset;
-
-  /* check if we are at the end of the buffer, we add 3 because we also want to
-   * ensure we can read the type, which is always at offset 1 and 2 bytes long. */
-  if (offset + 3 > size)
-    goto packet_end;
-
-  /* read type */
-  packet->type = GST_READ_UINT16_BE (&dataoffset + 1);
-
-  length = -1;
-  length_offset = -1;
-
-  /* figure out the length of the packet, this depends on the type */
-  if (GST_RDT_IS_DATA_TYPE (packet->type)) {
-    if (dataoffset & 0x80)
-      /* length is present */
-      length_offset = 3;
-  } else {
-    switch (packet->type) {
-      case GST_RDT_TYPE_ASMACTION:
-        if (dataoffset & 0x80)
-          length_offset = 5;
-        break;
-      case GST_RDT_TYPE_BWREPORT:
-        if (dataoffset & 0x80)
-          length_offset = 3;
-        break;
-      case GST_RDT_TYPE_ACK:
-        if (dataoffset & 0x80)
-          length_offset = 3;
-        break;
-      case GST_RDT_TYPE_RTTREQ:
-        length = 3;
-        break;
-      case GST_RDT_TYPE_RTTRESP:
-        length = 11;
-        break;
-      case GST_RDT_TYPE_CONGESTION:
-        length = 11;
-        break;
-      case GST_RDT_TYPE_STREAMEND:
-        length = 9;
-        /* total_reliable */
-        if (dataoffset & 0x80)
-          length += 2;
-        /* stream_id_expansion */
-        if ((dataoffset & 0x7c) == 0x7c)
-          length += 2;
-        /* ext_flag, FIXME, get string length */
-        if ((dataoffset & 0x1) == 0x1)
-          length += 7;
-        break;
-      case GST_RDT_TYPE_REPORT:
-        if (dataoffset & 0x80)
-          length_offset = 3;
-        break;
-      case GST_RDT_TYPE_LATENCY:
-        if (dataoffset & 0x80)
-          length_offset = 3;
-        break;
-      case GST_RDT_TYPE_INFOREQ:
-        length = 3;
-        /* request_time_ms */
-        if (dataoffset & 0x2)
-          length += 2;
-        break;
-      case GST_RDT_TYPE_INFORESP:
-        length = 3;
-        /* has_rtt_info */
-        if (dataoffset & 0x4) {
-          length += 4;
-          /* is_delayed */
-          if (dataoffset & 0x2) {
-            length += 4;
-          }
-        }
-        if (dataoffset & 0x1) {
-          /* buffer_info_count, FIXME read and skip */
-          length += 2;
-        }
-        break;
-      case GST_RDT_TYPE_AUTOBW:
-        if (dataoffset & 0x80)
-          length_offset = 3;
-        break;
-      case GST_RDT_TYPE_INVALID:
-      default:
-        goto unknown_packet;
-    }
-  }
-
-  if (length != -1) {
-    /* we have a fixed length */
-    packet->length = length;
-  } else if (length_offset != -1) {
-    /* we can read the length from an offset */
-    packet->length = GST_READ_UINT16_BE (&datalength_offset);
-  } else {
-    /* length is remainder of packet */
-    packet->length = size - offset;
-  }
-  gst_buffer_unmap (packet->buffer, &map);
-
-  /* the length should be smaller than the remaining size */
-  if (packet->length + offset > size)
-    goto invalid_length;
-
-  return TRUE;
-
-  /* ERRORS */
-packet_end:
-  {
-    gst_buffer_unmap (packet->buffer, &map);
-    return FALSE;
-  }
-unknown_packet:
-  {
-    packet->type = GST_RDT_TYPE_INVALID;
-    gst_buffer_unmap (packet->buffer, &map);
-    return FALSE;
-  }
-invalid_length:
-  {
-    packet->type = GST_RDT_TYPE_INVALID;
-    packet->length = 0;
-    return FALSE;
-  }
-}
-
-gboolean
-gst_rdt_buffer_get_first_packet (GstBuffer * buffer, GstRDTPacket * packet)
-{
-  g_return_val_if_fail (GST_IS_BUFFER (buffer), FALSE);
-  g_return_val_if_fail (packet != NULL, FALSE);
-
-  /* init to 0 */
-  packet->buffer = buffer;
-  packet->offset = 0;
-  packet->type = GST_RDT_TYPE_INVALID;
-  memset (&packet->map, 0, sizeof (GstMapInfo));
-
-  if (!read_packet_header (packet))
-    return FALSE;
-
-  return TRUE;
-}
-
-gboolean
-gst_rdt_packet_move_to_next (GstRDTPacket * packet)
-{
-  g_return_val_if_fail (packet != NULL, FALSE);
-  g_return_val_if_fail (packet->type != GST_RDT_TYPE_INVALID, FALSE);
-  g_return_val_if_fail (GST_IS_BUFFER (packet->buffer), FALSE);
-
-  /* if we have an invalid packet, it must be the last, 
-   * return FALSE */
-  if (packet->type == GST_RDT_TYPE_INVALID)
-    goto end;
-
-  /* move to next packet */
-  packet->offset += packet->length;
-
-  /* try to read new header */
-  if (!read_packet_header (packet))
-    goto end;
-
-  return TRUE;
-
-  /* ERRORS */
-end:
-  {
-    packet->type = GST_RDT_TYPE_INVALID;
-    return FALSE;
-  }
-}
-
-GstRDTType
-gst_rdt_packet_get_type (GstRDTPacket * packet)
-{
-  g_return_val_if_fail (packet != NULL, GST_RDT_TYPE_INVALID);
-  g_return_val_if_fail (packet->type != GST_RDT_TYPE_INVALID,
-      GST_RDT_TYPE_INVALID);
-
-  return packet->type;
-}
-
-guint16
-gst_rdt_packet_get_length (GstRDTPacket * packet)
-{
-  g_return_val_if_fail (packet != NULL, 0);
-  g_return_val_if_fail (packet->type != GST_RDT_TYPE_INVALID, 0);
-
-  return packet->length;
-}
-
-GstBuffer *
-gst_rdt_packet_to_buffer (GstRDTPacket * packet)
-{
-  GstBuffer *result;
-
-  g_return_val_if_fail (packet != NULL, NULL);
-  g_return_val_if_fail (packet->type != GST_RDT_TYPE_INVALID, NULL);
-
-  result =
-      gst_buffer_copy_region (packet->buffer, GST_BUFFER_COPY_ALL,
-      packet->offset, packet->length);
-  /* timestamp applies to all packets in this buffer */
-  GST_BUFFER_TIMESTAMP (result) = GST_BUFFER_TIMESTAMP (packet->buffer);
-
-  return result;
-}
-
-gint
-gst_rdt_buffer_compare_seqnum (guint16 seqnum1, guint16 seqnum2)
-{
-  return (gint16) (seqnum2 - seqnum1);
-}
-
-guint16
-gst_rdt_packet_data_get_seq (GstRDTPacket * packet)
-{
-  GstMapInfo map;
-  guint header;
-  guint16 result;
-
-  g_return_val_if_fail (packet != NULL, FALSE);
-  g_return_val_if_fail (GST_RDT_IS_DATA_TYPE (packet->type), FALSE);
-
-  gst_buffer_map (packet->buffer, &map, GST_MAP_READ);
-
-  /* skip header bits */
-  header = packet->offset + 1;
-
-  /* read seq_no */
-  result = GST_READ_UINT16_BE (&map.dataheader);
-
-  gst_buffer_unmap (packet->buffer, &map);
-
-  return result;
-}
-
-guint8 *
-gst_rdt_packet_data_map (GstRDTPacket * packet, guint * size)
-{
-  GstMapInfo map;
-  guint header;
-  gboolean length_included_flag;
-  gboolean need_reliable_flag;
-  guint8 stream_id;
-  guint8 asm_rule_number;
-
-  g_return_val_if_fail (packet != NULL, NULL);
-  g_return_val_if_fail (packet->map.data == NULL, NULL);
-  g_return_val_if_fail (GST_RDT_IS_DATA_TYPE (packet->type), NULL);
-
-  gst_buffer_map (packet->buffer, &map, GST_MAP_READ);
-
-  header = packet->offset;
-
-  length_included_flag = (map.dataheader & 0x80) == 0x80;
-  need_reliable_flag = (map.dataheader & 0x40) == 0x40;
-  stream_id = (map.dataheader & 0x3e) >> 1;
-
-  /* skip seq_no and header bits */
-  header += 3;
-
-  if (length_included_flag) {
-    /* skip length */
-    header += 2;
-  }
-  asm_rule_number = (map.dataheader & 0x3f);
-
-  /* skip timestamp and asm_rule_number */
-  header += 5;
-
-  if (stream_id == 0x1f) {
-    /* skip stream_id_expansion */
-    header += 2;
-  }
-  if (need_reliable_flag) {
-    /* skip total_reliable */
-    header += 2;
-  }
-  if (asm_rule_number == 63) {
-    /* skip asm_rule_number_expansion */
-    header += 2;
-  }
-
-  if (size)
-    *size = packet->length - (header - packet->offset);
-
-  packet->map = map;
-
-  return &map.dataheader;
-}
-
-gboolean
-gst_rdt_packet_data_unmap (GstRDTPacket * packet)
-{
-  g_return_val_if_fail (packet != NULL, FALSE);
-  g_return_val_if_fail (packet->map.data != NULL, FALSE);
-
-  gst_buffer_unmap (packet->buffer, &packet->map);
-  packet->map.data = NULL;
-
-  return TRUE;
-}
-
-guint16
-gst_rdt_packet_data_get_stream_id (GstRDTPacket * packet)
-{
-  GstMapInfo map;
-  guint16 result;
-  guint header;
-  gboolean length_included_flag;
-
-  g_return_val_if_fail (packet != NULL, 0);
-  g_return_val_if_fail (GST_RDT_IS_DATA_TYPE (packet->type), 0);
-
-  gst_buffer_map (packet->buffer, &map, GST_MAP_READ);
-
-  header = packet->offset;
-
-  length_included_flag = (map.dataheader & 0x80) == 0x80;
-  result = (map.dataheader & 0x3e) >> 1;
-  if (result == 31) {
-    /* skip seq_no and header bits */
-    header += 3;
-
-    if (length_included_flag) {
-      /* skip length */
-      header += 2;
-    }
-    /* skip asm_rule_number and timestamp */
-    header += 5;
-
-    /* stream_id_expansion */
-    result = GST_READ_UINT16_BE (&map.dataheader);
-  }
-  gst_buffer_unmap (packet->buffer, &map);
-
-  return result;
-}
-
-guint32
-gst_rdt_packet_data_get_timestamp (GstRDTPacket * packet)
-{
-  GstMapInfo map;
-  guint header;
-  gboolean length_included_flag;
-  guint32 result;
-
-  g_return_val_if_fail (packet != NULL, 0);
-  g_return_val_if_fail (GST_RDT_IS_DATA_TYPE (packet->type), 0);
-
-  gst_buffer_map (packet->buffer, &map, GST_MAP_READ);
-
-  header = packet->offset;
-
-  length_included_flag = (map.dataheader & 0x80) == 0x80;
-
-  /* skip seq_no and header bits */
-  header += 3;
-
-  if (length_included_flag) {
-    /* skip length */
-    header += 2;
-  }
-  /* skip asm_rule_number */
-  header += 1;
-
-  /* get timestamp */
-  result = GST_READ_UINT32_BE (&map.dataheader);
-  gst_buffer_unmap (packet->buffer, &map);
-
-  return result;
-}
-
-guint8
-gst_rdt_packet_data_get_flags (GstRDTPacket * packet)
-{
-  GstMapInfo map;
-  guint8 result;
-  guint header;
-  gboolean length_included_flag;
-
-  g_return_val_if_fail (packet != NULL, 0);
-  g_return_val_if_fail (GST_RDT_IS_DATA_TYPE (packet->type), 0);
-
-  gst_buffer_map (packet->buffer, &map, GST_MAP_READ);
-
-  header = packet->offset;
-
-  length_included_flag = (map.dataheader & 0x80) == 0x80;
-
-  /* skip seq_no and header bits */
-  header += 3;
-
-  if (length_included_flag) {
-    /* skip length */
-    header += 2;
-  }
-  /* get flags */
-  result = map.dataheader;
-  gst_buffer_unmap (packet->buffer, &map);
-
-  return result;
-}

gst-plugins-ugly-1.22.7.tar.xz/gst/realmedia/gstrdtbuffer.h Deleted

@@ -1,121 +0,0 @@
-/* GStreamer
- * Copyright (C) <2008> Wim Taymans <wim.taymans@gmail.com>
- *
- * gstrdtbuffer.h: various helper functions to manipulate buffers
- *     with RDT payload.
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-
-#ifndef __GST_RDTBUFFER_H__
-#define __GST_RDTBUFFER_H__
-
-#include <gst/gst.h>
-
-G_BEGIN_DECLS
-/**
- * GstRDTType:
- * @GST_RDT_TYPE_INVALID:
- * @GST_RDT_TYPE_ASMACTION:
- * @GST_RDT_TYPE_ACK:
- * @GST_RDT_TYPE_RTTREQ:
- * @GST_RDT_TYPE_RTTRESP:
- * @GST_RDT_TYPE_CONGESTION:
- * @GST_RDT_TYPE_STREAMEND:
- * @GST_RDT_TYPE_LATENCY:
- * @GST_RDT_TYPE_INFOREQ:
- * @GST_RDT_TYPE_INFORESP:
- * @GST_RDT_TYPE_AUTOBW:
- *
- * Different RDT packet types.
- */
-typedef enum
-{
-  GST_RDT_TYPE_INVALID     = 0xffff,
-  GST_RDT_TYPE_ASMACTION   = 0xff00,
-  GST_RDT_TYPE_BWREPORT    = 0xff01,
-  GST_RDT_TYPE_ACK         = 0xff02,
-  GST_RDT_TYPE_RTTREQ      = 0xff03,
-  GST_RDT_TYPE_RTTRESP     = 0xff04,
-  GST_RDT_TYPE_CONGESTION  = 0xff05,
-  GST_RDT_TYPE_STREAMEND   = 0xff06,
-  GST_RDT_TYPE_REPORT      = 0xff07,
-  GST_RDT_TYPE_LATENCY     = 0xff08,
-  GST_RDT_TYPE_INFOREQ     = 0xff09,
-  GST_RDT_TYPE_INFORESP    = 0xff0a,
-  GST_RDT_TYPE_AUTOBW      = 0xff0b
-} GstRDTType;
-
-/**
- * GST_RDT_IS_DATA_TYPE:
- * @t: the #GstRDTType to check
- *
- * Check if @t is a data packet type.
- */
-#define GST_RDT_IS_DATA_TYPE(t) ((t) < 0xff00)
-
-typedef struct _GstRDTPacket GstRDTPacket;
-
-/**
- * GstRDTPacket:
- * @buffer: pointer to RDT buffer
- * @offset: offset of packet in buffer data
- *
- * Data structure that points to a packet at @offset in @buffer.
- * The size of the structure is made public to allow stack allocations.
- */
-struct _GstRDTPacket
-{
-  GstBuffer   *buffer;
-  guint        offset;
-
-  /*< private >*/
-  GstRDTType   type;         /* type of current packet */
-  guint16      length;       /* length of current packet in bytes */
-  GstMapInfo   map;          /* last mapped data */
-};
-
-/* validate buffers */
-gboolean        gst_rdt_buffer_validate_data      (guint8 *data, guint len);
-gboolean        gst_rdt_buffer_validate           (GstBuffer *buffer);
-
-/* retrieving packets */
-guint           gst_rdt_buffer_get_packet_count   (GstBuffer *buffer);
-gboolean        gst_rdt_buffer_get_first_packet   (GstBuffer *buffer, GstRDTPacket *packet);
-gboolean        gst_rdt_packet_move_to_next       (GstRDTPacket *packet);
-
-/* working with packets */
-GstRDTType      gst_rdt_packet_get_type           (GstRDTPacket *packet);
-guint16         gst_rdt_packet_get_length         (GstRDTPacket *packet);
-GstBuffer*      gst_rdt_packet_to_buffer          (GstRDTPacket *packet);
-
-
-/* data packets */
-guint16         gst_rdt_packet_data_get_seq       (GstRDTPacket *packet);
-guint8 *        gst_rdt_packet_data_map           (GstRDTPacket *packet, guint *size);
-gboolean        gst_rdt_packet_data_unmap         (GstRDTPacket *packet);
-guint16         gst_rdt_packet_data_get_stream_id (GstRDTPacket *packet);
-guint32         gst_rdt_packet_data_get_timestamp (GstRDTPacket *packet);
-
-guint8          gst_rdt_packet_data_get_flags     (GstRDTPacket * packet);
-
-/* utils */
-gint            gst_rdt_buffer_compare_seqnum     (guint16 seqnum1, guint16 seqnum2);
-
-G_END_DECLS
-
-#endif /* __GST_RDTBUFFER_H__ */
-

gst-plugins-ugly-1.22.7.tar.xz/gst/realmedia/rdtdepay.c Deleted

@@ -1,496 +0,0 @@
-/* GStreamer
- * Copyright (C) <2006> Lutz Mueller <lutz at topfrose dot de>
- *               <2006> Wim Taymans <wim@fluendo.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-
-#ifdef HAVE_CONFIG_H
-#  include "config.h"
-#endif
-
-#include <string.h>
-
-#include "gstrdtbuffer.h"
-#include "rdtdepay.h"
-
-GST_DEBUG_CATEGORY_STATIC (rdtdepay_debug);
-#define GST_CAT_DEFAULT rdtdepay_debug
-
-/* RDTDepay signals and args */
-enum
-{
-  /* FILL ME */
-  LAST_SIGNAL
-};
-
-enum
-{
-  PROP_0,
-};
-
-static GstStaticPadTemplate gst_rdt_depay_src_template =
-GST_STATIC_PAD_TEMPLATE ("src",
-    GST_PAD_SRC,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("application/vnd.rn-realmedia")
-    );
-
-static GstStaticPadTemplate gst_rdt_depay_sink_template =
-GST_STATIC_PAD_TEMPLATE ("sink",
-    GST_PAD_SINK,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("application/x-rdt, "
-        "media = (string) \"application\", "
-        "clock-rate = (int) 1, MAX , "
-        "encoding-name = (string) \"X-REAL-RDT\""
-        /* All optional parameters
-         *
-         * "config=" 
-         */
-    )
-    );
-
-#define gst_rdt_depay_parent_class parent_class
-G_DEFINE_TYPE (GstRDTDepay, gst_rdt_depay, GST_TYPE_ELEMENT);
-GST_ELEMENT_REGISTER_DEFINE (rdtdepay, "rdtdepay",
-    GST_RANK_MARGINAL, GST_TYPE_RDT_DEPAY);
-
-static void gst_rdt_depay_finalize (GObject * object);
-
-static GstStateChangeReturn gst_rdt_depay_change_state (GstElement *
-    element, GstStateChange transition);
-
-static gboolean gst_rdt_depay_sink_event (GstPad * pad, GstObject * parent,
-    GstEvent * event);
-static GstFlowReturn gst_rdt_depay_chain (GstPad * pad, GstObject * parent,
-    GstBuffer * buf);
-
-static void
-gst_rdt_depay_class_init (GstRDTDepayClass * klass)
-{
-  GObjectClass *gobject_class;
-  GstElementClass *gstelement_class;
-
-  gobject_class = (GObjectClass *) klass;
-  gstelement_class = (GstElementClass *) klass;
-
-  parent_class = g_type_class_peek_parent (klass);
-
-  gobject_class->finalize = gst_rdt_depay_finalize;
-
-  gstelement_class->change_state = gst_rdt_depay_change_state;
-
-  gst_element_class_add_static_pad_template (gstelement_class,
-      &gst_rdt_depay_src_template);
-  gst_element_class_add_static_pad_template (gstelement_class,
-      &gst_rdt_depay_sink_template);
-
-  gst_element_class_set_static_metadata (gstelement_class, "RDT packet parser",
-      "Codec/Depayloader/Network",
-      "Extracts RealMedia from RDT packets",
-      "Lutz Mueller <lutz at topfrose dot de>, "
-      "Wim Taymans <wim@fluendo.com>");
-
-  GST_DEBUG_CATEGORY_INIT (rdtdepay_debug, "rdtdepay",
-      0, "Depayloader for RDT RealMedia packets");
-}
-
-static void
-gst_rdt_depay_init (GstRDTDepay * rdtdepay)
-{
-  rdtdepay->sinkpad =
-      gst_pad_new_from_static_template (&gst_rdt_depay_sink_template, "sink");
-  gst_pad_set_chain_function (rdtdepay->sinkpad, gst_rdt_depay_chain);
-  gst_pad_set_event_function (rdtdepay->sinkpad, gst_rdt_depay_sink_event);
-  gst_element_add_pad (GST_ELEMENT_CAST (rdtdepay), rdtdepay->sinkpad);
-
-  rdtdepay->srcpad =
-      gst_pad_new_from_static_template (&gst_rdt_depay_src_template, "src");
-  gst_element_add_pad (GST_ELEMENT_CAST (rdtdepay), rdtdepay->srcpad);
-}
-
-static void
-gst_rdt_depay_finalize (GObject * object)
-{
-  GstRDTDepay *rdtdepay;
-
-  rdtdepay = GST_RDT_DEPAY (object);
-
-  if (rdtdepay->header)
-    gst_buffer_unref (rdtdepay->header);
-
-  G_OBJECT_CLASS (parent_class)->finalize (object);
-}
-
-static gboolean
-gst_rdt_depay_setcaps (GstPad * pad, GstCaps * caps)
-{
-  GstStructure *structure;
-  GstRDTDepay *rdtdepay;
-  GstCaps *srccaps;
-  gint clock_rate = 1000;       /* default */
-  const GValue *value;
-  GstBuffer *header;
-
-  rdtdepay = GST_RDT_DEPAY (GST_PAD_PARENT (pad));
-
-  structure = gst_caps_get_structure (caps, 0);
-
-  if (gst_structure_has_field (structure, "clock-rate"))
-    gst_structure_get_int (structure, "clock-rate", &clock_rate);
-
-  /* config contains the RealMedia header as a buffer. */
-  value = gst_structure_get_value (structure, "config");
-  if (!value)
-    goto no_header;
-
-  header = gst_value_get_buffer (value);
-  if (!header)
-    goto no_header;
-
-  /* get other values for newsegment */
-  value = gst_structure_get_value (structure, "npt-start");
-  if (value && G_VALUE_HOLDS_UINT64 (value))
-    rdtdepay->npt_start = g_value_get_uint64 (value);
-  else
-    rdtdepay->npt_start = 0;
-  GST_DEBUG_OBJECT (rdtdepay, "NPT start %" G_GUINT64_FORMAT,
-      rdtdepay->npt_start);
-
-  value = gst_structure_get_value (structure, "npt-stop");
-  if (value && G_VALUE_HOLDS_UINT64 (value))
-    rdtdepay->npt_stop = g_value_get_uint64 (value);
-  else
-    rdtdepay->npt_stop = -1;
-
-  GST_DEBUG_OBJECT (rdtdepay, "NPT stop %" G_GUINT64_FORMAT,
-      rdtdepay->npt_stop);
-
-  value = gst_structure_get_value (structure, "play-speed");
-  if (value && G_VALUE_HOLDS_DOUBLE (value))
-    rdtdepay->play_speed = g_value_get_double (value);
-  else
-    rdtdepay->play_speed = 1.0;
-
-  value = gst_structure_get_value (structure, "play-scale");
-  if (value && G_VALUE_HOLDS_DOUBLE (value))
-    rdtdepay->play_scale = g_value_get_double (value);
-  else
-    rdtdepay->play_scale = 1.0;
-
-  /* caps seem good, configure element */
-  rdtdepay->clock_rate = clock_rate;
-
-  /* set caps on pad and on header */
-  srccaps = gst_caps_new_empty_simple ("application/vnd.rn-realmedia");
-  gst_pad_set_caps (rdtdepay->srcpad, srccaps);
-  gst_caps_unref (srccaps);
-
-  if (rdtdepay->header)
-    gst_buffer_unref (rdtdepay->header);
-  rdtdepay->header = gst_buffer_ref (header);
-
-  return TRUE;
-
-  /* ERRORS */
-no_header:
-  {
-    GST_ERROR_OBJECT (rdtdepay, "no header found in caps, no 'config' field");
-    return FALSE;
-  }
-}
-
-static gboolean
-gst_rdt_depay_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
-{
-  GstRDTDepay *depay;
-  gboolean res = TRUE;
-
-  depay = GST_RDT_DEPAY (parent);
-
-  switch (GST_EVENT_TYPE (event)) {
-    case GST_EVENT_CAPS:
-    {
-      GstCaps *caps;
-
-      gst_event_parse_caps (event, &caps);
-      res = gst_rdt_depay_setcaps (pad, caps);
-      gst_event_unref (event);
-      break;
-    }
-    case GST_EVENT_FLUSH_STOP:
-      res = gst_pad_push_event (depay->srcpad, event);
-
-      gst_segment_init (&depay->segment, GST_FORMAT_UNDEFINED);
-      depay->need_newsegment = TRUE;
-      depay->next_seqnum = -1;
-      break;
-    case GST_EVENT_SEGMENT:
-    {
-      gst_event_copy_segment (event, &depay->segment);
-      /* don't pass the event downstream, we generate our own segment
-       * including the NTP time and other things we receive in caps */
-      gst_event_unref (event);
-      break;
-    }
-    default:
-      /* pass other events forward */
-      res = gst_pad_push_event (depay->srcpad, event);
-      break;
-  }
-  return res;
-}
-
-static GstEvent *
-create_segment_event (GstRDTDepay * depay, gboolean update,
-    GstClockTime position)
-{
-  GstSegment segment;
-
-  gst_segment_init (&segment, GST_FORMAT_TIME);
-  segment.rate = depay->play_speed;
-  segment.applied_rate = depay->play_scale;
-  segment.start = position;
-
-  if (depay->npt_stop != -1)
-    segment.stop = depay->npt_stop - depay->npt_start;
-  else
-    segment.stop = -1;
-
-  segment.time = position + depay->npt_start;
-
-  return gst_event_new_segment (&segment);
-}
-
-static GstFlowReturn
-gst_rdt_depay_push (GstRDTDepay * rdtdepay, GstBuffer * buffer)
-{
-  GstFlowReturn ret;
-
-  if (rdtdepay->need_newsegment) {
-    GstEvent *event;
-
-    event = create_segment_event (rdtdepay, FALSE, 0);
-    gst_pad_push_event (rdtdepay->srcpad, event);
-
-    rdtdepay->need_newsegment = FALSE;
-  }
-
-  if (rdtdepay->discont) {
-    GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
-    rdtdepay->discont = FALSE;
-  }
-  ret = gst_pad_push (rdtdepay->srcpad, buffer);
-
-  return ret;
-}
-
-static GstFlowReturn
-gst_rdt_depay_handle_data (GstRDTDepay * rdtdepay, GstClockTime outtime,
-    GstRDTPacket * packet)
-{
-  GstFlowReturn ret;
-  GstBuffer *outbuf;
-  GstMapInfo outmap;
-  guint8 *data, *outdata;
-  guint size;
-  guint16 stream_id;
-  guint32 timestamp;
-  gint gap;
-  guint16 seqnum;
-  guint8 flags;
-  guint16 outflags;
-
-  /* get pointers to the packet data */
-  data = gst_rdt_packet_data_map (packet, &size);
-
-  outbuf = gst_buffer_new_and_alloc (12 + size);
-  GST_BUFFER_TIMESTAMP (outbuf) = outtime;
-
-  GST_DEBUG_OBJECT (rdtdepay, "have size %u", size);
-
-  /* copy over some things */
-  stream_id = gst_rdt_packet_data_get_stream_id (packet);
-  timestamp = gst_rdt_packet_data_get_timestamp (packet);
-  flags = gst_rdt_packet_data_get_flags (packet);
-
-  seqnum = gst_rdt_packet_data_get_seq (packet);
-
-  GST_DEBUG_OBJECT (rdtdepay, "stream_id %u, timestamp %u, seqnum %d, flags %d",
-      stream_id, timestamp, seqnum, flags);
-
-  if (rdtdepay->next_seqnum != -1) {
-    gap = gst_rdt_buffer_compare_seqnum (seqnum, rdtdepay->next_seqnum);
-
-    /* if we have no gap, all is fine */
-    if (G_UNLIKELY (gap != 0)) {
-      GST_LOG_OBJECT (rdtdepay, "got packet %u, expected %u, gap %d", seqnum,
-          rdtdepay->next_seqnum, gap);
-      if (gap < 0) {
-        /* seqnum > next_seqnum, we are missing some packets, this is always a
-         * DISCONT. */
-        GST_LOG_OBJECT (rdtdepay, "%d missing packets", gap);
-        rdtdepay->discont = TRUE;
-      } else {
-        /* seqnum < next_seqnum, we have seen this packet before or the sender
-         * could be restarted. If the packet is not too old, we throw it away as
-         * a duplicate, otherwise we mark discont and continue. 100 misordered
-         * packets is a good threshold. See also RFC 4737. */
-        if (gap < 100)
-          goto dropping;
-
-        GST_LOG_OBJECT (rdtdepay,
-            "%d > 100, packet too old, sender likely restarted", gap);
-        rdtdepay->discont = TRUE;
-      }
-    }
-  }
-  rdtdepay->next_seqnum = (seqnum + 1);
-  if (rdtdepay->next_seqnum == 0xff00)
-    rdtdepay->next_seqnum = 0;
-
-  if ((flags & 1) == 0)
-    outflags = 2;
-  else
-    outflags = 0;
-
-  gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE);
-  outdata = outmap.data;
-  GST_WRITE_UINT16_BE (outdata + 0, 0); /* version   */
-  GST_WRITE_UINT16_BE (outdata + 2, size + 12); /* length    */
-  GST_WRITE_UINT16_BE (outdata + 4, stream_id); /* stream    */
-  GST_WRITE_UINT32_BE (outdata + 6, timestamp); /* timestamp */
-  GST_WRITE_UINT16_BE (outdata + 10, outflags); /* flags     */
-  memcpy (outdata + 12, data, size);
-  gst_buffer_unmap (outbuf, &outmap);
-  gst_buffer_resize (outbuf, 0, 12 + size);
-
-  gst_rdt_packet_data_unmap (packet);
-
-  GST_DEBUG_OBJECT (rdtdepay, "Pushing packet, outtime %" GST_TIME_FORMAT,
-      GST_TIME_ARGS (outtime));
-
-  ret = gst_rdt_depay_push (rdtdepay, outbuf);
-
-  return ret;
-
-  /* ERRORS */
-dropping:
-  {
-    GST_WARNING_OBJECT (rdtdepay, "%d <= 100, dropping old packet", gap);
-    return GST_FLOW_OK;
-  }
-}
-
-static GstFlowReturn
-gst_rdt_depay_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
-{
-  GstRDTDepay *rdtdepay;
-  GstFlowReturn ret;
-  GstClockTime timestamp;
-  gboolean more;
-  GstRDTPacket packet;
-
-  rdtdepay = GST_RDT_DEPAY (parent);
-
-  if (GST_BUFFER_IS_DISCONT (buf)) {
-    GST_LOG_OBJECT (rdtdepay, "received discont");
-    rdtdepay->discont = TRUE;
-  }
-
-  if (rdtdepay->header) {
-    GstBuffer *out;
-
-    out = rdtdepay->header;
-    rdtdepay->header = NULL;
-
-    /* push header data first */
-    gst_rdt_depay_push (rdtdepay, out);
-  }
-
-  /* save timestamp */
-  timestamp = GST_BUFFER_TIMESTAMP (buf);
-
-  ret = GST_FLOW_OK;
-
-  GST_LOG_OBJECT (rdtdepay, "received buffer timestamp %" GST_TIME_FORMAT,
-      GST_TIME_ARGS (timestamp));
-
-  /* data is in RDT format. */
-  more = gst_rdt_buffer_get_first_packet (buf, &packet);
-  while (more) {
-    GstRDTType type;
-
-    type = gst_rdt_packet_get_type (&packet);
-    GST_DEBUG_OBJECT (rdtdepay, "Have packet of type %04x", type);
-
-    if (GST_RDT_IS_DATA_TYPE (type)) {
-      GST_DEBUG_OBJECT (rdtdepay, "We have a data packet");
-      ret = gst_rdt_depay_handle_data (rdtdepay, timestamp, &packet);
-    } else {
-      switch (type) {
-        default:
-          GST_DEBUG_OBJECT (rdtdepay, "Ignoring packet");
-          break;
-      }
-    }
-    if (ret != GST_FLOW_OK)
-      break;
-
-    more = gst_rdt_packet_move_to_next (&packet);
-  }
-
-  gst_buffer_unref (buf);
-
-  return ret;
-}
-
-static GstStateChangeReturn
-gst_rdt_depay_change_state (GstElement * element, GstStateChange transition)
-{
-  GstRDTDepay *rdtdepay;
-  GstStateChangeReturn ret;
-
-  rdtdepay = GST_RDT_DEPAY (element);
-
-  switch (transition) {
-    case GST_STATE_CHANGE_NULL_TO_READY:
-      break;
-    case GST_STATE_CHANGE_READY_TO_PAUSED:
-      gst_segment_init (&rdtdepay->segment, GST_FORMAT_UNDEFINED);
-      rdtdepay->next_seqnum = -1;
-      rdtdepay->need_newsegment = TRUE;
-      break;
-    default:
-      break;
-  }
-
-  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
-
-  switch (transition) {
-    case GST_STATE_CHANGE_PAUSED_TO_READY:
-      if (rdtdepay->header)
-        gst_buffer_unref (rdtdepay->header);
-      rdtdepay->header = NULL;
-      break;
-    case GST_STATE_CHANGE_READY_TO_NULL:
-      break;
-    default:
-      break;
-  }
-  return ret;
-}

gst-plugins-ugly-1.22.7.tar.xz/gst/realmedia/rdtdepay.h Deleted

@@ -1,74 +0,0 @@
-/* GStreamer
- * Copyright (C) <2006> Lutz Mueller <lutz at topfrose dot de>
- *               <2006> Wim Taymans <wim@fluendo.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-
-#ifndef __GST_RDT_DEPAY_H__
-#define __GST_RDT_DEPAY_H__
-
-#include <gst/gst.h>
-
-G_BEGIN_DECLS
-
-#define GST_TYPE_RDT_DEPAY \
-  (gst_rdt_depay_get_type())
-#define GST_RDT_DEPAY(obj) \
-  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RDT_DEPAY,GstRDTDepay))
-#define GST_RDT_DEPAY_CLASS(klass) \
-  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RDT_DEPAY,GstRDTDepayClass))
-#define GST_IS_RDT_DEPAY(obj) \
-  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RDT_DEPAY))
-#define GST_IS_RDT_DEPAY_CLASS(klass) \
-  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RDT_DEPAY))
-
-typedef struct _GstRDTDepay GstRDTDepay;
-typedef struct _GstRDTDepayClass GstRDTDepayClass;
-
-struct _GstRDTDepay
-{
-  GstElement parent;
-
-  GstPad *sinkpad;
-  GstPad *srcpad;
-
-  guint clock_rate;
-  GstClockTime npt_start;
-  GstClockTime npt_stop;
-  gdouble play_speed;
-  gdouble play_scale;
-
-  guint32 next_seqnum;
-
-  gboolean discont;
-  gboolean need_newsegment;
-  GstSegment segment;
-  GstBuffer *header;
-};
-
-struct _GstRDTDepayClass
-{
-  GstElementClass parent_class;
-};
-
-GType gst_rdt_depay_get_type (void);
-
-GST_ELEMENT_REGISTER_DECLARE (rdtdepay);
-
-G_END_DECLS
-
-#endif /* __GST_RDT_DEPAY_H__ */

gst-plugins-ugly-1.22.7.tar.xz/gst/realmedia/rdtjitterbuffer.c Deleted

@@ -1,531 +0,0 @@
-/* GStreamer
- * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-#include <string.h>
-#include <stdlib.h>
-
-#include "rdtjitterbuffer.h"
-#include "gstrdtbuffer.h"
-
-GST_DEBUG_CATEGORY_STATIC (rdt_jitter_buffer_debug);
-#define GST_CAT_DEFAULT rdt_jitter_buffer_debug
-
-#define MAX_WINDOW	RDT_JITTER_BUFFER_MAX_WINDOW
-#define MAX_TIME	(2 * GST_SECOND)
-
-/* signals and args */
-enum
-{
-  LAST_SIGNAL
-};
-
-enum
-{
-  PROP_0
-};
-
-/* GObject vmethods */
-static void rdt_jitter_buffer_finalize (GObject * object);
-
-/* static guint rdt_jitter_buffer_signalsLAST_SIGNAL = { 0 }; */
-
-G_DEFINE_TYPE (RDTJitterBuffer, rdt_jitter_buffer, G_TYPE_OBJECT);
-
-static void
-rdt_jitter_buffer_class_init (RDTJitterBufferClass * klass)
-{
-  GObjectClass *gobject_class;
-
-  gobject_class = (GObjectClass *) klass;
-
-  gobject_class->finalize = rdt_jitter_buffer_finalize;
-
-  GST_DEBUG_CATEGORY_INIT (rdt_jitter_buffer_debug, "rdtjitterbuffer", 0,
-      "RDT Jitter Buffer");
-}
-
-static void
-rdt_jitter_buffer_init (RDTJitterBuffer * jbuf)
-{
-  jbuf->packets = g_queue_new ();
-
-  rdt_jitter_buffer_reset_skew (jbuf);
-}
-
-static void
-rdt_jitter_buffer_finalize (GObject * object)
-{
-  RDTJitterBuffer *jbuf;
-
-  jbuf = RDT_JITTER_BUFFER_CAST (object);
-
-  rdt_jitter_buffer_flush (jbuf);
-  g_queue_free (jbuf->packets);
-
-  G_OBJECT_CLASS (rdt_jitter_buffer_parent_class)->finalize (object);
-}
-
-/**
- * rdt_jitter_buffer_new:
- *
- * Create an #RDTJitterBuffer.
- *
- * Returns: a new #RDTJitterBuffer. Use g_object_unref() after usage.
- */
-RDTJitterBuffer *
-rdt_jitter_buffer_new (void)
-{
-  RDTJitterBuffer *jbuf;
-
-  jbuf = g_object_new (RDT_TYPE_JITTER_BUFFER, NULL);
-
-  return jbuf;
-}
-
-void
-rdt_jitter_buffer_reset_skew (RDTJitterBuffer * jbuf)
-{
-  jbuf->base_time = -1;
-  jbuf->base_rtptime = -1;
-  jbuf->ext_rtptime = -1;
-  jbuf->window_pos = 0;
-  jbuf->window_filling = TRUE;
-  jbuf->window_min = 0;
-  jbuf->skew = 0;
-  jbuf->prev_send_diff = -1;
-}
-
-/* For the clock skew we use a windowed low point averaging algorithm as can be
- * found in http://www.grame.fr/pub/TR-050601.pdf. The idea is that the jitter is
- * composed of:
- *
- *  J = N + n
- *
- *   N   : a constant network delay.
- *   n   : random added noise. The noise is concentrated around 0
- *
- * In the receiver we can track the elapsed time at the sender with:
- *
- *  send_diff(i) = (Tsi - Ts0);
- *
- *   Tsi : The time at the sender at packet i
- *   Ts0 : The time at the sender at the first packet
- *
- * This is the difference between the RDT timestamp in the first received packet
- * and the current packet.
- *
- * At the receiver we have to deal with the jitter introduced by the network.
- *
- *  recv_diff(i) = (Tri - Tr0)
- *
- *   Tri : The time at the receiver at packet i
- *   Tr0 : The time at the receiver at the first packet
- *
- * Both of these values contain a jitter Ji, a jitter for packet i, so we can
- * write:
- *
- *  recv_diff(i) = (Cri + D + ni) - (Cr0 + D + n0))
- *
- *    Cri    : The time of the clock at the receiver for packet i
- *    D + ni : The jitter when receiving packet i
- *
- * We see that the network delay is irrelevant here as we can eliminate D:
- *
- *  recv_diff(i) = (Cri + ni) - (Cr0 + n0))
- *
- * The drift is now expressed as:
- *
- *  Drift(i) = recv_diff(i) - send_diff(i);
- *
- * We now keep the W latest values of Drift and find the minimum (this is the
- * one with the lowest network jitter and thus the one which is least affected
- * by it). We average this lowest value to smooth out the resulting network skew.
- *
- * Both the window and the weighting used for averaging influence the accuracy
- * of the drift estimation. Finding the correct parameters turns out to be a
- * compromise between accuracy and inertia. 
- *
- * We use a 2 second window or up to 512 data points, which is statistically big
- * enough to catch spikes (FIXME, detect spikes).
- * We also use a rather large weighting factor (125) to smoothly adapt. During
- * startup, when filling the window, we use a parabolic weighting factor, the
- * more the window is filled, the faster we move to the detected possible skew.
- *
- * Returns: @time adjusted with the clock skew.
- */
-static GstClockTime
-calculate_skew (RDTJitterBuffer * jbuf, guint32 rtptime, GstClockTime time,
-    guint32 clock_rate)
-{
-  guint64 ext_rtptime;
-  guint64 send_diff, recv_diff;
-  gint64 delta;
-  gint64 old;
-  gint pos, i;
-  GstClockTime gstrtptime, out_time;
-
-  //ext_rtptime = gst_rtp_buffer_ext_timestamp (&jbuf->ext_rtptime, rtptime);
-  ext_rtptime = rtptime;
-
-  gstrtptime = gst_util_uint64_scale_int (ext_rtptime, GST_SECOND, clock_rate);
-
-again:
-  /* first time, lock on to time and gstrtptime */
-  if (jbuf->base_time == -1)
-    jbuf->base_time = time;
-  if (jbuf->base_rtptime == -1)
-    jbuf->base_rtptime = gstrtptime;
-
-  if (gstrtptime >= jbuf->base_rtptime)
-    send_diff = gstrtptime - jbuf->base_rtptime;
-  else {
-    /* elapsed time at sender, timestamps can go backwards and thus be smaller
-     * than our base time, take a new base time in that case. */
-    GST_DEBUG ("backward timestamps at server, taking new base time");
-    jbuf->base_rtptime = gstrtptime;
-    jbuf->base_time = time;
-    send_diff = 0;
-  }
-
-  GST_DEBUG ("extrtp %" G_GUINT64_FORMAT ", gstrtp %" GST_TIME_FORMAT ", base %"
-      GST_TIME_FORMAT ", send_diff %" GST_TIME_FORMAT, ext_rtptime,
-      GST_TIME_ARGS (gstrtptime), GST_TIME_ARGS (jbuf->base_rtptime),
-      GST_TIME_ARGS (send_diff));
-
-  if (jbuf->prev_send_diff != -1 && time != -1) {
-    gint64 delta_diff;
-
-    if (send_diff > jbuf->prev_send_diff)
-      delta_diff = send_diff - jbuf->prev_send_diff;
-    else
-      delta_diff = jbuf->prev_send_diff - send_diff;
-
-    /* server changed rtp timestamps too quickly, reset skew detection and start
-     * again. This value is sortof arbitrary and can be a bad measurement up if
-     * there are many packets missing because then we get a big gap that is
-     * unrelated to a timestamp switch. */
-    if (delta_diff > GST_SECOND) {
-      GST_DEBUG ("delta changed too quickly %" GST_TIME_FORMAT " reset skew",
-          GST_TIME_ARGS (delta_diff));
-      rdt_jitter_buffer_reset_skew (jbuf);
-      goto again;
-    }
-  }
-  jbuf->prev_send_diff = send_diff;
-
-  /* we don't have an arrival timestamp so we can't do skew detection. we
-   * should still apply a timestamp based on RDT timestamp and base_time */
-  if (time == -1)
-    goto no_skew;
-
-  /* elapsed time at receiver, includes the jitter */
-  recv_diff = time - jbuf->base_time;
-
-  GST_DEBUG ("time %" GST_TIME_FORMAT ", base %" GST_TIME_FORMAT ", recv_diff %"
-      GST_TIME_FORMAT, GST_TIME_ARGS (time), GST_TIME_ARGS (jbuf->base_time),
-      GST_TIME_ARGS (recv_diff));
-
-  /* measure the diff */
-  delta = ((gint64) recv_diff) - ((gint64) send_diff);
-
-  pos = jbuf->window_pos;
-
-  if (jbuf->window_filling) {
-    /* we are filling the window */
-    GST_DEBUG ("filling %d, delta %" G_GINT64_FORMAT, pos, delta);
-    jbuf->windowpos++ = delta;
-    /* calc the min delta we observed */
-    if (pos == 1 || delta < jbuf->window_min)
-      jbuf->window_min = delta;
-
-    if (send_diff >= MAX_TIME || pos >= MAX_WINDOW) {
-      jbuf->window_size = pos;
-
-      /* window filled */
-      GST_DEBUG ("min %" G_GINT64_FORMAT, jbuf->window_min);
-
-      /* the skew is now the min */
-      jbuf->skew = jbuf->window_min;
-      jbuf->window_filling = FALSE;
-    } else {
-      gint perc_time, perc_window, perc;
-
-      /* figure out how much we filled the window, this depends on the amount of
-       * time we have or the max number of points we keep. */
-      perc_time = send_diff * 100 / MAX_TIME;
-      perc_window = pos * 100 / MAX_WINDOW;
-      perc = MAX (perc_time, perc_window);
-
-      /* make a parabolic function, the closer we get to the MAX, the more value
-       * we give to the scaling factor of the new value */
-      perc = perc * perc;
-
-      /* quickly go to the min value when we are filling up, slowly when we are
-       * just starting because we're not sure it's a good value yet. */
-      jbuf->skew =
-          (perc * jbuf->window_min + ((10000 - perc) * jbuf->skew)) / 10000;
-      jbuf->window_size = pos + 1;
-    }
-  } else {
-    /* pick old value and store new value. We keep the previous value in order
-     * to quickly check if the min of the window changed */
-    old = jbuf->windowpos;
-    jbuf->windowpos++ = delta;
-
-    if (delta <= jbuf->window_min) {
-      /* if the new value we inserted is smaller or equal to the current min,
-       * it becomes the new min */
-      jbuf->window_min = delta;
-    } else if (old == jbuf->window_min) {
-      gint64 min = G_MAXINT64;
-
-      /* if we removed the old min, we have to find a new min */
-      for (i = 0; i < jbuf->window_size; i++) {
-        /* we found another value equal to the old min, we can stop searching now */
-        if (jbuf->windowi == old) {
-          min = old;
-          break;
-        }
-        if (jbuf->windowi < min)
-          min = jbuf->windowi;
-      }
-      jbuf->window_min = min;
-    }
-    /* average the min values */
-    jbuf->skew = (jbuf->window_min + (124 * jbuf->skew)) / 125;
-    GST_DEBUG ("delta %" G_GINT64_FORMAT ", new min: %" G_GINT64_FORMAT,
-        delta, jbuf->window_min);
-  }
-  /* wrap around in the window */
-  if (pos >= jbuf->window_size)
-    pos = 0;
-  jbuf->window_pos = pos;
-
-no_skew:
-  /* the output time is defined as the base timestamp plus the RDT time
-   * adjusted for the clock skew .*/
-  out_time = jbuf->base_time + send_diff + jbuf->skew;
-
-  GST_DEBUG ("skew %" G_GINT64_FORMAT ", out %" GST_TIME_FORMAT,
-      jbuf->skew, GST_TIME_ARGS (out_time));
-
-  return out_time;
-}
-
-/**
- * rdt_jitter_buffer_insert:
- * @jbuf: an #RDTJitterBuffer
- * @buf: a buffer
- * @time: a running_time when this buffer was received in nanoseconds
- * @clock_rate: the clock-rate of the payload of @buf
- * @tail: TRUE when the tail element changed.
- *
- * Inserts @buf into the packet queue of @jbuf. The sequence number of the
- * packet will be used to sort the packets. This function takes ownerhip of
- * @buf when the function returns %TRUE.
- * @buf should have writable metadata when calling this function.
- *
- * Returns: %FALSE if a packet with the same number already existed.
- */
-gboolean
-rdt_jitter_buffer_insert (RDTJitterBuffer * jbuf, GstBuffer * buf,
-    GstClockTime time, guint32 clock_rate, gboolean * tail)
-{
-  GList *list;
-  guint32 rtptime;
-  guint16 seqnum;
-  GstRDTPacket packet;
-  gboolean more;
-
-  g_return_val_if_fail (jbuf != NULL, FALSE);
-  g_return_val_if_fail (buf != NULL, FALSE);
-
-  more = gst_rdt_buffer_get_first_packet (buf, &packet);
-  /* programmer error */
-  g_return_val_if_fail (more == TRUE, FALSE);
-
-  seqnum = gst_rdt_packet_data_get_seq (&packet);
-  /* do skew calculation by measuring the difference between rtptime and the
-   * receive time, this function will retimestamp @buf with the skew corrected
-   * running time. */
-  rtptime = gst_rdt_packet_data_get_timestamp (&packet);
-
-  /* loop the list to skip strictly smaller seqnum buffers */
-  for (list = jbuf->packets->head; list; list = g_list_next (list)) {
-    guint16 qseq;
-    gint gap;
-
-    more =
-        gst_rdt_buffer_get_first_packet (GST_BUFFER_CAST (list->data), &packet);
-    /* programmer error */
-    g_return_val_if_fail (more == TRUE, FALSE);
-
-    qseq = gst_rdt_packet_data_get_seq (&packet);
-
-    /* compare the new seqnum to the one in the buffer */
-    gap = gst_rdt_buffer_compare_seqnum (seqnum, qseq);
-
-    /* we hit a packet with the same seqnum, notify a duplicate */
-    if (G_UNLIKELY (gap == 0))
-      goto duplicate;
-
-    /* seqnum > qseq, we can stop looking */
-    if (G_LIKELY (gap < 0))
-      break;
-  }
-
-
-  if (clock_rate) {
-    time = calculate_skew (jbuf, rtptime, time, clock_rate);
-    GST_BUFFER_TIMESTAMP (buf) = time;
-  }
-
-  if (list)
-    g_queue_insert_before (jbuf->packets, list, buf);
-  else
-    g_queue_push_tail (jbuf->packets, buf);
-
-  /* tail was changed when we did not find a previous packet, we set the return
-   * flag when requested. */
-  if (tail)
-    *tail = (list == NULL);
-
-  return TRUE;
-
-  /* ERRORS */
-duplicate:
-  {
-    GST_WARNING ("duplicate packet %d found", (gint) seqnum);
-    return FALSE;
-  }
-}
-
-/**
- * rdt_jitter_buffer_pop:
- * @jbuf: an #RDTJitterBuffer
- *
- * Pops the oldest buffer from the packet queue of @jbuf. The popped buffer will
- * have its timestamp adjusted with the incoming running_time and the detected
- * clock skew.
- *
- * Returns: a #GstBuffer or %NULL when there was no packet in the queue.
- */
-GstBuffer *
-rdt_jitter_buffer_pop (RDTJitterBuffer * jbuf)
-{
-  GstBuffer *buf;
-
-  g_return_val_if_fail (jbuf != NULL, FALSE);
-
-  buf = g_queue_pop_tail (jbuf->packets);
-
-  return buf;
-}
-
-/**
- * rdt_jitter_buffer_peek:
- * @jbuf: an #RDTJitterBuffer
- *
- * Peek the oldest buffer from the packet queue of @jbuf. Register a callback
- * with rdt_jitter_buffer_set_tail_changed() to be notified when an older packet
- * was inserted in the queue.
- *
- * Returns: a #GstBuffer or %NULL when there was no packet in the queue.
- */
-GstBuffer *
-rdt_jitter_buffer_peek (RDTJitterBuffer * jbuf)
-{
-  GstBuffer *buf;
-
-  g_return_val_if_fail (jbuf != NULL, FALSE);
-
-  buf = g_queue_peek_tail (jbuf->packets);
-
-  return buf;
-}
-
-/**
- * rdt_jitter_buffer_flush:
- * @jbuf: an #RDTJitterBuffer
- *
- * Flush all packets from the jitterbuffer.
- */
-void
-rdt_jitter_buffer_flush (RDTJitterBuffer * jbuf)
-{
-  GstBuffer *buffer;
-
-  g_return_if_fail (jbuf != NULL);
-
-  while ((buffer = g_queue_pop_head (jbuf->packets)))
-    gst_buffer_unref (buffer);
-}
-
-/**
- * rdt_jitter_buffer_num_packets:
- * @jbuf: an #RDTJitterBuffer
- *
- * Get the number of packets currently in "jbuf.
- *
- * Returns: The number of packets in @jbuf.
- */
-guint
-rdt_jitter_buffer_num_packets (RDTJitterBuffer * jbuf)
-{
-  g_return_val_if_fail (jbuf != NULL, 0);
-
-  return jbuf->packets->length;
-}
-
-/**
- * rdt_jitter_buffer_get_ts_diff:
- * @jbuf: an #RDTJitterBuffer
- *
- * Get the difference between the timestamps of first and last packet in the
- * jitterbuffer.
- *
- * Returns: The difference expressed in the timestamp units of the packets.
- */
-guint32
-rdt_jitter_buffer_get_ts_diff (RDTJitterBuffer * jbuf)
-{
-  guint64 high_ts, low_ts;
-  GstBuffer *high_buf, *low_buf;
-  guint32 result;
-
-  g_return_val_if_fail (jbuf != NULL, 0);
-
-  high_buf = g_queue_peek_head (jbuf->packets);
-  low_buf = g_queue_peek_tail (jbuf->packets);
-
-  if (!high_buf || !low_buf || high_buf == low_buf)
-    return 0;
-
-  //high_ts = gst_rtp_buffer_get_timestamp (high_buf);
-  //low_ts = gst_rtp_buffer_get_timestamp (low_buf);
-  high_ts = 0;
-  low_ts = 0;
-
-  /* it needs to work if ts wraps */
-  if (high_ts >= low_ts) {
-    result = (guint32) (high_ts - low_ts);
-  } else {
-    result = (guint32) (high_ts + G_MAXUINT32 + 1 - low_ts);
-  }
-  return result;
-}

gst-plugins-ugly-1.22.7.tar.xz/gst/realmedia/rdtjitterbuffer.h Deleted

@@ -1,91 +0,0 @@
-/* GStreamer
- * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-
-#ifndef __RDT_JITTER_BUFFER_H__
-#define __RDT_JITTER_BUFFER_H__
-
-#include <gst/gst.h>
-
-typedef struct _RDTJitterBuffer RDTJitterBuffer;
-typedef struct _RDTJitterBufferClass RDTJitterBufferClass;
-
-#define RDT_TYPE_JITTER_BUFFER             (rdt_jitter_buffer_get_type())
-#define RDT_JITTER_BUFFER(src)             (G_TYPE_CHECK_INSTANCE_CAST((src),RDT_TYPE_JITTER_BUFFER,RDTJitterBuffer))
-#define RDT_JITTER_BUFFER_CLASS(klass)     (G_TYPE_CHECK_CLASS_CAST((klass),RDT_TYPE_JITTER_BUFFER,RDTJitterBufferClass))
-#define RDT_IS_JITTER_BUFFER(src)          (G_TYPE_CHECK_INSTANCE_TYPE((src),RDT_TYPE_JITTER_BUFFER))
-#define RDT_IS_JITTER_BUFFER_CLASS(klass)  (G_TYPE_CHECK_CLASS_TYPE((klass),RDT_TYPE_JITTER_BUFFER))
-#define RDT_JITTER_BUFFER_CAST(src)        ((RDTJitterBuffer *)(src))
-
-/**
- * RTPTailChanged:
- * @jbuf: an #RDTJitterBuffer
- * @user_data: user data specified when registering
- *
- * This callback will be called when the tail buffer of @jbuf changed.
- */
-typedef void (*RTPTailChanged) (RDTJitterBuffer *jbuf, gpointer user_data);
-
-#define RDT_JITTER_BUFFER_MAX_WINDOW 512
-/**
- * RDTJitterBuffer:
- *
- * A JitterBuffer in the #RTPSession
- */
-struct _RDTJitterBuffer {
-  GObject        object;
-
-  GQueue        *packets;
-
-  /* for calculating skew */
-  GstClockTime   base_time;
-  GstClockTime   base_rtptime;
-  guint64        ext_rtptime;
-  gint64         windowRDT_JITTER_BUFFER_MAX_WINDOW;
-  guint          window_pos;
-  guint          window_size;
-  gboolean       window_filling;
-  gint64         window_min;
-  gint64         skew;
-  gint64         prev_send_diff;
-};
-
-struct _RDTJitterBufferClass {
-  GObjectClass   parent_class;
-};
-
-GType rdt_jitter_buffer_get_type (void);
-
-/* managing lifetime */
-RDTJitterBuffer*      rdt_jitter_buffer_new              (void);
-
-void                  rdt_jitter_buffer_reset_skew       (RDTJitterBuffer *jbuf);
-
-gboolean              rdt_jitter_buffer_insert           (RDTJitterBuffer *jbuf, GstBuffer *buf,
-		                                          GstClockTime time,
-		                                          guint32 clock_rate,
-		                                          gboolean *tail);
-GstBuffer *           rdt_jitter_buffer_peek             (RDTJitterBuffer *jbuf);
-GstBuffer *           rdt_jitter_buffer_pop              (RDTJitterBuffer *jbuf);
-
-void                  rdt_jitter_buffer_flush            (RDTJitterBuffer *jbuf);
-
-guint                 rdt_jitter_buffer_num_packets      (RDTJitterBuffer *jbuf);
-guint32               rdt_jitter_buffer_get_ts_diff      (RDTJitterBuffer *jbuf);
-
-#endif /* __RDT_JITTER_BUFFER_H__ */

gst-plugins-ugly-1.22.7.tar.xz/gst/realmedia/rdtmanager.c Deleted

@@ -1,1371 +0,0 @@
-/* GStreamer
- * Copyright (C) <2005,2006> Wim Taymans <wim@fluendo.com>
- *               <2013> Wim Taymans <wim.taymans@gmail.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-/*
- * Unless otherwise indicated, Source Code is licensed under MIT license.
- * See further explanation attached in License Statement (distributed in the file
- * LICENSE).
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy of
- * this software and associated documentation files (the "Software"), to deal in
- * the Software without restriction, including without limitation the rights to
- * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
- * of the Software, and to permit persons to whom the Software is furnished to do
- * so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in all
- * copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
- * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
- * SOFTWARE.
- */
-/* Element-Checklist-Version: 5 */
-
-/**
- * SECTION:element-rdtmanager
- * @title: rdtmanager
- * @see_also: GstRtspSrc
- *
- * A simple RTP session manager used internally by rtspsrc.
- */
-
-/* #define HAVE_RTCP */
-
-#include "gstrdtbuffer.h"
-#include "rdtmanager.h"
-#include "rdtjitterbuffer.h"
-
-#include <gst/glib-compat-private.h>
-
-#include <stdio.h>
-
-GST_DEBUG_CATEGORY_STATIC (rdtmanager_debug);
-#define GST_CAT_DEFAULT (rdtmanager_debug)
-
-/* GstRDTManager signals and args */
-enum
-{
-  SIGNAL_REQUEST_PT_MAP,
-  SIGNAL_CLEAR_PT_MAP,
-
-  SIGNAL_ON_NEW_SSRC,
-  SIGNAL_ON_SSRC_COLLISION,
-  SIGNAL_ON_SSRC_VALIDATED,
-  SIGNAL_ON_SSRC_ACTIVE,
-  SIGNAL_ON_SSRC_SDES,
-  SIGNAL_ON_BYE_SSRC,
-  SIGNAL_ON_BYE_TIMEOUT,
-  SIGNAL_ON_TIMEOUT,
-  SIGNAL_ON_NPT_STOP,
-  LAST_SIGNAL
-};
-
-#define DEFAULT_LATENCY_MS      200
-
-enum
-{
-  PROP_0,
-  PROP_LATENCY
-};
-
-static GstStaticPadTemplate gst_rdt_manager_recv_rtp_sink_template =
-GST_STATIC_PAD_TEMPLATE ("recv_rtp_sink_%u",
-    GST_PAD_SINK,
-    GST_PAD_REQUEST,
-    GST_STATIC_CAPS ("application/x-rdt")
-    );
-
-static GstStaticPadTemplate gst_rdt_manager_recv_rtcp_sink_template =
-GST_STATIC_PAD_TEMPLATE ("recv_rtcp_sink_%u",
-    GST_PAD_SINK,
-    GST_PAD_REQUEST,
-    GST_STATIC_CAPS ("application/x-rtcp")
-    );
-
-static GstStaticPadTemplate gst_rdt_manager_recv_rtp_src_template =
-GST_STATIC_PAD_TEMPLATE ("recv_rtp_src_%u_%u_%u",
-    GST_PAD_SRC,
-    GST_PAD_SOMETIMES,
-    GST_STATIC_CAPS ("application/x-rdt")
-    );
-
-static GstStaticPadTemplate gst_rdt_manager_rtcp_src_template =
-GST_STATIC_PAD_TEMPLATE ("rtcp_src_%u",
-    GST_PAD_SRC,
-    GST_PAD_REQUEST,
-    GST_STATIC_CAPS ("application/x-rtcp")
-    );
-
-static void gst_rdt_manager_finalize (GObject * object);
-static void gst_rdt_manager_set_property (GObject * object,
-    guint prop_id, const GValue * value, GParamSpec * pspec);
-static void gst_rdt_manager_get_property (GObject * object,
-    guint prop_id, GValue * value, GParamSpec * pspec);
-
-static gboolean gst_rdt_manager_query_src (GstPad * pad, GstObject * parent,
-    GstQuery * query);
-static gboolean gst_rdt_manager_src_activate_mode (GstPad * pad,
-    GstObject * parent, GstPadMode mode, gboolean active);
-
-static GstClock *gst_rdt_manager_provide_clock (GstElement * element);
-static GstStateChangeReturn gst_rdt_manager_change_state (GstElement * element,
-    GstStateChange transition);
-static GstPad *gst_rdt_manager_request_new_pad (GstElement * element,
-    GstPadTemplate * templ, const gchar * name, const GstCaps * caps);
-static void gst_rdt_manager_release_pad (GstElement * element, GstPad * pad);
-
-static gboolean gst_rdt_manager_parse_caps (GstRDTManager * rdtmanager,
-    GstRDTManagerSession * session, GstCaps * caps);
-static gboolean gst_rdt_manager_event_rdt (GstPad * pad, GstObject * parent,
-    GstEvent * event);
-
-static GstFlowReturn gst_rdt_manager_chain_rdt (GstPad * pad,
-    GstObject * parent, GstBuffer * buffer);
-static GstFlowReturn gst_rdt_manager_chain_rtcp (GstPad * pad,
-    GstObject * parent, GstBuffer * buffer);
-static void gst_rdt_manager_loop (GstPad * pad);
-
-static guint gst_rdt_manager_signalsLAST_SIGNAL = { 0 };
-
-#define JBUF_LOCK(sess)   (g_mutex_lock (&(sess)->jbuf_lock))
-
-#define JBUF_LOCK_CHECK(sess,label) G_STMT_START {    \
-  JBUF_LOCK (sess);                                   \
-  if (sess->srcresult != GST_FLOW_OK)                 \
-    goto label;                                       \
-} G_STMT_END
-
-#define JBUF_UNLOCK(sess) (g_mutex_unlock (&(sess)->jbuf_lock))
-#define JBUF_WAIT(sess)   (g_cond_wait (&(sess)->jbuf_cond, &(sess)->jbuf_lock))
-
-#define JBUF_WAIT_CHECK(sess,label) G_STMT_START {    \
-  JBUF_WAIT(sess);                                    \
-  if (sess->srcresult != GST_FLOW_OK)                 \
-    goto label;                                       \
-} G_STMT_END
-
-#define JBUF_SIGNAL(sess) (g_cond_signal (&(sess)->jbuf_cond))
-
-/* Manages the receiving end of the packets.
- *
- * There is one such structure for each RTP session (audio/video/...).
- * We get the RTP/RTCP packets and stuff them into the session manager. 
- */
-struct _GstRDTManagerSession
-{
-  /* session id */
-  gint id;
-  /* the parent bin */
-  GstRDTManager *dec;
-
-  gboolean active;
-  /* we only support one ssrc and one pt */
-  guint32 ssrc;
-  guint8 pt;
-  gint clock_rate;
-  GstCaps *caps;
-  gint64 clock_base;
-
-  GstSegment segment;
-
-  /* the last seqnum we pushed out */
-  guint32 last_popped_seqnum;
-  /* the next expected seqnum */
-  guint32 next_seqnum;
-  /* last output time */
-  GstClockTime last_out_time;
-
-  /* the pads of the session */
-  GstPad *recv_rtp_sink;
-  GstPad *recv_rtp_src;
-  GstPad *recv_rtcp_sink;
-  GstPad *rtcp_src;
-
-  GstFlowReturn srcresult;
-  gboolean blocked;
-  gboolean eos;
-  gboolean waiting;
-  gboolean discont;
-  GstClockID clock_id;
-
-  /* jitterbuffer, lock and cond */
-  RDTJitterBuffer *jbuf;
-  GMutex jbuf_lock;
-  GCond jbuf_cond;
-
-  /* some accounting */
-  guint64 num_late;
-  guint64 num_duplicates;
-};
-
-/* find a session with the given id */
-static GstRDTManagerSession *
-find_session_by_id (GstRDTManager * rdtmanager, gint id)
-{
-  GSList *walk;
-
-  for (walk = rdtmanager->sessions; walk; walk = g_slist_next (walk)) {
-    GstRDTManagerSession *sess = (GstRDTManagerSession *) walk->data;
-
-    if (sess->id == id)
-      return sess;
-  }
-  return NULL;
-}
-
-/* create a session with the given id */
-static GstRDTManagerSession *
-create_session (GstRDTManager * rdtmanager, gint id)
-{
-  GstRDTManagerSession *sess;
-
-  sess = g_new0 (GstRDTManagerSession, 1);
-  sess->id = id;
-  sess->dec = rdtmanager;
-  sess->jbuf = rdt_jitter_buffer_new ();
-  g_mutex_init (&sess->jbuf_lock);
-  g_cond_init (&sess->jbuf_cond);
-  rdtmanager->sessions = g_slist_prepend (rdtmanager->sessions, sess);
-
-  return sess;
-}
-
-static gboolean
-forward_sticky_events (GstPad * pad, GstEvent ** event, gpointer user_data)
-{
-  GstPad *srcpad = GST_PAD_CAST (user_data);
-
-  gst_pad_push_event (srcpad, gst_event_ref (*event));
-
-  return TRUE;
-}
-
-static gboolean
-activate_session (GstRDTManager * rdtmanager, GstRDTManagerSession * session,
-    guint32 ssrc, guint8 pt)
-{
-  GstPadTemplate *templ;
-  GstElementClass *klass;
-  gchar *name;
-  GstCaps *caps;
-  GValue ret = { 0 };
-  GValue args3 = { {0}
-  , {0}
-  , {0}
-  };
-
-  GST_DEBUG_OBJECT (rdtmanager, "creating stream");
-
-  session->ssrc = ssrc;
-  session->pt = pt;
-
-  /* get pt map */
-  g_value_init (&args0, GST_TYPE_ELEMENT);
-  g_value_set_object (&args0, rdtmanager);
-  g_value_init (&args1, G_TYPE_UINT);
-  g_value_set_uint (&args1, session->id);
-  g_value_init (&args2, G_TYPE_UINT);
-  g_value_set_uint (&args2, pt);
-
-  g_value_init (&ret, GST_TYPE_CAPS);
-  g_value_set_boxed (&ret, NULL);
-
-  g_signal_emitv (args, gst_rdt_manager_signalsSIGNAL_REQUEST_PT_MAP, 0,
-      &ret);
-
-  g_value_unset (&args0);
-  g_value_unset (&args1);
-  g_value_unset (&args2);
-  caps = (GstCaps *) g_value_dup_boxed (&ret);
-  g_value_unset (&ret);
-
-  if (caps)
-    gst_rdt_manager_parse_caps (rdtmanager, session, caps);
-
-  name = g_strdup_printf ("recv_rtp_src_%u_%u_%u", session->id, ssrc, pt);
-  klass = GST_ELEMENT_GET_CLASS (rdtmanager);
-  templ = gst_element_class_get_pad_template (klass, "recv_rtp_src_%u_%u_%u");
-  session->recv_rtp_src = gst_pad_new_from_template (templ, name);
-  g_free (name);
-
-  gst_pad_set_element_private (session->recv_rtp_src, session);
-  gst_pad_set_query_function (session->recv_rtp_src, gst_rdt_manager_query_src);
-  gst_pad_set_activatemode_function (session->recv_rtp_src,
-      gst_rdt_manager_src_activate_mode);
-
-  gst_pad_set_active (session->recv_rtp_src, TRUE);
-
-  gst_pad_sticky_events_foreach (session->recv_rtp_sink, forward_sticky_events,
-      session->recv_rtp_src);
-  gst_pad_set_caps (session->recv_rtp_src, caps);
-  gst_caps_unref (caps);
-
-  gst_element_add_pad (GST_ELEMENT_CAST (rdtmanager), session->recv_rtp_src);
-
-  return TRUE;
-}
-
-static void
-free_session (GstRDTManagerSession * session)
-{
-  g_object_unref (session->jbuf);
-  g_cond_clear (&session->jbuf_cond);
-  g_mutex_clear (&session->jbuf_lock);
-  g_free (session);
-}
-
-#define gst_rdt_manager_parent_class parent_class
-G_DEFINE_TYPE (GstRDTManager, gst_rdt_manager, GST_TYPE_ELEMENT);
-GST_ELEMENT_REGISTER_DEFINE (rdtmanager, "rdtmanager",
-    GST_RANK_NONE, GST_TYPE_RDT_MANAGER);
-
-/* BOXED:UINT,UINT */
-#define g_marshal_value_peek_uint(v)     g_value_get_uint (v)
-
-static void
-gst_rdt_manager_marshal_BOXED__UINT_UINT (GClosure * closure,
-    GValue * return_value,
-    guint n_param_values,
-    const GValue * param_values,
-    gpointer invocation_hint, gpointer marshal_data)
-{
-  typedef gpointer (*GMarshalFunc_BOXED__UINT_UINT) (gpointer data1,
-      guint arg_1, guint arg_2, gpointer data2);
-  register GMarshalFunc_BOXED__UINT_UINT callback;
-  register GCClosure *cc = (GCClosure *) closure;
-  register gpointer data1, data2;
-  gpointer v_return;
-
-  g_return_if_fail (return_value != NULL);
-  g_return_if_fail (n_param_values == 3);
-
-  if (G_CCLOSURE_SWAP_DATA (closure)) {
-    data1 = closure->data;
-    data2 = g_value_peek_pointer (param_values + 0);
-  } else {
-    data1 = g_value_peek_pointer (param_values + 0);
-    data2 = closure->data;
-  }
-  callback =
-      (GMarshalFunc_BOXED__UINT_UINT) (marshal_data ? marshal_data :
-      cc->callback);
-
-  v_return = callback (data1,
-      g_marshal_value_peek_uint (param_values + 1),
-      g_marshal_value_peek_uint (param_values + 2), data2);
-
-  g_value_take_boxed (return_value, v_return);
-}
-
-static void
-gst_rdt_manager_marshal_VOID__UINT_UINT (GClosure * closure,
-    GValue * return_value,
-    guint n_param_values,
-    const GValue * param_values,
-    gpointer invocation_hint, gpointer marshal_data)
-{
-  typedef void (*GMarshalFunc_VOID__UINT_UINT) (gpointer data1,
-      guint arg_1, guint arg_2, gpointer data2);
-  register GMarshalFunc_VOID__UINT_UINT callback;
-  register GCClosure *cc = (GCClosure *) closure;
-  register gpointer data1, data2;
-
-  g_return_if_fail (n_param_values == 3);
-
-  if (G_CCLOSURE_SWAP_DATA (closure)) {
-    data1 = closure->data;
-    data2 = g_value_peek_pointer (param_values + 0);
-  } else {
-    data1 = g_value_peek_pointer (param_values + 0);
-    data2 = closure->data;
-  }
-  callback =
-      (GMarshalFunc_VOID__UINT_UINT) (marshal_data ? marshal_data :
-      cc->callback);
-
-  callback (data1,
-      g_marshal_value_peek_uint (param_values + 1),
-      g_marshal_value_peek_uint (param_values + 2), data2);
-}
-
-static void
-gst_rdt_manager_class_init (GstRDTManagerClass * g_class)
-{
-  GObjectClass *gobject_class;
-  GstElementClass *gstelement_class;
-  GstRDTManagerClass *klass;
-
-  klass = (GstRDTManagerClass *) g_class;
-  gobject_class = (GObjectClass *) klass;
-  gstelement_class = (GstElementClass *) klass;
-
-  gobject_class->finalize = gst_rdt_manager_finalize;
-  gobject_class->set_property = gst_rdt_manager_set_property;
-  gobject_class->get_property = gst_rdt_manager_get_property;
-
-  g_object_class_install_property (gobject_class, PROP_LATENCY,
-      g_param_spec_uint ("latency", "Buffer latency in ms",
-          "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
-          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
-
-  /**
-   * GstRDTManager::request-pt-map:
-   * @rdtmanager: the object which received the signal
-   * @session: the session
-   * @pt: the pt
-   *
-   * Request the payload type as #GstCaps for @pt in @session.
-   */
-  gst_rdt_manager_signalsSIGNAL_REQUEST_PT_MAP =
-      g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRDTManagerClass, request_pt_map),
-      NULL, NULL, gst_rdt_manager_marshal_BOXED__UINT_UINT, GST_TYPE_CAPS, 2,
-      G_TYPE_UINT, G_TYPE_UINT);
-
-  /**
-   * GstRDTManager::clear-pt-map:
-   * @rtpbin: the object which received the signal
-   *
-   * Clear all previously cached pt-mapping obtained with
-   * GstRDTManager::request-pt-map.
-   */
-  gst_rdt_manager_signalsSIGNAL_CLEAR_PT_MAP =
-      g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRDTManagerClass, clear_pt_map),
-      NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
-
-  /**
-   * GstRDTManager::on-bye-ssrc:
-   * @rtpbin: the object which received the signal
-   * @session: the session
-   * @ssrc: the SSRC
-   *
-   * Notify of an SSRC that became inactive because of a BYE packet.
-   */
-  gst_rdt_manager_signalsSIGNAL_ON_BYE_SSRC =
-      g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRDTManagerClass, on_bye_ssrc),
-      NULL, NULL, gst_rdt_manager_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
-      G_TYPE_UINT, G_TYPE_UINT);
-  /**
-   * GstRDTManager::on-bye-timeout:
-   * @rtpbin: the object which received the signal
-   * @session: the session
-   * @ssrc: the SSRC
-   *
-   * Notify of an SSRC that has timed out because of BYE
-   */
-  gst_rdt_manager_signalsSIGNAL_ON_BYE_TIMEOUT =
-      g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRDTManagerClass, on_bye_timeout),
-      NULL, NULL, gst_rdt_manager_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
-      G_TYPE_UINT, G_TYPE_UINT);
-  /**
-   * GstRDTManager::on-timeout:
-   * @rtpbin: the object which received the signal
-   * @session: the session
-   * @ssrc: the SSRC
-   *
-   * Notify of an SSRC that has timed out
-   */
-  gst_rdt_manager_signalsSIGNAL_ON_TIMEOUT =
-      g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRDTManagerClass, on_timeout),
-      NULL, NULL, gst_rdt_manager_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
-      G_TYPE_UINT, G_TYPE_UINT);
-
-  /**
-   * GstRDTManager::on-npt-stop:
-   * @rtpbin: the object which received the signal
-   * @session: the session
-   * @ssrc: the SSRC
-   *
-   * Notify that SSRC sender has sent data up to the configured NPT stop time.
-   */
-  gst_rdt_manager_signalsSIGNAL_ON_NPT_STOP =
-      g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRDTManagerClass, on_npt_stop),
-      NULL, NULL, gst_rdt_manager_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
-      G_TYPE_UINT, G_TYPE_UINT);
-
-
-  gstelement_class->provide_clock =
-      GST_DEBUG_FUNCPTR (gst_rdt_manager_provide_clock);
-  gstelement_class->change_state =
-      GST_DEBUG_FUNCPTR (gst_rdt_manager_change_state);
-  gstelement_class->request_new_pad =
-      GST_DEBUG_FUNCPTR (gst_rdt_manager_request_new_pad);
-  gstelement_class->release_pad =
-      GST_DEBUG_FUNCPTR (gst_rdt_manager_release_pad);
-
-  /* sink pads */
-  gst_element_class_add_static_pad_template (gstelement_class,
-      &gst_rdt_manager_recv_rtp_sink_template);
-  gst_element_class_add_static_pad_template (gstelement_class,
-      &gst_rdt_manager_recv_rtcp_sink_template);
-  /* src pads */
-  gst_element_class_add_static_pad_template (gstelement_class,
-      &gst_rdt_manager_recv_rtp_src_template);
-  gst_element_class_add_static_pad_template (gstelement_class,
-      &gst_rdt_manager_rtcp_src_template);
-
-  gst_element_class_set_static_metadata (gstelement_class, "RTP Decoder",
-      "Codec/Parser/Network",
-      "Accepts raw RTP and RTCP packets and sends them forward",
-      "Wim Taymans <wim.taymans@gmail.com>");
-
-  GST_DEBUG_CATEGORY_INIT (rdtmanager_debug, "rdtmanager", 0, "RTP decoder");
-}
-
-static void
-gst_rdt_manager_init (GstRDTManager * rdtmanager)
-{
-  rdtmanager->provided_clock = gst_system_clock_obtain ();
-  rdtmanager->latency = DEFAULT_LATENCY_MS;
-  GST_OBJECT_FLAG_SET (rdtmanager, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
-}
-
-static void
-gst_rdt_manager_finalize (GObject * object)
-{
-  GstRDTManager *rdtmanager;
-
-  rdtmanager = GST_RDT_MANAGER (object);
-
-  g_slist_foreach (rdtmanager->sessions, (GFunc) free_session, NULL);
-  g_slist_free (rdtmanager->sessions);
-  g_clear_object (&rdtmanager->provided_clock);
-
-  G_OBJECT_CLASS (parent_class)->finalize (object);
-}
-
-static gboolean
-gst_rdt_manager_query_src (GstPad * pad, GstObject * parent, GstQuery * query)
-{
-  GstRDTManager *rdtmanager;
-  gboolean res;
-
-  rdtmanager = GST_RDT_MANAGER (parent);
-
-  switch (GST_QUERY_TYPE (query)) {
-    case GST_QUERY_LATENCY:
-    {
-      GstClockTime latency;
-
-      latency = rdtmanager->latency * GST_MSECOND;
-
-      /* we pretend to be live with a 3 second latency */
-      gst_query_set_latency (query, TRUE, latency, -1);
-
-      GST_DEBUG_OBJECT (rdtmanager, "reporting %" GST_TIME_FORMAT " of latency",
-          GST_TIME_ARGS (latency));
-      res = TRUE;
-      break;
-    }
-    default:
-      res = gst_pad_query_default (pad, parent, query);
-      break;
-  }
-  return res;
-}
-
-static gboolean
-gst_rdt_manager_src_activate_mode (GstPad * pad, GstObject * parent,
-    GstPadMode mode, gboolean active)
-{
-  gboolean result;
-  GstRDTManager *rdtmanager;
-  GstRDTManagerSession *session;
-
-  session = gst_pad_get_element_private (pad);
-  rdtmanager = session->dec;
-
-  switch (mode) {
-    case GST_PAD_MODE_PUSH:
-      if (active) {
-        /* allow data processing */
-        JBUF_LOCK (session);
-        GST_DEBUG_OBJECT (rdtmanager, "Enabling pop on queue");
-        /* Mark as non flushing */
-        session->srcresult = GST_FLOW_OK;
-        gst_segment_init (&session->segment, GST_FORMAT_TIME);
-        session->last_popped_seqnum = -1;
-        session->last_out_time = -1;
-        session->next_seqnum = -1;
-        session->eos = FALSE;
-        JBUF_UNLOCK (session);
-
-        /* start pushing out buffers */
-        GST_DEBUG_OBJECT (rdtmanager, "Starting task on srcpad");
-        result =
-            gst_pad_start_task (pad, (GstTaskFunction) gst_rdt_manager_loop,
-            pad, NULL);
-      } else {
-        /* make sure all data processing stops ASAP */
-        JBUF_LOCK (session);
-        /* mark ourselves as flushing */
-        session->srcresult = GST_FLOW_FLUSHING;
-        GST_DEBUG_OBJECT (rdtmanager, "Disabling pop on queue");
-        /* this unblocks any waiting pops on the src pad task */
-        JBUF_SIGNAL (session);
-        /* unlock clock, we just unschedule, the entry will be released by
-         * the locking streaming thread. */
-        if (session->clock_id)
-          gst_clock_id_unschedule (session->clock_id);
-        JBUF_UNLOCK (session);
-
-        /* NOTE this will hardlock if the state change is called from the src pad
-         * task thread because we will _join() the thread. */
-        GST_DEBUG_OBJECT (rdtmanager, "Stopping task on srcpad");
-        result = gst_pad_stop_task (pad);
-      }
-      break;
-    default:
-      result = FALSE;
-      break;
-  }
-  return result;
-}
-
-static GstFlowReturn
-gst_rdt_manager_handle_data_packet (GstRDTManagerSession * session,
-    GstClockTime timestamp, GstRDTPacket * packet)
-{
-  GstRDTManager *rdtmanager;
-  guint16 seqnum;
-  gboolean tail;
-  GstFlowReturn res;
-  GstBuffer *buffer;
-
-  rdtmanager = session->dec;
-
-  res = GST_FLOW_OK;
-
-  seqnum = 0;
-  GST_DEBUG_OBJECT (rdtmanager,
-      "Received packet #%d at time %" GST_TIME_FORMAT, seqnum,
-      GST_TIME_ARGS (timestamp));
-
-  buffer = gst_rdt_packet_to_buffer (packet);
-
-  JBUF_LOCK_CHECK (session, out_flushing);
-
-  /* insert the packet into the queue now, FIXME, use seqnum */
-  if (!rdt_jitter_buffer_insert (session->jbuf, buffer, timestamp,
-          session->clock_rate, &tail))
-    goto duplicate;
-
-  /* signal addition of new buffer when the _loop is waiting. */
-  if (session->waiting)
-    JBUF_SIGNAL (session);
-
-finished:
-  JBUF_UNLOCK (session);
-
-  return res;
-
-  /* ERRORS */
-out_flushing:
-  {
-    res = session->srcresult;
-    GST_DEBUG_OBJECT (rdtmanager, "flushing %s", gst_flow_get_name (res));
-    gst_buffer_unref (buffer);
-    goto finished;
-  }
-duplicate:
-  {
-    GST_WARNING_OBJECT (rdtmanager, "Duplicate packet #%d detected, dropping",
-        seqnum);
-    session->num_duplicates++;
-    gst_buffer_unref (buffer);
-    goto finished;
-  }
-}
-
-static gboolean
-gst_rdt_manager_parse_caps (GstRDTManager * rdtmanager,
-    GstRDTManagerSession * session, GstCaps * caps)
-{
-  GstStructure *caps_struct;
-  guint val;
-
-  /* first parse the caps */
-  caps_struct = gst_caps_get_structure (caps, 0);
-
-  GST_DEBUG_OBJECT (rdtmanager, "got caps");
-
-  /* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
-   * measure the amount of data in the buffer */
-  if (!gst_structure_get_int (caps_struct, "clock-rate", &session->clock_rate))
-    session->clock_rate = 1000;
-
-  if (session->clock_rate <= 0)
-    goto wrong_rate;
-
-  GST_DEBUG_OBJECT (rdtmanager, "got clock-rate %d", session->clock_rate);
-
-  /* gah, clock-base is uint. If we don't have a base, we will use the first
-   * buffer timestamp as the base time. This will screw up sync but it's better
-   * than nothing. */
-  if (gst_structure_get_uint (caps_struct, "clock-base", &val))
-    session->clock_base = val;
-  else
-    session->clock_base = -1;
-
-  GST_DEBUG_OBJECT (rdtmanager, "got clock-base %" G_GINT64_FORMAT,
-      session->clock_base);
-
-  /* first expected seqnum */
-  if (gst_structure_get_uint (caps_struct, "seqnum-base", &val))
-    session->next_seqnum = val;
-  else
-    session->next_seqnum = -1;
-
-  GST_DEBUG_OBJECT (rdtmanager, "got seqnum-base %d", session->next_seqnum);
-
-  return TRUE;
-
-  /* ERRORS */
-wrong_rate:
-  {
-    GST_DEBUG_OBJECT (rdtmanager, "Invalid clock-rate %d", session->clock_rate);
-    return FALSE;
-  }
-}
-
-static gboolean
-gst_rdt_manager_event_rdt (GstPad * pad, GstObject * parent, GstEvent * event)
-{
-  GstRDTManager *rdtmanager;
-  GstRDTManagerSession *session;
-  gboolean res;
-
-  rdtmanager = GST_RDT_MANAGER (parent);
-  /* find session */
-  session = gst_pad_get_element_private (pad);
-
-  switch (GST_EVENT_TYPE (event)) {
-    case GST_EVENT_CAPS:
-    {
-      GstCaps *caps;
-
-      gst_event_parse_caps (event, &caps);
-      res = gst_rdt_manager_parse_caps (rdtmanager, session, caps);
-      gst_event_unref (event);
-      break;
-    }
-    default:
-      res = gst_pad_event_default (pad, parent, event);
-      break;
-  }
-  return res;
-}
-
-static GstFlowReturn
-gst_rdt_manager_chain_rdt (GstPad * pad, GstObject * parent, GstBuffer * buffer)
-{
-  GstFlowReturn res;
-  GstRDTManager *rdtmanager;
-  GstRDTManagerSession *session;
-  GstClockTime timestamp;
-  GstRDTPacket packet;
-  guint32 ssrc;
-  guint8 pt;
-  gboolean more;
-
-  rdtmanager = GST_RDT_MANAGER (parent);
-
-  GST_DEBUG_OBJECT (rdtmanager, "got RDT packet");
-
-  ssrc = 0;
-  pt = 0;
-
-  GST_DEBUG_OBJECT (rdtmanager, "SSRC %08x, PT %d", ssrc, pt);
-
-  /* find session */
-  session = gst_pad_get_element_private (pad);
-
-  /* see if we have the pad */
-  if (!session->active) {
-    activate_session (rdtmanager, session, ssrc, pt);
-    session->active = TRUE;
-  }
-
-  if (GST_BUFFER_IS_DISCONT (buffer)) {
-    GST_DEBUG_OBJECT (rdtmanager, "received discont");
-    session->discont = TRUE;
-  }
-
-  res = GST_FLOW_OK;
-
-  /* take the timestamp of the buffer. This is the time when the packet was
-   * received and is used to calculate jitter and clock skew. We will adjust
-   * this timestamp with the smoothed value after processing it in the
-   * jitterbuffer. */
-  timestamp = GST_BUFFER_TIMESTAMP (buffer);
-  /* bring to running time */
-  timestamp = gst_segment_to_running_time (&session->segment, GST_FORMAT_TIME,
-      timestamp);
-
-  more = gst_rdt_buffer_get_first_packet (buffer, &packet);
-  while (more) {
-    GstRDTType type;
-
-    type = gst_rdt_packet_get_type (&packet);
-    GST_DEBUG_OBJECT (rdtmanager, "Have packet of type %04x", type);
-
-    if (GST_RDT_IS_DATA_TYPE (type)) {
-      GST_DEBUG_OBJECT (rdtmanager, "We have a data packet");
-      res = gst_rdt_manager_handle_data_packet (session, timestamp, &packet);
-    } else {
-      switch (type) {
-        default:
-          GST_DEBUG_OBJECT (rdtmanager, "Ignoring packet");
-          break;
-      }
-    }
-    if (res != GST_FLOW_OK)
-      break;
-
-    more = gst_rdt_packet_move_to_next (&packet);
-  }
-
-  gst_buffer_unref (buffer);
-
-  return res;
-}
-
-/* push packets from the queue to the downstream demuxer */
-static void
-gst_rdt_manager_loop (GstPad * pad)
-{
-  GstRDTManager *rdtmanager;
-  GstRDTManagerSession *session;
-  GstBuffer *buffer;
-  GstFlowReturn result;
-
-  rdtmanager = GST_RDT_MANAGER (GST_PAD_PARENT (pad));
-
-  session = gst_pad_get_element_private (pad);
-
-  JBUF_LOCK_CHECK (session, flushing);
-  GST_DEBUG_OBJECT (rdtmanager, "Peeking item");
-  while (TRUE) {
-    /* always wait if we are blocked */
-    if (!session->blocked) {
-      /* if we have a packet, we can exit the loop and grab it */
-      if (rdt_jitter_buffer_num_packets (session->jbuf) > 0)
-        break;
-      /* no packets but we are EOS, do eos logic */
-      if (session->eos)
-        goto do_eos;
-    }
-    /* underrun, wait for packets or flushing now */
-    session->waiting = TRUE;
-    JBUF_WAIT_CHECK (session, flushing);
-    session->waiting = FALSE;
-  }
-
-  buffer = rdt_jitter_buffer_pop (session->jbuf);
-
-  GST_DEBUG_OBJECT (rdtmanager, "Got item %p", buffer);
-
-  if (session->discont) {
-    GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
-    session->discont = FALSE;
-  }
-
-  JBUF_UNLOCK (session);
-
-  result = gst_pad_push (session->recv_rtp_src, buffer);
-  if (result != GST_FLOW_OK)
-    goto pause;
-
-  return;
-
-  /* ERRORS */
-flushing:
-  {
-    GST_DEBUG_OBJECT (rdtmanager, "we are flushing");
-    gst_pad_pause_task (session->recv_rtp_src);
-    JBUF_UNLOCK (session);
-    return;
-  }
-do_eos:
-  {
-    /* store result, we are flushing now */
-    GST_DEBUG_OBJECT (rdtmanager, "We are EOS, pushing EOS downstream");
-    session->srcresult = GST_FLOW_EOS;
-    gst_pad_pause_task (session->recv_rtp_src);
-    gst_pad_push_event (session->recv_rtp_src, gst_event_new_eos ());
-    JBUF_UNLOCK (session);
-    return;
-  }
-pause:
-  {
-    GST_DEBUG_OBJECT (rdtmanager, "pausing task, reason %s",
-        gst_flow_get_name (result));
-
-    JBUF_LOCK (session);
-    /* store result */
-    session->srcresult = result;
-    /* we don't post errors or anything because upstream will do that for us
-     * when we pass the return value upstream. */
-    gst_pad_pause_task (session->recv_rtp_src);
-    JBUF_UNLOCK (session);
-    return;
-  }
-}
-
-static GstFlowReturn
-gst_rdt_manager_chain_rtcp (GstPad * pad, GstObject * parent,
-    GstBuffer * buffer)
-{
-  GstRDTManager *src;
-
-#ifdef HAVE_RTCP
-  gboolean valid;
-  GstRTCPPacket packet;
-  gboolean more;
-#endif
-
-  src = GST_RDT_MANAGER (parent);
-
-  GST_DEBUG_OBJECT (src, "got rtcp packet");
-
-#ifdef HAVE_RTCP
-  valid = gst_rtcp_buffer_validate (buffer);
-  if (!valid)
-    goto bad_packet;
-
-  /* position on first packet */
-  more = gst_rtcp_buffer_get_first_packet (buffer, &packet);
-  while (more) {
-    switch (gst_rtcp_packet_get_type (&packet)) {
-      case GST_RTCP_TYPE_SR:
-      {
-        guint32 ssrc, rtptime, packet_count, octet_count;
-        guint64 ntptime;
-        guint count, i;
-
-        gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, &ntptime, &rtptime,
-            &packet_count, &octet_count);
-
-        GST_DEBUG_OBJECT (src,
-            "got SR packet: SSRC %08x, NTP %" G_GUINT64_FORMAT
-            ", RTP %u, PC %u, OC %u", ssrc, ntptime, rtptime, packet_count,
-            octet_count);
-
-        count = gst_rtcp_packet_get_rb_count (&packet);
-        for (i = 0; i < count; i++) {
-          guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
-          guint8 fractionlost;
-          gint32 packetslost;
-
-          gst_rtcp_packet_get_rb (&packet, i, &ssrc, &fractionlost,
-              &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
-
-          GST_DEBUG_OBJECT (src, "got RB packet %d: SSRC %08x, FL %u"
-              ", PL %u, HS %u, JITTER %u, LSR %u, DLSR %u", ssrc, fractionlost,
-              packetslost, exthighestseq, jitter, lsr, dlsr);
-        }
-        break;
-      }
-      case GST_RTCP_TYPE_RR:
-      {
-        guint32 ssrc;
-        guint count, i;
-
-        ssrc = gst_rtcp_packet_rr_get_ssrc (&packet);
-
-        GST_DEBUG_OBJECT (src, "got RR packet: SSRC %08x", ssrc);
-
-        count = gst_rtcp_packet_get_rb_count (&packet);
-        for (i = 0; i < count; i++) {
-          guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
-          guint8 fractionlost;
-          gint32 packetslost;
-
-          gst_rtcp_packet_get_rb (&packet, i, &ssrc, &fractionlost,
-              &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
-
-          GST_DEBUG_OBJECT (src, "got RB packet %d: SSRC %08x, FL %u"
-              ", PL %u, HS %u, JITTER %u, LSR %u, DLSR %u", ssrc, fractionlost,
-              packetslost, exthighestseq, jitter, lsr, dlsr);
-        }
-        break;
-      }
-      case GST_RTCP_TYPE_SDES:
-      {
-        guint chunks, i, j;
-        gboolean more_chunks, more_items;
-
-        chunks = gst_rtcp_packet_sdes_get_chunk_count (&packet);
-        GST_DEBUG_OBJECT (src, "got SDES packet with %d chunks", chunks);
-
-        more_chunks = gst_rtcp_packet_sdes_first_chunk (&packet);
-        i = 0;
-        while (more_chunks) {
-          guint32 ssrc;
-
-          ssrc = gst_rtcp_packet_sdes_get_ssrc (&packet);
-
-          GST_DEBUG_OBJECT (src, "chunk %d, SSRC %08x", i, ssrc);
-
-          more_items = gst_rtcp_packet_sdes_first_item (&packet);
-          j = 0;
-          while (more_items) {
-            GstRTCPSDESType type;
-            guint8 len;
-            gchar *data;
-
-            gst_rtcp_packet_sdes_get_item (&packet, &type, &len, &data);
-
-            GST_DEBUG_OBJECT (src, "item %d, type %d, len %d, data %s", j,
-                type, len, data);
-
-            more_items = gst_rtcp_packet_sdes_next_item (&packet);
-            j++;
-          }
-          more_chunks = gst_rtcp_packet_sdes_next_chunk (&packet);
-          i++;
-        }
-        break;
-      }
-      case GST_RTCP_TYPE_BYE:
-      {
-        guint count, i;
-        gchar *reason;
-
-        reason = gst_rtcp_packet_bye_get_reason (&packet);
-        GST_DEBUG_OBJECT (src, "got BYE packet (reason: %s)",
-            GST_STR_NULL (reason));
-        g_free (reason);
-
-        count = gst_rtcp_packet_bye_get_ssrc_count (&packet);
-        for (i = 0; i < count; i++) {
-          guint32 ssrc;
-
-
-          ssrc = gst_rtcp_packet_bye_get_nth_ssrc (&packet, i);
-
-          GST_DEBUG_OBJECT (src, "SSRC: %08x", ssrc);
-        }
-        break;
-      }
-      case GST_RTCP_TYPE_APP:
-        GST_DEBUG_OBJECT (src, "got APP packet");
-        break;
-      default:
-        GST_WARNING_OBJECT (src, "got unknown RTCP packet");
-        break;
-    }
-    more = gst_rtcp_packet_move_to_next (&packet);
-  }
-  gst_buffer_unref (buffer);
-  return GST_FLOW_OK;
-
-bad_packet:
-  {
-    GST_WARNING_OBJECT (src, "got invalid RTCP packet");
-    return GST_FLOW_OK;
-  }
-#else
-  return GST_FLOW_OK;
-#endif
-}
-
-static void
-gst_rdt_manager_set_property (GObject * object, guint prop_id,
-    const GValue * value, GParamSpec * pspec)
-{
-  GstRDTManager *src;
-
-  src = GST_RDT_MANAGER (object);
-
-  switch (prop_id) {
-    case PROP_LATENCY:
-      src->latency = g_value_get_uint (value);
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-}
-
-static void
-gst_rdt_manager_get_property (GObject * object, guint prop_id, GValue * value,
-    GParamSpec * pspec)
-{
-  GstRDTManager *src;
-
-  src = GST_RDT_MANAGER (object);
-
-  switch (prop_id) {
-    case PROP_LATENCY:
-      g_value_set_uint (value, src->latency);
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-}
-
-static GstClock *
-gst_rdt_manager_provide_clock (GstElement * element)
-{
-  GstRDTManager *rdtmanager;
-
-  rdtmanager = GST_RDT_MANAGER (element);
-
-  return GST_CLOCK_CAST (gst_object_ref (rdtmanager->provided_clock));
-}
-
-static GstStateChangeReturn
-gst_rdt_manager_change_state (GstElement * element, GstStateChange transition)
-{
-  GstStateChangeReturn ret;
-
-  switch (transition) {
-    default:
-      break;
-  }
-
-  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
-
-  switch (transition) {
-    case GST_STATE_CHANGE_READY_TO_PAUSED:
-    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
-      /* we're NO_PREROLL when going to PAUSED */
-      ret = GST_STATE_CHANGE_NO_PREROLL;
-      break;
-    default:
-      break;
-  }
-
-  return ret;
-}
-
-/* Create a pad for receiving RTP for the session in @name
- */
-static GstPad *
-create_recv_rtp (GstRDTManager * rdtmanager, GstPadTemplate * templ,
-    const gchar * name)
-{
-  guint sessid;
-  GstRDTManagerSession *session;
-
-  /* first get the session number */
-  if (name == NULL || sscanf (name, "recv_rtp_sink_%u", &sessid) != 1)
-    goto no_name;
-
-  GST_DEBUG_OBJECT (rdtmanager, "finding session %d", sessid);
-
-  /* get or create session */
-  session = find_session_by_id (rdtmanager, sessid);
-  if (!session) {
-    GST_DEBUG_OBJECT (rdtmanager, "creating session %d", sessid);
-    /* create session now */
-    session = create_session (rdtmanager, sessid);
-    if (session == NULL)
-      goto create_error;
-  }
-  /* check if pad was requested */
-  if (session->recv_rtp_sink != NULL)
-    goto existed;
-
-  GST_DEBUG_OBJECT (rdtmanager, "getting RTP sink pad");
-
-  session->recv_rtp_sink = gst_pad_new_from_template (templ, name);
-  gst_pad_set_element_private (session->recv_rtp_sink, session);
-  gst_pad_set_event_function (session->recv_rtp_sink,
-      gst_rdt_manager_event_rdt);
-  gst_pad_set_chain_function (session->recv_rtp_sink,
-      gst_rdt_manager_chain_rdt);
-  gst_pad_set_active (session->recv_rtp_sink, TRUE);
-  gst_element_add_pad (GST_ELEMENT_CAST (rdtmanager), session->recv_rtp_sink);
-
-  return session->recv_rtp_sink;
-
-  /* ERRORS */
-no_name:
-  {
-    g_warning ("rdtmanager: invalid name given");
-    return NULL;
-  }
-create_error:
-  {
-    /* create_session already warned */
-    return NULL;
-  }
-existed:
-  {
-    g_warning ("rdtmanager: recv_rtp pad already requested for session %d",
-        sessid);
-    return NULL;
-  }
-}
-
-/* Create a pad for receiving RTCP for the session in @name
- */
-static GstPad *
-create_recv_rtcp (GstRDTManager * rdtmanager, GstPadTemplate * templ,
-    const gchar * name)
-{
-  guint sessid;
-  GstRDTManagerSession *session;
-
-  /* first get the session number */
-  if (name == NULL || sscanf (name, "recv_rtcp_sink_%u", &sessid) != 1)
-    goto no_name;
-
-  GST_DEBUG_OBJECT (rdtmanager, "finding session %d", sessid);
-
-  /* get the session, it must exist or we error */
-  session = find_session_by_id (rdtmanager, sessid);
-  if (!session)
-    goto no_session;
-
-  /* check if pad was requested */
-  if (session->recv_rtcp_sink != NULL)
-    goto existed;
-
-  GST_DEBUG_OBJECT (rdtmanager, "getting RTCP sink pad");
-
-  session->recv_rtcp_sink = gst_pad_new_from_template (templ, name);
-  gst_pad_set_element_private (session->recv_rtp_sink, session);
-  gst_pad_set_chain_function (session->recv_rtcp_sink,
-      gst_rdt_manager_chain_rtcp);
-  gst_pad_set_active (session->recv_rtcp_sink, TRUE);
-  gst_element_add_pad (GST_ELEMENT_CAST (rdtmanager), session->recv_rtcp_sink);
-
-  return session->recv_rtcp_sink;
-
-  /* ERRORS */
-no_name:
-  {
-    g_warning ("rdtmanager: invalid name given");
-    return NULL;
-  }
-no_session:
-  {
-    g_warning ("rdtmanager: no session with id %d", sessid);
-    return NULL;
-  }
-existed:
-  {
-    g_warning ("rdtmanager: recv_rtcp pad already requested for session %d",
-        sessid);
-    return NULL;
-  }
-}
-
-/* Create a pad for sending RTCP for the session in @name
- */
-static GstPad *
-create_rtcp (GstRDTManager * rdtmanager, GstPadTemplate * templ,
-    const gchar * name)
-{
-  guint sessid;
-  GstRDTManagerSession *session;
-
-  /* first get the session number */
-  if (name == NULL || sscanf (name, "rtcp_src_%u", &sessid) != 1)
-    goto no_name;
-
-  /* get or create session */
-  session = find_session_by_id (rdtmanager, sessid);
-  if (!session)
-    goto no_session;
-
-  /* check if pad was requested */
-  if (session->rtcp_src != NULL)
-    goto existed;
-
-  session->rtcp_src = gst_pad_new_from_template (templ, name);
-  gst_pad_set_active (session->rtcp_src, TRUE);
-  gst_element_add_pad (GST_ELEMENT_CAST (rdtmanager), session->rtcp_src);
-
-  return session->rtcp_src;
-
-  /* ERRORS */
-no_name:
-  {
-    g_warning ("rdtmanager: invalid name given");
-    return NULL;
-  }
-no_session:
-  {
-    g_warning ("rdtmanager: session with id %d does not exist", sessid);
-    return NULL;
-  }
-existed:
-  {
-    g_warning ("rdtmanager: rtcp_src pad already requested for session %d",
-        sessid);
-    return NULL;
-  }
-}
-
-/* 
- */
-static GstPad *
-gst_rdt_manager_request_new_pad (GstElement * element,
-    GstPadTemplate * templ, const gchar * name, const GstCaps * caps)
-{
-  GstRDTManager *rdtmanager;
-  GstElementClass *klass;
-  GstPad *result;
-
-  g_return_val_if_fail (templ != NULL, NULL);
-  g_return_val_if_fail (GST_IS_RDT_MANAGER (element), NULL);
-
-  rdtmanager = GST_RDT_MANAGER (element);
-  klass = GST_ELEMENT_GET_CLASS (element);
-
-  /* figure out the template */
-  if (templ == gst_element_class_get_pad_template (klass, "recv_rtp_sink_%u")) {
-    result = create_recv_rtp (rdtmanager, templ, name);
-  } else if (templ == gst_element_class_get_pad_template (klass,
-          "recv_rtcp_sink_%u")) {
-    result = create_recv_rtcp (rdtmanager, templ, name);
-  } else if (templ == gst_element_class_get_pad_template (klass, "rtcp_src_%u")) {
-    result = create_rtcp (rdtmanager, templ, name);
-  } else
-    goto wrong_template;
-
-  return result;
-
-  /* ERRORS */
-wrong_template:
-  {
-    g_warning ("rdtmanager: this is not our template");
-    return NULL;
-  }
-}
-
-static void
-gst_rdt_manager_release_pad (GstElement * element, GstPad * pad)
-{
-}

gst-plugins-ugly-1.22.7.tar.xz/gst/realmedia/rdtmanager.h Deleted

@@ -1,93 +0,0 @@
-/* GStreamer
- * Copyright (C) <2005,2006> Wim Taymans <wim@fluendo.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-/*
- * Unless otherwise indicated, Source Code is licensed under MIT license.
- * See further explanation attached in License Statement (distributed in the file
- * LICENSE).
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy of
- * this software and associated documentation files (the "Software"), to deal in
- * the Software without restriction, including without limitation the rights to
- * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
- * of the Software, and to permit persons to whom the Software is furnished to do
- * so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in all
- * copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
- * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
- * SOFTWARE.
- */
-
-#ifndef __GST_RDT_MANAGER_H__
-#define __GST_RDT_MANAGER_H__
-
-#include <gst/gst.h>
-
-G_BEGIN_DECLS
-
-#define GST_TYPE_RDT_MANAGER  		(gst_rdt_manager_get_type())
-#define GST_IS_RDT_MANAGER(obj)  	(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RDT_MANAGER))
-#define GST_IS_RDT_MANAGER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RDT_MANAGER))
-#define GST_RDT_MANAGER(obj)  		(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RDT_MANAGER, GstRDTManager))
-#define GST_RDT_MANAGER_CLASS(klass)  	(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RDT_MANAGER, GstRDTManagerClass))
-
-typedef struct _GstRDTManager GstRDTManager;
-typedef struct _GstRDTManagerClass GstRDTManagerClass;
-typedef struct _GstRDTManagerSession GstRDTManagerSession;
-
-struct _GstRDTManager {
-  GstElement  element;
-
-  guint       latency;
-  GSList     *sessions;
-  GstClock   *provided_clock;
-};
-
-struct _GstRDTManagerClass {
-  GstElementClass parent_class;
-
-  /* get the caps for pt */
-  GstCaps* (*request_pt_map)    (GstRDTManager *rtpdec, guint session, guint pt);
-
-  void     (*clear_pt_map)      (GstRDTManager *rtpdec);
-
-  void     (*on_new_ssrc)       (GstRDTManager *rtpdec, guint session, guint32 ssrc);
-  void     (*on_ssrc_collision) (GstRDTManager *rtpdec, guint session, guint32 ssrc);
-  void     (*on_ssrc_validated) (GstRDTManager *rtpdec, guint session, guint32 ssrc);
-  void     (*on_ssrc_active)    (GstRDTManager *rtpdec, guint session, guint32 ssrc);
-  void     (*on_ssrc_sdes)      (GstRDTManager *rtpdec, guint session, guint32 ssrc);
-  void     (*on_bye_ssrc)       (GstRDTManager *rtpdec, guint session, guint32 ssrc);
-  void     (*on_bye_timeout)    (GstRDTManager *rtpdec, guint session, guint32 ssrc);
-  void     (*on_timeout)        (GstRDTManager *rtpdec, guint session, guint32 ssrc);
-  void     (*on_npt_stop)       (GstRDTManager *rtpdec, guint session, guint32 ssrc);
-};
-
-GType gst_rdt_manager_get_type(void);
-
-GST_ELEMENT_REGISTER_DECLARE (rdtmanager);
-
-G_END_DECLS
-
-#endif /* __GST_RDT_MANAGER_H__ */

gst-plugins-ugly-1.22.7.tar.xz/gst/realmedia/realhash.c Deleted

@@ -1,324 +0,0 @@
-/* GStreamer
- * Copyright (C) <2005,2006> Wim Taymans <wim@fluendo.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-/* Element-Checklist-Version: 5 */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#ifdef HAVE_STDINT_H
-#include <stdint.h>
-#endif
-
-#include <string.h>
-
-#include <gst/gst.h>
-
-#include "realhash.h"
-
-void rtsp_ext_real_calc_response_and_checksum (char *response,
-    char *chksum, char *challenge);
-
-/*
- * The following code has been copied from
- * xine-lib-1.1.1/src/input/libreal/real.c.
- */
-
-static const unsigned char xor_table = {
-  0x05, 0x18, 0x74, 0xd0, 0x0d, 0x09, 0x02, 0x53,
-  0xc0, 0x01, 0x05, 0x05, 0x67, 0x03, 0x19, 0x70,
-  0x08, 0x27, 0x66, 0x10, 0x10, 0x72, 0x08, 0x09,
-  0x63, 0x11, 0x03, 0x71, 0x08, 0x08, 0x70, 0x02,
-  0x10, 0x57, 0x05, 0x18, 0x54, 0x00, 0x00, 0x00
-};
-
-#define LE_32(x) GST_READ_UINT32_LE(x)
-#define BE_32C(x,y) GST_WRITE_UINT32_BE(x,y)
-#define LE_32C(x,y) GST_WRITE_UINT32_LE(x,y)
-
-static void
-hash (char *field, char *param)
-{
-  uint32_t a, b, c, d;
-
-  /* fill variables */
-  a = LE_32 (field);
-  b = LE_32 (field + 4);
-  c = LE_32 (field + 8);
-  d = LE_32 (field + 12);
-
-  a = ((b & c) | (~b & d)) + LE_32 ((param + 0x00)) + a - 0x28955B88;
-  a = ((a << 0x07) | (a >> 0x19)) + b;
-  d = ((a & b) | (~a & c)) + LE_32 ((param + 0x04)) + d - 0x173848AA;
-  d = ((d << 0x0c) | (d >> 0x14)) + a;
-  c = ((d & a) | (~d & b)) + LE_32 ((param + 0x08)) + c + 0x242070DB;
-  c = ((c << 0x11) | (c >> 0x0f)) + d;
-  b = ((c & d) | (~c & a)) + LE_32 ((param + 0x0c)) + b - 0x3E423112;
-  b = ((b << 0x16) | (b >> 0x0a)) + c;
-  a = ((b & c) | (~b & d)) + LE_32 ((param + 0x10)) + a - 0x0A83F051;
-  a = ((a << 0x07) | (a >> 0x19)) + b;
-  d = ((a & b) | (~a & c)) + LE_32 ((param + 0x14)) + d + 0x4787C62A;
-  d = ((d << 0x0c) | (d >> 0x14)) + a;
-  c = ((d & a) | (~d & b)) + LE_32 ((param + 0x18)) + c - 0x57CFB9ED;
-  c = ((c << 0x11) | (c >> 0x0f)) + d;
-  b = ((c & d) | (~c & a)) + LE_32 ((param + 0x1c)) + b - 0x02B96AFF;
-  b = ((b << 0x16) | (b >> 0x0a)) + c;
-  a = ((b & c) | (~b & d)) + LE_32 ((param + 0x20)) + a + 0x698098D8;
-  a = ((a << 0x07) | (a >> 0x19)) + b;
-  d = ((a & b) | (~a & c)) + LE_32 ((param + 0x24)) + d - 0x74BB0851;
-  d = ((d << 0x0c) | (d >> 0x14)) + a;
-  c = ((d & a) | (~d & b)) + LE_32 ((param + 0x28)) + c - 0x0000A44F;
-  c = ((c << 0x11) | (c >> 0x0f)) + d;
-  b = ((c & d) | (~c & a)) + LE_32 ((param + 0x2C)) + b - 0x76A32842;
-  b = ((b << 0x16) | (b >> 0x0a)) + c;
-  a = ((b & c) | (~b & d)) + LE_32 ((param + 0x30)) + a + 0x6B901122;
-  a = ((a << 0x07) | (a >> 0x19)) + b;
-  d = ((a & b) | (~a & c)) + LE_32 ((param + 0x34)) + d - 0x02678E6D;
-  d = ((d << 0x0c) | (d >> 0x14)) + a;
-  c = ((d & a) | (~d & b)) + LE_32 ((param + 0x38)) + c - 0x5986BC72;
-  c = ((c << 0x11) | (c >> 0x0f)) + d;
-  b = ((c & d) | (~c & a)) + LE_32 ((param + 0x3c)) + b + 0x49B40821;
-  b = ((b << 0x16) | (b >> 0x0a)) + c;
-
-  a = ((b & d) | (~d & c)) + LE_32 ((param + 0x04)) + a - 0x09E1DA9E;
-  a = ((a << 0x05) | (a >> 0x1b)) + b;
-  d = ((a & c) | (~c & b)) + LE_32 ((param + 0x18)) + d - 0x3FBF4CC0;
-  d = ((d << 0x09) | (d >> 0x17)) + a;
-  c = ((d & b) | (~b & a)) + LE_32 ((param + 0x2c)) + c + 0x265E5A51;
-  c = ((c << 0x0e) | (c >> 0x12)) + d;
-  b = ((c & a) | (~a & d)) + LE_32 ((param + 0x00)) + b - 0x16493856;
-  b = ((b << 0x14) | (b >> 0x0c)) + c;
-  a = ((b & d) | (~d & c)) + LE_32 ((param + 0x14)) + a - 0x29D0EFA3;
-  a = ((a << 0x05) | (a >> 0x1b)) + b;
-  d = ((a & c) | (~c & b)) + LE_32 ((param + 0x28)) + d + 0x02441453;
-  d = ((d << 0x09) | (d >> 0x17)) + a;
-  c = ((d & b) | (~b & a)) + LE_32 ((param + 0x3c)) + c - 0x275E197F;
-  c = ((c << 0x0e) | (c >> 0x12)) + d;
-  b = ((c & a) | (~a & d)) + LE_32 ((param + 0x10)) + b - 0x182C0438;
-  b = ((b << 0x14) | (b >> 0x0c)) + c;
-  a = ((b & d) | (~d & c)) + LE_32 ((param + 0x24)) + a + 0x21E1CDE6;
-  a = ((a << 0x05) | (a >> 0x1b)) + b;
-  d = ((a & c) | (~c & b)) + LE_32 ((param + 0x38)) + d - 0x3CC8F82A;
-  d = ((d << 0x09) | (d >> 0x17)) + a;
-  c = ((d & b) | (~b & a)) + LE_32 ((param + 0x0c)) + c - 0x0B2AF279;
-  c = ((c << 0x0e) | (c >> 0x12)) + d;
-  b = ((c & a) | (~a & d)) + LE_32 ((param + 0x20)) + b + 0x455A14ED;
-  b = ((b << 0x14) | (b >> 0x0c)) + c;
-  a = ((b & d) | (~d & c)) + LE_32 ((param + 0x34)) + a - 0x561C16FB;
-  a = ((a << 0x05) | (a >> 0x1b)) + b;
-  d = ((a & c) | (~c & b)) + LE_32 ((param + 0x08)) + d - 0x03105C08;
-  d = ((d << 0x09) | (d >> 0x17)) + a;
-  c = ((d & b) | (~b & a)) + LE_32 ((param + 0x1c)) + c + 0x676F02D9;
-  c = ((c << 0x0e) | (c >> 0x12)) + d;
-  b = ((c & a) | (~a & d)) + LE_32 ((param + 0x30)) + b - 0x72D5B376;
-  b = ((b << 0x14) | (b >> 0x0c)) + c;
-
-  a = (b ^ c ^ d) + LE_32 ((param + 0x14)) + a - 0x0005C6BE;
-  a = ((a << 0x04) | (a >> 0x1c)) + b;
-  d = (a ^ b ^ c) + LE_32 ((param + 0x20)) + d - 0x788E097F;
-  d = ((d << 0x0b) | (d >> 0x15)) + a;
-  c = (d ^ a ^ b) + LE_32 ((param + 0x2c)) + c + 0x6D9D6122;
-  c = ((c << 0x10) | (c >> 0x10)) + d;
-  b = (c ^ d ^ a) + LE_32 ((param + 0x38)) + b - 0x021AC7F4;
-  b = ((b << 0x17) | (b >> 0x09)) + c;
-  a = (b ^ c ^ d) + LE_32 ((param + 0x04)) + a - 0x5B4115BC;
-  a = ((a << 0x04) | (a >> 0x1c)) + b;
-  d = (a ^ b ^ c) + LE_32 ((param + 0x10)) + d + 0x4BDECFA9;
-  d = ((d << 0x0b) | (d >> 0x15)) + a;
-  c = (d ^ a ^ b) + LE_32 ((param + 0x1c)) + c - 0x0944B4A0;
-  c = ((c << 0x10) | (c >> 0x10)) + d;
-  b = (c ^ d ^ a) + LE_32 ((param + 0x28)) + b - 0x41404390;
-  b = ((b << 0x17) | (b >> 0x09)) + c;
-  a = (b ^ c ^ d) + LE_32 ((param + 0x34)) + a + 0x289B7EC6;
-  a = ((a << 0x04) | (a >> 0x1c)) + b;
-  d = (a ^ b ^ c) + LE_32 ((param + 0x00)) + d - 0x155ED806;
-  d = ((d << 0x0b) | (d >> 0x15)) + a;
-  c = (d ^ a ^ b) + LE_32 ((param + 0x0c)) + c - 0x2B10CF7B;
-  c = ((c << 0x10) | (c >> 0x10)) + d;
-  b = (c ^ d ^ a) + LE_32 ((param + 0x18)) + b + 0x04881D05;
-  b = ((b << 0x17) | (b >> 0x09)) + c;
-  a = (b ^ c ^ d) + LE_32 ((param + 0x24)) + a - 0x262B2FC7;
-  a = ((a << 0x04) | (a >> 0x1c)) + b;
-  d = (a ^ b ^ c) + LE_32 ((param + 0x30)) + d - 0x1924661B;
-  d = ((d << 0x0b) | (d >> 0x15)) + a;
-  c = (d ^ a ^ b) + LE_32 ((param + 0x3c)) + c + 0x1fa27cf8;
-  c = ((c << 0x10) | (c >> 0x10)) + d;
-  b = (c ^ d ^ a) + LE_32 ((param + 0x08)) + b - 0x3B53A99B;
-  b = ((b << 0x17) | (b >> 0x09)) + c;
-
-  a = ((~d | b) ^ c) + LE_32 ((param + 0x00)) + a - 0x0BD6DDBC;
-  a = ((a << 0x06) | (a >> 0x1a)) + b;
-  d = ((~c | a) ^ b) + LE_32 ((param + 0x1c)) + d + 0x432AFF97;
-  d = ((d << 0x0a) | (d >> 0x16)) + a;
-  c = ((~b | d) ^ a) + LE_32 ((param + 0x38)) + c - 0x546BDC59;
-  c = ((c << 0x0f) | (c >> 0x11)) + d;
-  b = ((~a | c) ^ d) + LE_32 ((param + 0x14)) + b - 0x036C5FC7;
-  b = ((b << 0x15) | (b >> 0x0b)) + c;
-  a = ((~d | b) ^ c) + LE_32 ((param + 0x30)) + a + 0x655B59C3;
-  a = ((a << 0x06) | (a >> 0x1a)) + b;
-  d = ((~c | a) ^ b) + LE_32 ((param + 0x0C)) + d - 0x70F3336E;
-  d = ((d << 0x0a) | (d >> 0x16)) + a;
-  c = ((~b | d) ^ a) + LE_32 ((param + 0x28)) + c - 0x00100B83;
-  c = ((c << 0x0f) | (c >> 0x11)) + d;
-  b = ((~a | c) ^ d) + LE_32 ((param + 0x04)) + b - 0x7A7BA22F;
-  b = ((b << 0x15) | (b >> 0x0b)) + c;
-  a = ((~d | b) ^ c) + LE_32 ((param + 0x20)) + a + 0x6FA87E4F;
-  a = ((a << 0x06) | (a >> 0x1a)) + b;
-  d = ((~c | a) ^ b) + LE_32 ((param + 0x3c)) + d - 0x01D31920;
-  d = ((d << 0x0a) | (d >> 0x16)) + a;
-  c = ((~b | d) ^ a) + LE_32 ((param + 0x18)) + c - 0x5CFEBCEC;
-  c = ((c << 0x0f) | (c >> 0x11)) + d;
-  b = ((~a | c) ^ d) + LE_32 ((param + 0x34)) + b + 0x4E0811A1;
-  b = ((b << 0x15) | (b >> 0x0b)) + c;
-  a = ((~d | b) ^ c) + LE_32 ((param + 0x10)) + a - 0x08AC817E;
-  a = ((a << 0x06) | (a >> 0x1a)) + b;
-  d = ((~c | a) ^ b) + LE_32 ((param + 0x2c)) + d - 0x42C50DCB;
-  d = ((d << 0x0a) | (d >> 0x16)) + a;
-  c = ((~b | d) ^ a) + LE_32 ((param + 0x08)) + c + 0x2AD7D2BB;
-  c = ((c << 0x0f) | (c >> 0x11)) + d;
-  b = ((~a | c) ^ d) + LE_32 ((param + 0x24)) + b - 0x14792C6F;
-  b = ((b << 0x15) | (b >> 0x0b)) + c;
-
-  a += LE_32 (field);
-  b += LE_32 (field + 4);
-  c += LE_32 (field + 8);
-  d += LE_32 (field + 12);
-
-  LE_32C (field, a);
-  LE_32C (field + 4, b);
-  LE_32C (field + 8, c);
-  LE_32C (field + 12, d);
-}
-
-static void
-call_hash (char *key, char *challenge, int len)
-{
-  uint8_t *ptr1, *ptr2;
-  uint32_t a, b, c, d, tmp;
-
-  ptr1 = (uint8_t *) (key + 16);
-  ptr2 = (uint8_t *) (key + 20);
-
-  a = LE_32 (ptr1);
-  b = (a >> 3) & 0x3f;
-  a += len * 8;
-  LE_32C (ptr1, a);
-
-  if (a < (len << 3))
-    ptr2 += 4;
-
-  tmp = LE_32 (ptr2) + (len >> 0x1d);
-  LE_32C (ptr2, tmp);
-  a = 64 - b;
-  c = 0;
-  if (a <= len) {
-
-    memcpy (key + b + 24, challenge, a);
-    hash (key, key + 24);
-    c = a;
-    d = c + 0x3f;
-
-    while (d < len) {
-      hash (key, challenge + d - 0x3f);
-      d += 64;
-      c += 64;
-    }
-    b = 0;
-  }
-
-  memcpy (key + b + 24, challenge + c, len - c);
-}
-
-void
-gst_rtsp_ext_real_calc_response_and_checksum (char *response, char *chksum,
-    char *challenge)
-{
-  int ch_len, table_len, resp_len;
-  int i;
-  char *ptr;
-  char buf128;
-  char field128;
-  char zres20;
-  char buf1128;
-  char buf2128;
-
-  /* initialize return values */
-  memset (response, 0, 64);
-  memset (chksum, 0, 34);
-
-  /* initialize buffer */
-  memset (buf, 0, 128);
-  ptr = buf;
-  BE_32C (ptr, 0xa1e9149d);
-  ptr += 4;
-  BE_32C (ptr, 0x0e6b3b59);
-  ptr += 4;
-
-  if ((ch_len = MIN (strlen (challenge), 56)) == 40) {
-    challenge32 = 0;
-    ch_len = 32;
-  }
-  memcpy (ptr, challenge, ch_len);
-
-  /* xor challenge bytewise with xor_table */
-  table_len = MIN (strlen ((char *) xor_table), 56);
-  for (i = 0; i < table_len; i++)
-    ptri = ptri ^ xor_tablei;
-
-  /* initialize our field */
-  BE_32C (field, 0x01234567);
-  BE_32C (field + 4, 0x89ABCDEF);
-  BE_32C (field + 8, 0xFEDCBA98);
-  BE_32C (field + 12, 0x76543210);
-  BE_32C (field + 16, 0x00000000);
-  BE_32C (field + 20, 0x00000000);
-
-  /* calculate response */
-  call_hash (field, buf, 64);
-  memset (buf1, 0, 64);
-  *buf1 = (char) 128;
-  memcpy (buf2, field + 16, 8);
-  i = (LE_32 ((buf2)) >> 3) & 0x3f;
-  if (i < 56)
-    i = 56 - i;
-  else
-    i = 120 - i;
-  call_hash (field, buf1, i);
-  call_hash (field, buf2, 8);
-  memcpy (zres, field, 16);
-
-  /* convert zres to ascii string */
-  for (i = 0; i < 16; i++) {
-    char a, b;
-
-    a = (zresi >> 4) & 15;
-    b = zresi & 15;
-
-    responsei * 2 = ((a < 10) ? (a + 48) : (a + 87)) & 255;
-    responsei * 2 + 1 = ((b < 10) ? (b + 48) : (b + 87)) & 255;
-  }
-
-  /* add tail */
-  resp_len = strlen (response);
-  strcpy (&responseresp_len, "01d0a8e3");
-
-  /* calculate checksum */
-  for (i = 0; i < resp_len / 4; i++)
-    chksumi = responsei * 4;
-}

gst-plugins-ugly-1.22.7.tar.xz/gst/realmedia/realhash.h Deleted

@@ -1,31 +0,0 @@
-/* GStreamer
- * Copyright (C) <2005,2006> Wim Taymans <wim@fluendo.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-
-#ifndef __GST_RTSP_HASH_H__
-#define __GST_RTSP_HASH_H__
-
-G_BEGIN_DECLS
-
-void
-gst_rtsp_ext_real_calc_response_and_checksum (char *response, char *chksum,
-    char *challenge);
-
-G_END_DECLS
-
-#endif /* __GST_RTSP_HASH_H__ */

gst-plugins-ugly-1.22.7.tar.xz/gst/realmedia/rtspreal.c Deleted

@@ -1,735 +0,0 @@
-/* GStreamer
- * Copyright (C) <2005,2006> Wim Taymans <wim@fluendo.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-/* Element-Checklist-Version: 5 */
-
-/**
- * SECTION:element-rtspreal
- * @title: rtspreal
- *
- * A RealMedia RTSP extension
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#ifdef HAVE_STDINT_H
-#include <stdint.h>
-#endif
-
-#include <stdlib.h>
-#include <string.h>
-
-#include <gst/rtsp/gstrtspextension.h>
-
-#include "realhash.h"
-#include "rtspreal.h"
-#include "asmrules.h"
-
-GST_DEBUG_CATEGORY_STATIC (rtspreal_debug);
-#define GST_CAT_DEFAULT (rtspreal_debug)
-
-#define SERVER_PREFIX "RealServer"
-#define DEFAULT_BANDWIDTH	"10485800"
-
-static GstRTSPResult
-rtsp_ext_real_get_transports (GstRTSPExtension * ext,
-    GstRTSPLowerTrans protocols, gchar ** transport)
-{
-  GstRTSPReal *ctx = (GstRTSPReal *) ext;
-  GString *str;
-
-  if (!ctx->isreal)
-    return GST_RTSP_OK;
-
-  GST_DEBUG_OBJECT (ext, "generating transports for %d", protocols);
-
-  str = g_string_new ("");
-
-  /*
-     if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST) {
-     g_string_append (str, "x-real-rdt/mcast;client_port=%%u1;mode=play,");
-     }
-     if (protocols & GST_RTSP_LOWER_TRANS_UDP) {
-     g_string_append (str, "x-real-rdt/udp;client_port=%%u1;mode=play,");
-     g_string_append (str, "x-pn-tng/udp;client_port=%%u1;mode=play,");
-     }
-   */
-  if (protocols & GST_RTSP_LOWER_TRANS_TCP) {
-    g_string_append (str, "x-real-rdt/tcp;mode=play,");
-    g_string_append (str, "x-pn-tng/tcp;mode=play,");
-  }
-
-  /* if we added something, remove trailing ',' */
-  if (str->len > 0)
-    g_string_truncate (str, str->len - 1);
-
-  *transport = g_string_free (str, FALSE);
-
-  return GST_RTSP_OK;
-}
-
-static GstRTSPResult
-rtsp_ext_real_before_send (GstRTSPExtension * ext, GstRTSPMessage * request)
-{
-  GstRTSPReal *ctx = (GstRTSPReal *) ext;
-
-  switch (request->type_data.request.method) {
-    case GST_RTSP_OPTIONS:
-    {
-      gst_rtsp_message_add_header (request, GST_RTSP_HDR_USER_AGENT,
-          //"RealMedia Player (" GST_PACKAGE_NAME ")");
-          "RealMedia Player Version 6.0.9.1235 (linux-2.0-libc6-i386-gcc2.95)");
-      gst_rtsp_message_add_header (request, GST_RTSP_HDR_CLIENT_CHALLENGE,
-          "9e26d33f2984236010ef6253fb1887f7");
-      gst_rtsp_message_add_header (request, GST_RTSP_HDR_COMPANY_ID,
-          "KnKV4M4I/B2FjJ1TToLycw==");
-      gst_rtsp_message_add_header (request, GST_RTSP_HDR_GUID,
-          "00000000-0000-0000-0000-000000000000");
-      gst_rtsp_message_add_header (request, GST_RTSP_HDR_REGION_DATA, "0");
-      gst_rtsp_message_add_header (request, GST_RTSP_HDR_PLAYER_START_TIME,
-          "28/03/2003:22:50:23 00:00");
-      gst_rtsp_message_add_header (request, GST_RTSP_HDR_CLIENT_ID,
-          "Linux_2.4_6.0.9.1235_play32_RN01_EN_586");
-      ctx->isreal = FALSE;
-      break;
-    }
-    case GST_RTSP_DESCRIBE:
-    {
-      if (ctx->isreal) {
-        gst_rtsp_message_add_header (request, GST_RTSP_HDR_BANDWIDTH,
-            DEFAULT_BANDWIDTH);
-        gst_rtsp_message_add_header (request, GST_RTSP_HDR_GUID,
-            "00000000-0000-0000-0000-000000000000");
-        gst_rtsp_message_add_header (request, GST_RTSP_HDR_REGION_DATA, "0");
-        gst_rtsp_message_add_header (request, GST_RTSP_HDR_CLIENT_ID,
-            "Linux_2.4_6.0.9.1235_play32_RN01_EN_586");
-        gst_rtsp_message_add_header (request, GST_RTSP_HDR_MAX_ASM_WIDTH, "1");
-        gst_rtsp_message_add_header (request, GST_RTSP_HDR_LANGUAGE, "en-US");
-        gst_rtsp_message_add_header (request, GST_RTSP_HDR_REQUIRE,
-            "com.real.retain-entity-for-setup");
-      }
-      break;
-    }
-    case GST_RTSP_SETUP:
-    {
-      if (ctx->isreal) {
-        gchar *value =
-            g_strdup_printf ("%s, sd=%s", ctx->challenge2, ctx->checksum);
-        gst_rtsp_message_add_header (request, GST_RTSP_HDR_REAL_CHALLENGE2,
-            value);
-        gst_rtsp_message_add_header (request, GST_RTSP_HDR_IF_MATCH, ctx->etag);
-        g_free (value);
-      }
-      break;
-    }
-    default:
-      break;
-  }
-  return GST_RTSP_OK;
-}
-
-static GstRTSPResult
-rtsp_ext_real_after_send (GstRTSPExtension * ext, GstRTSPMessage * req,
-    GstRTSPMessage * resp)
-{
-  GstRTSPReal *ctx = (GstRTSPReal *) ext;
-
-  switch (req->type_data.request.method) {
-    case GST_RTSP_OPTIONS:
-    {
-      gchar *challenge1 = NULL;
-      gchar *server = NULL;
-
-      gst_rtsp_message_get_header (resp, GST_RTSP_HDR_SERVER, &server, 0);
-
-      gst_rtsp_message_get_header (resp, GST_RTSP_HDR_REAL_CHALLENGE1,
-          &challenge1, 0);
-      if (!challenge1)
-        goto no_challenge1;
-
-      gst_rtsp_ext_real_calc_response_and_checksum (ctx->challenge2,
-          ctx->checksum, challenge1);
-
-      GST_DEBUG_OBJECT (ctx, "Found Real challenge tag");
-      ctx->isreal = TRUE;
-      break;
-    }
-    case GST_RTSP_DESCRIBE:
-    {
-      gchar *etag = NULL;
-      guint len;
-
-      gst_rtsp_message_get_header (resp, GST_RTSP_HDR_ETAG, &etag, 0);
-      if (etag) {
-        len = sizeof (ctx->etag);
-        strncpy (ctx->etag, etag, len);
-        ctx->etaglen - 1 = '\0';
-      }
-      break;
-    }
-    default:
-      break;
-  }
-  return GST_RTSP_OK;
-
-  /* ERRORS */
-no_challenge1:
-  {
-    GST_DEBUG_OBJECT (ctx, "Could not find challenge tag.");
-    ctx->isreal = FALSE;
-    return GST_RTSP_OK;
-  }
-}
-
-#define ENSURE_SIZE(size)              \
-G_STMT_START {			       \
-  while (data_len < size) {            \
-    data_len += 1024;                  \
-    data = g_realloc (data, data_len); \
-  }                                    \
-} G_STMT_END
-
-#define READ_BUFFER_GEN(src, func, name, dest, dest_len)    \
-G_STMT_START {			                            \
-  dest = (gchar *)func (src, name);                         \
-  dest_len = 0;						    \
-  if (!dest) {                                              \
-    dest = (char *) "";                                     \
-  }                                                         \
-  else if (!strncmp (dest, "buffer;\"", 8)) {               \
-    dest += 8;                                              \
-    dest_len = strlen (dest) - 1;                           \
-    destdest_len = '\0';                                  \
-    g_base64_decode_inplace (dest, &dest_len);            \
-  }                                                         \
-} G_STMT_END
-
-#define READ_BUFFER(sdp, name, dest, dest_len)        \
- READ_BUFFER_GEN(sdp, gst_sdp_message_get_attribute_val, name, dest, dest_len)
-#define READ_BUFFER_M(media, name, dest, dest_len)    \
- READ_BUFFER_GEN(media, gst_sdp_media_get_attribute_val, name, dest, dest_len)
-
-#define READ_INT_GEN(src, func, name, dest)               \
-G_STMT_START {			                          \
-  const gchar *val = func (src, name);                          \
-  if (val && !strncmp (val, "integer;", 8))               \
-      dest = atoi (val + 8);                              \
-    else                                                  \
-      dest = 0;                                           \
-} G_STMT_END
-
-#define READ_INT(sdp, name, dest)                             \
- READ_INT_GEN(sdp, gst_sdp_message_get_attribute_val, name, dest)
-#define READ_INT_M(media, name, dest)                         \
- READ_INT_GEN(media, gst_sdp_media_get_attribute_val, name, dest)
-
-#define READ_STRING(media, name, dest, dest_len)              \
-G_STMT_START {			                              \
-  const gchar *val = gst_sdp_media_get_attribute_val (media, name); \
-  if (val && !strncmp (val, "string;\"", 8)) {                \
-    dest = (gchar *) val + 8;                                 \
-    dest_len = strlen (dest) - 1;                             \
-    destdest_len = '\0';                                    \
-  } else {                                                    \
-    dest = (char *) "";                                       \
-    dest_len = 0;                                             \
-  }                                                           \
-} G_STMT_END
-
-#define WRITE_STRING1(datap, str, str_len)            \
-G_STMT_START {			                      \
-  *datap = str_len;                                   \
-  memcpy ((datap) + 1, str, str_len);                 \
-  datap += str_len + 1;                               \
-} G_STMT_END
-
-#define WRITE_STRING2(datap, str, str_len)            \
-G_STMT_START {			                      \
-  GST_WRITE_UINT16_BE (datap, str_len);               \
-  memcpy (datap + 2, str, str_len);                   \
-  datap += str_len + 2;                               \
-} G_STMT_END
-
-static GstRTSPResult
-rtsp_ext_real_parse_sdp (GstRTSPExtension * ext, GstSDPMessage * sdp,
-    GstStructure * props)
-{
-  GstRTSPReal *ctx = (GstRTSPReal *) ext;
-  guint size;
-  gint i;
-  gchar *title, *author, *copyright, *comment;
-  gsize title_len, author_len, copyright_len, comment_len;
-  guint8 *data = NULL, *datap;
-  guint data_len = 0, offset;
-  GstBuffer *buf;
-  gchar *opaque_data;
-  gsize opaque_data_len, asm_rule_book_len;
-  GHashTable *vars;
-  GString *rules;
-
-  /* don't bother for non-real formats */
-  READ_INT (sdp, "IsRealDataType", ctx->isreal);
-  if (!ctx->isreal)
-    return TRUE;
-
-  /* Force PAUSE | PLAY */
-  //src->methods |= GST_RTSP_PLAY | GST_RTSP_PAUSE;
-
-  ctx->n_streams = gst_sdp_message_medias_len (sdp);
-
-  ctx->max_bit_rate = 0;
-  ctx->avg_bit_rate = 0;
-  ctx->max_packet_size = 0;
-  ctx->avg_packet_size = 0;
-  ctx->duration = 0;
-
-  for (i = 0; i < ctx->n_streams; i++) {
-    const GstSDPMedia *media;
-    gint intval;
-
-    media = gst_sdp_message_get_media (sdp, i);
-
-    READ_INT_M (media, "MaxBitRate", intval);
-    ctx->max_bit_rate += intval;
-    READ_INT_M (media, "AvgBitRate", intval);
-    ctx->avg_bit_rate += intval;
-    READ_INT_M (media, "MaxPacketSize", intval);
-    ctx->max_packet_size = MAX (ctx->max_packet_size, intval);
-    READ_INT_M (media, "AvgPacketSize", intval);
-    ctx->avg_packet_size = (ctx->avg_packet_size * i + intval) / (i + 1);
-    READ_INT_M (media, "Duration", intval);
-    ctx->duration = MAX (ctx->duration, intval);
-  }
-
-  /* FIXME: use GstByteWriter to write the header */
-  /* PROP */
-  offset = 0;
-  size = 50;
-  ENSURE_SIZE (size);
-  datap = data + offset;
-
-  memcpy (datap + 0, "PROP", 4);
-  GST_WRITE_UINT32_BE (datap + 4, size);
-  GST_WRITE_UINT16_BE (datap + 8, 0);
-  GST_WRITE_UINT32_BE (datap + 10, ctx->max_bit_rate);
-  GST_WRITE_UINT32_BE (datap + 14, ctx->avg_bit_rate);
-  GST_WRITE_UINT32_BE (datap + 18, ctx->max_packet_size);
-  GST_WRITE_UINT32_BE (datap + 22, ctx->avg_packet_size);
-  GST_WRITE_UINT32_BE (datap + 26, 0);
-  GST_WRITE_UINT32_BE (datap + 30, ctx->duration);
-  GST_WRITE_UINT32_BE (datap + 34, 0);
-  GST_WRITE_UINT32_BE (datap + 38, 0);
-  GST_WRITE_UINT32_BE (datap + 42, 0);
-  GST_WRITE_UINT16_BE (datap + 46, ctx->n_streams);
-  GST_WRITE_UINT16_BE (datap + 48, 0);
-  offset += size;
-
-  /* CONT */
-  READ_BUFFER (sdp, "Title", title, title_len);
-  READ_BUFFER (sdp, "Author", author, author_len);
-  READ_BUFFER (sdp, "Comment", comment, comment_len);
-  READ_BUFFER (sdp, "Copyright", copyright, copyright_len);
-
-  size = 18 + title_len + author_len + comment_len + copyright_len;
-  ENSURE_SIZE (offset + size);
-  datap = data + offset;
-
-  memcpy (datap, "CONT", 4);
-  GST_WRITE_UINT32_BE (datap + 4, size);
-  GST_WRITE_UINT16_BE (datap + 8, 0);   /* Version */
-  datap += 10;
-  WRITE_STRING2 (datap, title, title_len);
-  WRITE_STRING2 (datap, author, author_len);
-  WRITE_STRING2 (datap, copyright, copyright_len);
-  WRITE_STRING2 (datap, comment, comment_len);
-  offset += size;
-
-  /* fix the hashtale for the rule parser */
-  rules = g_string_new ("");
-  vars = g_hash_table_new (g_str_hash, g_str_equal);
-  g_hash_table_insert (vars, (gchar *) "Bandwidth",
-      (gchar *) DEFAULT_BANDWIDTH);
-
-  /* MDPR */
-  for (i = 0; i < ctx->n_streams; i++) {
-    const GstSDPMedia *media;
-    guint32 len;
-    GstRTSPRealStream *stream;
-    gchar *str;
-    gint rulematchesMAX_RULEMATCHES;
-    gint sel, j, n;
-
-    media = gst_sdp_message_get_media (sdp, i);
-
-    if (media->media && !strcmp (media->media, "data"))
-      continue;
-
-    stream = g_new0 (GstRTSPRealStream, 1);
-    ctx->streams = g_list_append (ctx->streams, stream);
-
-    READ_INT_M (media, "MaxBitRate", stream->max_bit_rate);
-    READ_INT_M (media, "AvgBitRate", stream->avg_bit_rate);
-    READ_INT_M (media, "MaxPacketSize", stream->max_packet_size);
-    READ_INT_M (media, "AvgPacketSize", stream->avg_packet_size);
-    READ_INT_M (media, "StartTime", stream->start_time);
-    READ_INT_M (media, "Preroll", stream->preroll);
-    READ_INT_M (media, "Duration", stream->duration);
-    READ_STRING (media, "StreamName", str, stream->stream_name_len);
-    stream->stream_name = g_strndup (str, stream->stream_name_len);
-    READ_STRING (media, "mimetype", str, stream->mime_type_len);
-    stream->mime_type = g_strndup (str, stream->mime_type_len);
-
-    /* FIXME: Depending on the current bandwidth, we need to select one
-     * bandwidth out of a list offered by the server. Someone needs to write
-     * a parser for strings like
-     *
-     * #($Bandwidth < 67959),TimestampDelivery=T,DropByN=T,priority=9;
-     * #($Bandwidth >= 67959) && ($Bandwidth < 167959),AverageBandwidth=67959,
-     * Priority=9;#($Bandwidth >= 67959) && ($Bandwidth < 167959),
-     * AverageBandwidth=0,Priority=5,OnDepend=\"1\";
-     * #($Bandwidth >= 167959) && ($Bandwidth < 267959),
-     * AverageBandwidth=167959,Priority=9;
-     * #($Bandwidth >= 167959) && ($Bandwidth < 267959),AverageBandwidth=0,
-     * Priority=5,OnDepend=\"3\";#($Bandwidth >= 267959),
-     * AverageBandwidth=267959,Priority=9;#($Bandwidth >= 267959),
-     * AverageBandwidth=0,Priority=5,OnDepend=\"5\";
-     *
-     * As I don't know how to do that, I just use the first entry (sel = 0).
-     * But to give you a starting point, I offer you above string
-     * in the variable 'asm_rule_book'.
-     */
-    READ_STRING (media, "ASMRuleBook", str, asm_rule_book_len);
-    stream->rulebook = gst_asm_rule_book_new (str);
-
-    n = gst_asm_rule_book_match (stream->rulebook, vars, rulematches);
-    for (j = 0; j < n; j++) {
-      g_string_append_printf (rules, "stream=%u;rule=%u,", i, rulematchesj);
-    }
-
-    /* get the MLTI for the first matched rules */
-    sel = rulematches0;
-
-    READ_BUFFER_M (media, "OpaqueData", opaque_data, opaque_data_len);
-
-    if (opaque_data_len < 4) {
-      GST_DEBUG_OBJECT (ctx, "opaque_data_len %" G_GSIZE_FORMAT " < 4",
-          opaque_data_len);
-      goto strange_opaque_data;
-    }
-    if (strncmp (opaque_data, "MLTI", 4)) {
-      GST_DEBUG_OBJECT (ctx, "no MLTI found, appending all");
-      stream->type_specific_data_len = opaque_data_len;
-      stream->type_specific_data = g_memdup2 (opaque_data, opaque_data_len);
-      goto no_type_specific;
-    }
-    opaque_data += 4;
-    opaque_data_len -= 4;
-
-    if (opaque_data_len < 2) {
-      GST_DEBUG_OBJECT (ctx, "opaque_data_len %" G_GSIZE_FORMAT " < 2",
-          opaque_data_len);
-      goto strange_opaque_data;
-    }
-    stream->num_rules = GST_READ_UINT16_BE (opaque_data);
-    opaque_data += 2;
-    opaque_data_len -= 2;
-
-    if (sel >= stream->num_rules) {
-      GST_DEBUG_OBJECT (ctx, "sel %d >= num_rules %d", sel, stream->num_rules);
-      goto strange_opaque_data;
-    }
-
-    if (opaque_data_len < 2 * sel) {
-      GST_DEBUG_OBJECT (ctx, "opaque_data_len %" G_GSIZE_FORMAT
-          " < 2 * sel (%d)", opaque_data_len, 2 * sel);
-      goto strange_opaque_data;
-    }
-    opaque_data += 2 * sel;
-    opaque_data_len -= 2 * sel;
-
-    if (opaque_data_len < 2) {
-      GST_DEBUG_OBJECT (ctx, "opaque_data_len %" G_GSIZE_FORMAT " < 2",
-          opaque_data_len);
-      goto strange_opaque_data;
-    }
-    stream->codec = GST_READ_UINT16_BE (opaque_data);
-    opaque_data += 2;
-    opaque_data_len -= 2;
-
-    if (opaque_data_len < 2 * (stream->num_rules - sel - 1)) {
-      GST_DEBUG_OBJECT (ctx, "opaque_data_len %" G_GSIZE_FORMAT
-          " < %d", opaque_data_len, 2 * (stream->num_rules - sel - 1));
-      goto strange_opaque_data;
-    }
-    opaque_data += 2 * (stream->num_rules - sel - 1);
-    opaque_data_len -= 2 * (stream->num_rules - sel - 1);
-
-    if (opaque_data_len < 2) {
-      GST_DEBUG_OBJECT (ctx, "opaque_data_len %" G_GSIZE_FORMAT " < 2",
-          opaque_data_len);
-      goto strange_opaque_data;
-    }
-    stream->num_rules = GST_READ_UINT16_BE (opaque_data);
-    opaque_data += 2;
-    opaque_data_len -= 2;
-
-    if (stream->codec > stream->num_rules) {
-      GST_DEBUG_OBJECT (ctx, "codec %d > num_rules %d", stream->codec,
-          stream->num_rules);
-      goto strange_opaque_data;
-    }
-
-    for (j = 0; j < stream->codec; j++) {
-      if (opaque_data_len < 4) {
-        GST_DEBUG_OBJECT (ctx, "opaque_data_len %" G_GSIZE_FORMAT " < 4",
-            opaque_data_len);
-        goto strange_opaque_data;
-      }
-      len = GST_READ_UINT32_BE (opaque_data);
-      opaque_data += 4;
-      opaque_data_len -= 4;
-
-      if (opaque_data_len < len) {
-        GST_DEBUG_OBJECT (ctx, "opaque_data_len %" G_GSIZE_FORMAT " < len %d",
-            opaque_data_len, len);
-        goto strange_opaque_data;
-      }
-      opaque_data += len;
-      opaque_data_len -= len;
-    }
-
-    if (opaque_data_len < 4) {
-      GST_DEBUG_OBJECT (ctx, "opaque_data_len %" G_GSIZE_FORMAT " < 4",
-          opaque_data_len);
-      goto strange_opaque_data;
-    }
-    stream->type_specific_data_len = GST_READ_UINT32_BE (opaque_data);
-    opaque_data += 4;
-    opaque_data_len -= 4;
-
-    if (opaque_data_len < stream->type_specific_data_len) {
-      GST_DEBUG_OBJECT (ctx, "opaque_data_len %" G_GSIZE_FORMAT " < %d",
-          opaque_data_len, stream->type_specific_data_len);
-      goto strange_opaque_data;
-    }
-    stream->type_specific_data =
-        g_memdup2 (opaque_data, stream->type_specific_data_len);
-
-  no_type_specific:
-    size =
-        46 + stream->stream_name_len + stream->mime_type_len +
-        stream->type_specific_data_len;
-    ENSURE_SIZE (offset + size);
-    datap = data + offset;
-
-    memcpy (datap, "MDPR", 4);
-    GST_WRITE_UINT32_BE (datap + 4, size);
-    GST_WRITE_UINT16_BE (datap + 8, 0);
-    GST_WRITE_UINT16_BE (datap + 10, i);
-    GST_WRITE_UINT32_BE (datap + 12, stream->max_bit_rate);
-    GST_WRITE_UINT32_BE (datap + 16, stream->avg_bit_rate);
-    GST_WRITE_UINT32_BE (datap + 20, stream->max_packet_size);
-    GST_WRITE_UINT32_BE (datap + 24, stream->avg_packet_size);
-    GST_WRITE_UINT32_BE (datap + 28, stream->start_time);
-    GST_WRITE_UINT32_BE (datap + 32, stream->preroll);
-    GST_WRITE_UINT32_BE (datap + 36, stream->duration);
-    datap += 40;
-    WRITE_STRING1 (datap, stream->stream_name, stream->stream_name_len);
-    WRITE_STRING1 (datap, stream->mime_type, stream->mime_type_len);
-    GST_WRITE_UINT32_BE (datap, stream->type_specific_data_len);
-    if (stream->type_specific_data_len)
-      memcpy (datap + 4, stream->type_specific_data,
-          stream->type_specific_data_len);
-    offset += size;
-  }
-
-  /* destroy the rulebook hashtable now */
-  g_hash_table_destroy (vars);
-
-  /* strip final , if we added some stream rules */
-  if (rules->len > 0) {
-    rules = g_string_truncate (rules, rules->len - 1);
-  }
-
-  /* and store rules in the context */
-  ctx->rules = g_string_free (rules, FALSE);
-
-  /* DATA */
-  size = 18;
-  ENSURE_SIZE (offset + size);
-  datap = data + offset;
-
-  memcpy (datap, "DATA", 4);
-  GST_WRITE_UINT32_BE (datap + 4, size);
-  GST_WRITE_UINT16_BE (datap + 8, 0);
-  GST_WRITE_UINT32_BE (datap + 10, 0);  /* number of packets */
-  GST_WRITE_UINT32_BE (datap + 14, 0);  /* next data header */
-  offset += size;
-
-  buf = gst_buffer_new_wrapped (data, offset);
-
-  /* Set on caps */
-  GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_HEADER);
-  gst_structure_set (props, "config", GST_TYPE_BUFFER, buf, NULL);
-  gst_buffer_unref (buf);
-
-  /* Overwrite encoding and media fields */
-  gst_structure_set (props, "encoding-name", G_TYPE_STRING, "X-REAL-RDT", NULL);
-  gst_structure_set (props, "media", G_TYPE_STRING, "application", NULL);
-
-  return TRUE;
-
-  /* ERRORS */
-strange_opaque_data:
-  {
-    g_string_free (rules, TRUE);
-    g_hash_table_destroy (vars);
-    g_free (data);
-
-    GST_ELEMENT_ERROR (ctx, RESOURCE, WRITE, ("Strange opaque data."), (NULL));
-    return FALSE;
-  }
-}
-
-static GstRTSPResult
-rtsp_ext_real_stream_select (GstRTSPExtension * ext, GstRTSPUrl * url)
-{
-  GstRTSPReal *ctx = (GstRTSPReal *) ext;
-  GstRTSPResult res;
-  GstRTSPMessage request = { 0 };
-  GstRTSPMessage response = { 0 };
-  gchar *req_url;
-
-  if (!ctx->isreal)
-    return GST_RTSP_OK;
-
-  if (!ctx->rules)
-    return GST_RTSP_OK;
-
-  req_url = gst_rtsp_url_get_request_uri (url);
-
-  /* create SET_PARAMETER */
-  if ((res = gst_rtsp_message_init_request (&request, GST_RTSP_SET_PARAMETER,
-              req_url)) < 0)
-    goto create_request_failed;
-
-  g_free (req_url);
-
-  gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SUBSCRIBE, ctx->rules);
-
-  /* send SET_PARAMETER */
-  if ((res = gst_rtsp_extension_send (ext, &request, &response)) < 0)
-    goto send_error;
-
-  gst_rtsp_message_unset (&request);
-  gst_rtsp_message_unset (&response);
-
-  return GST_RTSP_OK;
-
-  /* ERRORS */
-create_request_failed:
-  {
-    GST_ELEMENT_ERROR (ctx, LIBRARY, INIT,
-        ("Could not create request."), (NULL));
-    g_free (req_url);
-    goto reset;
-  }
-send_error:
-  {
-    GST_ELEMENT_ERROR (ctx, RESOURCE, WRITE,
-        ("Could not send message."), (NULL));
-    goto reset;
-  }
-reset:
-  {
-    gst_rtsp_message_unset (&request);
-    gst_rtsp_message_unset (&response);
-    return res;
-  }
-}
-
-static void gst_rtsp_real_extension_init (gpointer g_iface,
-    gpointer iface_data);
-static void gst_rtsp_real_finalize (GObject * obj);
-
-#define gst_rtsp_real_parent_class parent_class
-G_DEFINE_TYPE_WITH_CODE (GstRTSPReal, gst_rtsp_real, GST_TYPE_ELEMENT,
-    G_IMPLEMENT_INTERFACE (GST_TYPE_RTSP_EXTENSION,
-        gst_rtsp_real_extension_init));
-GST_ELEMENT_REGISTER_DEFINE (rtspreal, "rtspreal",
-    GST_RANK_MARGINAL, GST_TYPE_RTSP_REAL);
-
-static void
-gst_rtsp_real_class_init (GstRTSPRealClass * g_class)
-{
-  GObjectClass *gobject_class = (GObjectClass *) g_class;
-  GstElementClass *gstelement_class = (GstElementClass *) g_class;
-
-  gobject_class->finalize = gst_rtsp_real_finalize;
-
-  gst_element_class_set_static_metadata (gstelement_class,
-      "RealMedia RTSP Extension", "Network/Extension/Protocol",
-      "Extends RTSP so that it can handle RealMedia setup",
-      "Wim Taymans <wim.taymans@gmail.com>");
-
-  GST_DEBUG_CATEGORY_INIT (rtspreal_debug, "rtspreal", 0,
-      "RealMedia RTSP extension");
-}
-
-static void
-gst_rtsp_real_init (GstRTSPReal * rtspreal)
-{
-  rtspreal->isreal = FALSE;
-}
-
-static void
-gst_rtsp_stream_free (GstRTSPRealStream * stream)
-{
-  g_free (stream->stream_name);
-  g_free (stream->mime_type);
-  gst_asm_rule_book_free (stream->rulebook);
-  g_free (stream->type_specific_data);
-
-  g_free (stream);
-}
-
-static void
-gst_rtsp_real_finalize (GObject * obj)
-{
-  GstRTSPReal *r = (GstRTSPReal *) obj;
-
-  g_list_foreach (r->streams, (GFunc) gst_rtsp_stream_free, NULL);
-  g_list_free (r->streams);
-  g_free (r->rules);
-
-  G_OBJECT_CLASS (parent_class)->finalize (obj);
-}
-
-static void
-gst_rtsp_real_extension_init (gpointer g_iface, gpointer iface_data)
-{
-  GstRTSPExtensionInterface *iface = (GstRTSPExtensionInterface *) g_iface;
-
-  iface->before_send = rtsp_ext_real_before_send;
-  iface->after_send = rtsp_ext_real_after_send;
-  iface->parse_sdp = rtsp_ext_real_parse_sdp;
-  iface->stream_select = rtsp_ext_real_stream_select;
-  iface->get_transports = rtsp_ext_real_get_transports;
-}

gst-plugins-ugly-1.22.7.tar.xz/gst/realmedia/rtspreal.h Deleted

gst-plugins-ugly-1.22.7.tar.xz/tests/check/elements/amrnbenc.c Deleted

@@ -1,144 +0,0 @@
-/*
- * GStreamer
- *
- * unit test for amrnbenc
- *
- * Copyright (C) 2006 Thomas Vander Stichele <thomas at apestaart dot org>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-
-#include <gst/check/gstcheck.h>
-#include <gst/audio/audio.h>
-
-#define SRC_CAPS "audio/x-raw, format = (string)" GST_AUDIO_NE (S16) ", " \
-    "layout = (string) interleaved, channels = (int) 1, rate = (int) 8000"
-#define SINK_CAPS "audio/AMR"
-
-static GstPad *srcpad, *sinkpad;
-
-static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
-    GST_PAD_SINK,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS (SINK_CAPS)
-    );
-
-static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
-    GST_PAD_SRC,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS (SRC_CAPS)
-    );
-
-static void
-buffer_unref (void *buffer, void *user_data)
-{
-  gst_buffer_unref (GST_BUFFER (buffer));
-}
-
-static GstElement *
-setup_amrnbenc (void)
-{
-  GstElement *amrnbenc;
-  GstCaps *caps;
-  GstBus *bus;
-
-  GST_DEBUG ("setup_amrnbenc");
-
-  amrnbenc = gst_check_setup_element ("amrnbenc");
-  srcpad = gst_check_setup_src_pad (amrnbenc, &srctemplate);
-  sinkpad = gst_check_setup_sink_pad (amrnbenc, &sinktemplate);
-  gst_pad_set_active (srcpad, TRUE);
-  gst_pad_set_active (sinkpad, TRUE);
-
-  bus = gst_bus_new ();
-  gst_element_set_bus (amrnbenc, bus);
-
-  fail_unless (gst_element_set_state (amrnbenc,
-          GST_STATE_PLAYING) != GST_STATE_CHANGE_FAILURE,
-      "could not set to playing");
-
-  caps = gst_caps_from_string (SRC_CAPS);
-  gst_check_setup_events (srcpad, amrnbenc, caps, GST_FORMAT_TIME);
-  gst_caps_unref (caps);
-
-  buffers = NULL;
-  return amrnbenc;
-}
-
-static void
-cleanup_amrnbenc (GstElement * amrnbenc)
-{
-  GstBus *bus;
-
-  /* free encoded buffers */
-  g_list_foreach (buffers, buffer_unref, NULL);
-  g_list_free (buffers);
-  buffers = NULL;
-
-  bus = GST_ELEMENT_BUS (amrnbenc);
-  gst_bus_set_flushing (bus, TRUE);
-  gst_object_unref (bus);
-
-  GST_DEBUG ("cleanup_amrnbenc");
-  gst_pad_set_active (srcpad, FALSE);
-  gst_pad_set_active (sinkpad, FALSE);
-  gst_check_teardown_src_pad (amrnbenc);
-  gst_check_teardown_sink_pad (amrnbenc);
-  gst_check_teardown_element (amrnbenc);
-}
-
-/* push a random block of audio of the given size */
-static void
-push_data (gint size, GstFlowReturn expected_return)
-{
-  GstBuffer *buffer;
-  GstFlowReturn res;
-
-  buffer = gst_buffer_new_and_alloc (size);
-  /* make valgrind happier */
-  gst_buffer_memset (buffer, 0, 0, size);
-
-  res = gst_pad_push (srcpad, buffer);
-  fail_unless (res == expected_return,
-      "pushing audio returned %d (%s) not %d (%s)", res,
-      gst_flow_get_name (res), expected_return,
-      gst_flow_get_name (expected_return));
-}
-
-GST_START_TEST (test_enc)
-{
-  GstElement *amrnbenc;
-
-  amrnbenc = setup_amrnbenc ();
-  push_data (1000, GST_FLOW_OK);
-
-  cleanup_amrnbenc (amrnbenc);
-}
-
-GST_END_TEST;
-
-static Suite *
-amrnbenc_suite ()
-{
-  Suite *s = suite_create ("amrnbenc");
-  TCase *tc_chain = tcase_create ("general");
-
-  suite_add_tcase (s, tc_chain);
-  tcase_add_test (tc_chain, test_enc);
-  return s;
-}
-
-GST_CHECK_MAIN (amrnbenc);

gst-plugins-ugly-1.22.7.tar.xz/ChangeLog -> gst-plugins-ugly-1.24.5.tar.xz/ChangeLog Changed

@@ -1,537 +1,384 @@
-=== release 1.22.7 ===
+=== release 1.24.5 ===
 
-2023-11-13 11:04:22 +0000  Tim-Philipp Müller <tim@centricular.com>
+2024-06-20 12:54:15 +0100  Tim-Philipp Müller <tim@centricular.com>
 
 	* NEWS:
 	* RELEASE:
 	* gst-plugins-ugly.doap:
 	* meson.build:
-	  Release 1.22.7
+	  Release 1.24.5
 
-2023-09-20 19:41:00 +0100  Tim-Philipp Müller <tim@centricular.com>
+2024-05-29 13:51:27 +0300  Tim-Philipp Müller <tim@centricular.com>
 
 	* meson.build:
-	  Back to development
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5371>
+	  Back to development after 1.24.4
 
-=== release 1.22.6 ===
+=== release 1.24.4 ===
 
-2023-09-20 18:10:57 +0100  Tim-Philipp Müller <tim@centricular.com>
+2024-05-29 13:44:50 +0300  Tim-Philipp Müller <tim@centricular.com>
 
 	* NEWS:
 	* RELEASE:
 	* gst-plugins-ugly.doap:
 	* meson.build:
-	  Release 1.22.6
+	  Release 1.24.4
 
-2023-07-20 16:57:47 +0100  Tim-Philipp Müller <tim@centricular.com>
+2024-04-30 00:36:59 +0100  Tim-Philipp Müller <tim@centricular.com>
 
 	* meson.build:
-	  Back to development
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5082>
+	  Back to development after 1.24.3
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6772>
 
-=== release 1.22.5 ===
+=== release 1.24.3 ===
 
-2023-07-20 15:22:48 +0100  Tim-Philipp Müller <tim@centricular.com>
+2024-04-30 00:15:23 +0100  Tim-Philipp Müller <tim@centricular.com>
 
 	* NEWS:
 	* RELEASE:
 	* gst-plugins-ugly.doap:
 	* meson.build:
-	  Release 1.22.5
-
-2023-07-07 10:08:21 +0300  Sebastian Dröge <sebastian@centricular.com>
-
-	* gst/realmedia/rmdemux.c:
-	  rmdemux: Check for integer overflow when calculation audio packet size
-	  Fixes ZDI-CAN-21444
-	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2782
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5073>
-
-2023-07-07 10:08:03 +0300  Sebastian Dröge <sebastian@centricular.com>
-
-	* gst/realmedia/rmdemux.c:
-	  rmdemux: Use GST_LOG_OBJECT instead of GST_LOG
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5073>
-
-2023-07-07 10:00:19 +0300  Sebastian Dröge <sebastian@centricular.com>
+	  Release 1.24.3
 
-	* gst/realmedia/rmdemux.c:
-	  rmdemux: Check that enough SIPR audio data is available when copying
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5073>
-
-2023-07-07 09:59:20 +0300  Sebastian Dröge <sebastian@centricular.com>
-
-	* gst/realmedia/rmdemux.c:
-	  rmdemux: Check for integer overflows when calculating the size of SIPR audio buffers
-	  Fixes ZDI-CAN-21443
-	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2782
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5073>
-
-2023-07-07 09:59:04 +0300  Sebastian Dröge <sebastian@centricular.com>
-
-	* gst/realmedia/rmdemux.c:
-	  rmdemux: Use GST_LOG_OBJECT instead of GST_LOG
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5073>
-
-2023-06-20 19:10:38 +0100  Tim-Philipp Müller <tim@centricular.com>
+2024-04-10 00:04:02 +0100  Tim-Philipp Müller <tim@centricular.com>
 
 	* meson.build:
-	  Back to development
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4907>
+	  Back to development after 1.24.2
 
-=== release 1.22.4 ===
+=== release 1.24.2 ===
 
-2023-06-20 17:42:25 +0100  Tim-Philipp Müller <tim@centricular.com>
+2024-04-09 21:48:55 +0100  Tim-Philipp Müller <tim@centricular.com>
 
 	* NEWS:
 	* RELEASE:
 	* gst-plugins-ugly.doap:
 	* meson.build:
-	  Release 1.22.4
+	  Release 1.24.2
 
-2023-05-19 12:36:19 +0100  Tim-Philipp Müller <tim@centricular.com>
+2024-03-22 01:38:06 +0100  Tim-Philipp Müller <tim@centricular.com>
 
 	* meson.build:
 	  Back to development
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4679>
 
-=== release 1.22.3 ===
+=== release 1.24.1 ===
 
-2023-05-19 09:23:19 +0100  Tim-Philipp Müller <tim@centricular.com>
+2024-03-21 21:47:53 +0100  Tim-Philipp Müller <tim@centricular.com>
 
 	* NEWS:
-	* README.md:
 	* RELEASE:
 	* gst-plugins-ugly.doap:
 	* meson.build:
-	  Release 1.22.3
+	  Release 1.24.1
 
-2023-04-12 00:31:17 +0100  Tim-Philipp Müller <tim@centricular.com>
+2024-03-05 13:45:27 +0000  Tim-Philipp Müller <tim@centricular.com>
 
 	* meson.build:
 	  Back to development
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6264>
 
-=== release 1.22.2 ===
+=== release 1.24.0 ===
 
-2023-04-11 17:29:28 +0100  Tim-Philipp Müller <tim@centricular.com>
-
-	* NEWS:
-	* RELEASE:
-	* gst-plugins-ugly.doap:
-	* meson.build:
-	  Release 1.22.2
-
-2023-03-17 16:32:45 +0200  Sebastian Dröge <sebastian@centricular.com>
-
-	* tests/check/elements/x264enc.c:
-	  plugins: Fix various trivial clang compiler warnings
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4221>
-
-2023-03-04 16:13:04 +0000  Tim-Philipp Müller <tim@centricular.com>
-
-	* meson.build:
-	  Back to development
-
-=== release 1.22.1 ===
-
-2023-03-04 13:42:32 +0000  Tim-Philipp Müller <tim@centricular.com>
+2024-03-04 23:51:42 +0000  Tim-Philipp Müller <tim@centricular.com>
 
 	* NEWS:
+	* README.md:
 	* RELEASE:
 	* gst-plugins-ugly.doap:
 	* meson.build:
-	  Release 1.22.1
+	  Release 1.24.0
 
-2023-01-25 16:46:42 +0000  Tim-Philipp Müller <tim@centricular.com>
+2024-02-26 09:27:40 +0100  Edward Hervey <edward@centricular.com>
 
-	* meson.build:
-	  Back to development
-
-=== release 1.22.0 ===
-
-2023-01-23 19:29:34 +0000  Tim-Philipp Müller <tim@centricular.com>
-
-	* NEWS:
+	* README.md:
 	* RELEASE:
-	* gst-plugins-ugly.doap:
-	* meson.build:
-	  Release 1.22.0
-
-2023-01-23 16:28:13 +0000  Tim-Philipp Müller <tim@centricular.com>
-
-	* po/hr.po:
-	* po/ro.po:
-	  gst-plugins-ugly: update translations
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3773>
+	  docs: Use Discourse and Matrix as prefered communication channels
+	  Part of: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6220
 
-=== release 1.21.90 ===
+=== release 1.23.90 ===
 
-2023-01-13 19:08:48 +0000  Tim-Philipp Müller <tim@centricular.com>
+2024-02-23 18:20:11 +0000  Tim-Philipp Müller <tim@centricular.com>
 
 	* NEWS:
 	* RELEASE:
 	* gst-plugins-ugly.doap:
 	* meson.build:
-	  Release 1.21.90
-
-2023-01-11 15:41:26 +0000  Tim-Philipp Müller <tim@centricular.com>
-
-	* po/LINGUAS:
-	* po/eo.po:
-	* po/es.po:
-	* po/hr.po:
-	* po/ka.po:
-	* po/lt.po:
-	* po/ru.po:
-	* po/sk.po:
-	* po/sl.po:
-	* po/sq.po:
-	* po/sr.po:
-	* po/uk.po:
-	* po/zh_TW.po:
-	  gst-plugins-ugly: update translations
-	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1575
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3711>
+	  Release 1.23.90
 
-2023-01-11 14:53:39 +0000  Tim-Philipp Müller <tim@centricular.com>
+2023-09-17 18:47:31 +0200  Jochen Henneberg <jh@henneberg-systemdesign.com>
 
-	* scripts/dist-translations.py:
-	* scripts/meson.build:
-	  Fix translation pot files when creating dist tarballs
-	  Add version as per Translation Project requirements and
-	  also add a .pot file without the ABI suffix.
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3711>
+	* gst/asfdemux/gstrtpasfdepay.c:
+	  rtpasfdepay: Enabled header extension aggregation
+	  Because this depayloader may build several output buffers within one
+	  process run we push them all into a GstBufferList and push them out at
+	  once to make sure that each buffer gets notified about each header
+	  extension.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378>
 
-2022-12-05 02:29:08 +0000  Tim-Philipp Müller <tim@centricular.com>
+2024-02-15 16:38:53 +0000  Tim-Philipp Müller <tim@centricular.com>
 
 	* meson.build:
 	  Back to development
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6126>
 
-=== release 1.21.3 ===
+=== release 1.23.2 ===
 
-2022-12-05 01:28:21 +0000  Tim-Philipp Müller <tim@centricular.com>
+2024-02-15 15:37:17 +0000  Tim-Philipp Müller <tim@centricular.com>
 
 	* NEWS:
 	* RELEASE:
 	* gst-plugins-ugly.doap:
 	* meson.build:
-	  Release 1.21.3
-
-2022-12-04 12:25:41 +0000  Tim-Philipp Müller <tim@centricular.com>
+	  Release 1.23.2
 
-	* ChangeLog:
-	* docs/random/ChangeLog-0.8:
-	  Remove ChangeLog files from git repository
-	  This information is tracked fully in the git repository, so
-	  no point having the ChangeLog duplicate it, and it interferes
-	  with grepping the repository.
-	  We are going to create the ChangeLogs on the fly when generating
-	  tarballs going forward (with a limited history), since it's still
-	  valuable for tarball consumers to be able to easily see a list of
-	  recent changes.
-	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer-project/-/issues/73
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3521>
+2024-02-14 00:38:10 +0000  Tim-Philipp Müller <tim@centricular.com>
 
-2022-11-07 00:10:39 +0000  Tim-Philipp Müller <tim@centricular.com>
+	* po/lv.po:
+	  gst-plugins-ugly: update translations
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6113>
 
-	* meson.build:
-	* scripts/gen-changelog.py:
-	  meson: Generate ChangeLog files for release tarballs on dist
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3521>
+2024-02-10 19:47:24 +0000  Tim-Philipp Müller <tim@centricular.com>
 
-2022-11-08 02:08:08 +0000  Tim-Philipp Müller <tim@centricular.com>
+	* docs/gst_plugins_cache.json:
+	* gst/realmedia/asmrules.c:
+	* gst/realmedia/asmrules.h:
+	* gst/realmedia/gstrdtbuffer.c:
+	* gst/realmedia/gstrdtbuffer.h:
+	* gst/realmedia/meson.build:
+	* gst/realmedia/rdtdepay.c:
+	* gst/realmedia/rdtdepay.h:
+	* gst/realmedia/rdtjitterbuffer.c:
+	* gst/realmedia/rdtjitterbuffer.h:
+	* gst/realmedia/rdtmanager.c:
+	* gst/realmedia/rdtmanager.h:
+	* gst/realmedia/realhash.c:
+	* gst/realmedia/realhash.h:
+	* gst/realmedia/realmedia.c:
+	* gst/realmedia/rtspreal.c:
+	* gst/realmedia/rtspreal.h:
+	  realmedia: remove RealServer RTSP extension, RDT handling and PNM source
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6091>
+
+2024-02-06 18:09:02 +0000  Tim-Philipp Müller <tim@centricular.com>
 
 	* meson.build:
 	  Back to development
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3358>
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6066>
 
-=== release 1.21.2 ===
+=== release 1.23.1 ===
 
-2022-11-07 23:53:59 +0000  Tim-Philipp Müller <tim@centricular.com>
+2024-02-06 16:37:19 +0000  Tim-Philipp Müller <tim@centricular.com>
 
-	* ChangeLog:
 	* NEWS:
 	* RELEASE:
 	* gst-plugins-ugly.doap:
 	* meson.build:
-	  Release 1.21.2
-
-2022-11-07 23:53:57 +0000  Tim-Philipp Müller <tim@centricular.com>
-
-	* ChangeLog:
-	  Update ChangeLogs for 1.21.2
-
-2022-10-23 20:32:35 +0100  Tim-Philipp Müller <tim@centricular.com>
-
-	* docs/gst_plugins_cache.json:
-	* gst/meson.build:
-	* gst/xingmux/gstxingmux.c:
-	* gst/xingmux/gstxingmux.h:
-	* gst/xingmux/meson.build:
-	* gst/xingmux/plugin.c:
-	* meson_options.txt:
-	* tests/check/elements/xingmux.c:
-	* tests/check/elements/xingmux_testdata.h:
-	* tests/check/meson.build:
-	  xingmux: move from gst-plugins-ugly to gst-plugins-good
-	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/415
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3251>
+	  Release 1.23.1
 
-2022-10-04 03:57:31 +0100  Tim-Philipp Müller <tim@centricular.com>
+2024-01-28 11:15:01 +0000  Tim-Philipp Müller <tim@centricular.com>
 
 	* meson.build:
-	  Back to development
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3115>
-
-=== release 1.21.1 ===
+	  meson: bump Meson requirement to >= 1.1 for all modules
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6002>
 
-2022-10-04 01:14:01 +0100  Tim-Philipp Müller <tim@centricular.com>
+2023-11-02 00:03:28 +0900  Seungha Yang <seungha@centricular.com>
 
-	* ChangeLog:
-	* NEWS:
-	* RELEASE:
-	* gst-plugins-ugly.doap:
 	* meson.build:
-	  Release 1.21.1
+	  meson: Fix MSVC build with GST_DISABLE_GST_DEBUG
+	  MSVC does not understand Wno-unused
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5585>
 
-2022-10-04 01:13:59 +0100  Tim-Philipp Müller <tim@centricular.com>
-
-	* ChangeLog:
-	  Update ChangeLogs for 1.21.1
-
-2022-09-21 19:19:45 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+2022-04-06 12:56:30 +0100  Tim-Philipp Müller <tim@centricular.com>
 
 	* meson.build:
-	  meson: Use implicit builtin dirs in pkgconfig generation
-	  Starting with Meson 0.62, meson automatically populates the variables
-	  list in the pkgconfig file if you reference builtin directories in the
-	  pkgconfig file (whether via a custom pkgconfig variable or elsewhere).
-	  We need this, because ${prefix}/libexec is a hard-coded value which is
-	  incorrect on, for example, Debian.
-	  Bump requirement to 0.62, and remove version compares that retained
-	  support for older Meson versions.
-	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1245
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3061>
-
-2022-08-29 10:17:45 -0400  Thibault Saunier <tsaunier@igalia.com>
-
-	* docs/meson.build:
-	  doc: Do not build plugins to build the doc
-	  It is not actually necessary
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2954>
+	  meson: Bump GLib requirement to >= 2.64
+	  This includes fixes to make GstBus watches non-racy.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2126>
 
-2021-03-24 14:20:18 -0500  Zebediah Figura <z.figura12@gmail.com>
+2023-09-13 10:12:44 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
 
-	* meson.build:
-	  meson: Build with -Wl,-z,nodelete to prevent unloading of dynamic libraries and plugins
-	  GLib made the unfortunate decision to prevent libgobject from ever being
-	  unloaded, which means that now any library which registers a static type
-	  can't ever be unloaded either (and any library that depends on those,
-	  ad nauseam).
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/778>
+	* ext/x264/meson.build:
+	  x264: Check that minimum version is met
+	  Since the addition of monochrome encoding support, we now depends on build 156.
+	  No version check was happening so far.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5325>
 
-2018-10-18 12:32:03 +0900  Yeongjin Jeong <yeongjin.jeong@navercorp.com>
+2023-06-08 14:21:07 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
 
+	* docs/gst_plugins_cache.json:
 	* ext/x264/gstx264enc.c:
-	  x264enc: Print full option-string applied to x264_encoder in debug log
-	  x264 encoder transfers all options applied to the encoder
-	  to the SEI payload when the headers(SPS, PPS) is requested.
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3014>
+	* tests/check/elements/x264enc.c:
+	  x264enc: Add monochrome 8bit support
+	  This is mapped to GStreamer GRAY8 format. This allow producing files with
+	  the chroma_format_idc 0 (monochrome). This is only availbable to high profile.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4812>
 
-2018-10-17 23:38:22 +0900  Yeongjin Jeong <yeongjin.jeong@navercorp.com>
+2023-07-07 10:08:21 +0300  Sebastian Dröge <sebastian@centricular.com>
 
-	* ext/x264/gstx264enc.c:
-	  x264enc: Don't assume the order of nal unit
-	  Just for extra safety, let's check the index of the nal unit
-	  through the enum value of the nal type provided by x264.
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3014>
+	* gst/realmedia/rmdemux.c:
+	  rmdemux: Check for integer overflow when calculation audio packet size
+	  Fixes ZDI-CAN-21444
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2782
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5072>
 
-2022-09-01 15:11:31 -0400  Thibault Saunier <tsaunier@igalia.com>
+2023-07-07 10:08:03 +0300  Sebastian Dröge <sebastian@centricular.com>
 
-	* ext/a52dec/meson.build:
-	* ext/amrnb/meson.build:
-	* ext/amrwbdec/meson.build:
-	* ext/cdio/meson.build:
-	* ext/dvdread/meson.build:
-	* ext/mpeg2dec/meson.build:
-	* ext/sidplay/meson.build:
-	* ext/x264/meson.build:
-	* gst/asfdemux/meson.build:
-	* gst/dvdlpcmdec/meson.build:
-	* gst/dvdsub/meson.build:
-	* gst/realmedia/meson.build:
-	* gst/xingmux/meson.build:
-	* meson.build:
-	  meson: Call pkgconfig.generate in the loop where we declare plugins dependencies
-	  Removing some copy pasted code
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2970>
+	* gst/realmedia/rmdemux.c:
+	  rmdemux: Use GST_LOG_OBJECT instead of GST_LOG
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5072>
 
-2022-09-01 11:51:48 -0400  Thibault Saunier <tsaunier@igalia.com>
+2023-07-07 10:00:19 +0300  Sebastian Dröge <sebastian@centricular.com>
 
-	* docs/meson.build:
-	  meson: Namespace the plugins_doc_dep/libraries variables
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2970>
+	* gst/realmedia/rmdemux.c:
+	  rmdemux: Check that enough SIPR audio data is available when copying
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5072>
 
-2022-08-31 18:44:14 -0400  Thibault Saunier <tsaunier@igalia.com>
+2023-07-07 09:59:20 +0300  Sebastian Dröge <sebastian@centricular.com>
 
-	* meson.build:
-	  meson: Rename plugins list and make them "dependency" objects
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2970>
+	* gst/realmedia/rmdemux.c:
+	  rmdemux: Check for integer overflows when calculating the size of SIPR audio buffers
+	  Fixes ZDI-CAN-21443
+	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2782
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5072>
 
-2022-05-31 15:18:03 +0200  Stéphane Cerveau <scerveau@collabora.com>
+2023-07-07 09:59:04 +0300  Sebastian Dröge <sebastian@centricular.com>
 
-	* docs/meson.build:
-	* meson.build:
-	  docs: disable in static build
-	  Following gst-plugins-base, disable docs if static_build
-	  in:
-	  - gstreamer
-	  - gst-plugins-good
-	  - gst-plugins-ugly
-	  - gst-libav
-	  - gstreamer-vaapi
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2529>
-
-2022-06-29 10:55:13 +0100  Tim-Philipp Müller <tim@centricular.com>
+	* gst/realmedia/rmdemux.c:
+	  rmdemux: Use GST_LOG_OBJECT instead of GST_LOG
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5072>
 
-	* meson.build:
-	  coding style: allow declarations after statement
-	  See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1243/
-	  and https://gitlab.freedesktop.org/gstreamer/gstreamer-project/-/issues/78
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2683>
+2023-07-05 18:46:25 -0600  Olivier Crête <olivier.crete@collabora.com>
 
-2022-06-25 19:50:10 +0100  Tim-Philipp Müller <tim@centricular.com>
+	* scripts/gen-changelog.py:
+	  gst-omx: Retire the whole package
+	  The OpenMAX standard is long dead and even the Raspberry Pi OS
+	  no longer supports it.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4976>
 
-	* tests/check/meson.build:
-	  tests: skip unit tests for dependency-less elements that have been disabled
-	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1136
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2660>
+2023-07-12 09:27:22 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
 
-2022-04-06 12:56:30 +0100  Tim-Philipp Müller <tim@centricular.com>
+	* tests/check/meson.build:
+	  meson: Always use forward slashes in defines with paths
+	  Fixes the following build failure on MSYS2:
+	  ```
+	  ../subprojects/gstreamer/tests/check/elements/filesrc.c: In function 'test_seeking':
+	  ../subprojects/gstreamer/tests/check/elements/filesrc.c:107:53: error: incomplete universal character name \U
+	  107 |   g_object_set (G_OBJECT (src), "location", TESTFILE, NULL);
+	  |                                                     ^
+	  ../subprojects/gstreamer/tests/check/elements/filesrc.c:107:53: warning: unknown escape sequence: '\A'
+	  ../subprojects/gstreamer/tests/check/elements/filesrc.c:107:53: warning: unknown escape sequence: '\g'
+	  ../subprojects/gstreamer/tests/check/elements/filesrc.c:107:53: warning: unknown escape sequence: '\s'
+	  ../subprojects/gstreamer/tests/check/elements/filesrc.c:107:53: warning: unknown escape sequence: '\g'
+	  ../subprojects/gstreamer/tests/check/elements/filesrc.c:107:53: warning: unknown escape sequence: '\c'
+	  ```
+	  Due to: `-DTESTFILE=\"C:\\Users\\Administrator\...`
+	  https://gitlab.freedesktop.org/nirbheek/gstreamer/-/jobs/45317733
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5018>
+
+2023-03-07 14:05:54 +0100  Stéphane Cerveau <scerveau@igalia.com>
+
+	* tests/meson.build:
+	  gstreamer-full: add full static support
+	  Allow a project to use gstreamer-full as a static library
+	  and link to create a binary without dependencies.
+	  Introduce the option 'gst-full-target-type' to
+	  select the build type, dynamic(default) or static.
+	  In gstreamer-full/static build configuration gstreamer (gst.c)
+	  needs the symbol gst_init_static_plugins which is defined
+	  in gstreamer-full.
+	  All the tests and examples are linking with gstreamer but the
+	  symbol gst_init_static_plugins is only defined in the gstreamer-full
+	  library. gstreamer-full can not be built first as it needs to know what plugins
+	  will be built.
+	  One option would be to build all the examples and tests after
+	  gstreamer-full as the tools.
+	  Disable tools build in subprojects too as it will be built at the end of
+	  build process.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4128>
+
+2023-05-17 22:58:46 +0200  Ruben Gonzalez <rgonzalez@fluendo.com>
 
-	* meson.build:
-	  Bump GLib requirement to >= 2.62
-	  Can't require 2.64 yet because of
-	  https://gitlab.freedesktop.org/gstreamer/cerbero/-/issues/323
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2568>
+	* README.md:
+	  README.md: fix current version
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4662>
 
-2022-03-25 10:18:34 -0400  Xavier Claessens <xavier.claessens@collabora.com>
+2023-05-10 08:57:40 +0200  Edward Hervey <edward@centricular.com>
 
-	* meson.build:
-	  Always define ENABLE_NLS
-	  GLib guarantees libintl API is always available, provided by
-	  proxy-libintl as last resort. GLib itself unconditionally define
-	  ENABLE_NLS.
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2028>
+	* ext/mpeg2dec/gstmpeg2dec.c:
+	  mpeg2dec: Fix conflicting type declaration
+	  It was always a GstFlowReturn
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4588>
 
-2022-03-25 10:20:24 -0400  Xavier Claessens <xavier.claessens@collabora.com>
+2023-04-28 16:06:24 +0100  Tim-Philipp Müller <tim@centricular.com>
 
-	* gst-libs/gst/gettext.h:
-	* gst-libs/gst/gst-i18n-plugin.h:
-	  Delete unused i18n headers
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2028>
+	* REQUIREMENTS:
+	* docs/gst_plugins_cache.json:
+	* ext/amrnb/GstAmrnbEnc.prs:
+	* ext/amrnb/README:
+	* ext/amrnb/amrnb.c:
+	* ext/amrnb/amrnbdec.c:
+	* ext/amrnb/amrnbdec.h:
+	* ext/amrnb/amrnbenc.c:
+	* ext/amrnb/amrnbenc.h:
+	* ext/amrnb/meson.build:
+	* ext/amrwbdec/README:
+	* ext/amrwbdec/amrwb.c:
+	* ext/amrwbdec/amrwbdec.c:
+	* ext/amrwbdec/amrwbdec.h:
+	* ext/amrwbdec/meson.build:
+	* ext/meson.build:
+	* meson_options.txt:
+	* tests/check/elements/amrnbenc.c:
+	* tests/check/gst-plugins-ugly.supp:
+	* tests/check/meson.build:
+	  amrnb, amrwbdec: move AMR-NB and AMR-WB plugins to -good
+	  Fedora ships these libraries as part of the main distribution now,
+	  and they are decades old anyway so don't implement any of the newer
+	  features.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4512>
 
-2022-03-25 09:59:23 -0400  Xavier Claessens <xavier.claessens@collabora.com>
+2023-03-19 15:35:29 -0300  Thibault Saunier <tsaunier@igalia.com>
 
-	* ext/cdio/gstcdiocddasrc.c:
-	* ext/dvdread/dvdreadsrc.c:
-	* gst/asfdemux/gstasf.c:
-	* gst/asfdemux/gstasfdemux.c:
-	* gst/asfdemux/gstasfelement.c:
-	  Replace gst-i18n-*.h with gi18n-lib.h
-	  GLib guarantees libintl is always present, using proxy-libintl as
-	  last resort. There is no need to mock gettex API any more.
-	  This fix static build on Windows because G_INTL_STATIC_COMPILATION must
-	  be defined before including libintl.h, and glib does it for us as part
-	  as including glib.h.
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2028>
+	* docs/meson.build:
+	  doc: Avoid shelling out to hotdoc to generate plugins config files
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4479>
 
-2022-04-15 16:53:25 +0300  Sebastian Dröge <sebastian@centricular.com>
+2023-04-11 11:50:57 -0400  Arun Raghavan <arun@asymptotic.io>
 
 	* ext/x264/gstx264enc.c:
-	  x264enc: Don't try to fixate ANY allowed caps
-	  Instead fall back to the template caps.
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2197>
-
-2022-03-30 11:06:02 -0400  Xavier Claessens <xavier.claessens@collabora.com>
-
-	* meson.build:
-	  Use gmodule-no-export-2.0
-	  We don't need `-Wl,--export-dynamic`, that's used only for executables
-	  that needs to export an API to be used by plugins they load.
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2031>
-
-2022-03-25 15:00:20 -0400  Xavier Claessens <xavier.claessens@collabora.com>
+	  x264enc: Check more video info fields to decide whether to reconfigure encoder
+	  The encoder is also initialised using interlace mode, colorimetry, chroma-site
+	  and multiview mode, so let's make sure we only skip reinitialising the encoder
+	  in set_format() if none of those have changed.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4395>
 
-	* ext/dvdread/meson.build:
-	* meson.build:
-	  Remove glib and gobject dependencies everywhere
-	  They are part of gst_dep already and we have to make sure to always have
-	  gst_dep. The order in dependencies matters, because it is also the order
-	  in which Meson will set -I args. We want gstreamer's config.h to take
-	  precedence over glib's private config.h when it's a subproject.
-	  While at it, remove useless fallback args for gmodule/gio dependencies,
-	  only gstreamer core needs it.
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2031>
+2023-03-17 16:32:45 +0200  Sebastian Dröge <sebastian@centricular.com>
 
-2022-03-18 13:42:27 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+	* tests/check/elements/x264enc.c:
+	  plugins: Fix various trivial clang compiler warnings
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4210>
 
-	* meson.build:
-	  meson: Bump all meson requirements to 0.60
-	  Lots of new warnings ever since
-	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1934
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1977>
+2023-03-15 18:53:09 +0000  Tim-Philipp Müller <tim@centricular.com>
 
-2022-02-23 11:10:11 +0100  Sebastian Fricke <sebastian.fricke@collabora.com>
+	* gst/asfdemux/gstasfdemux.c:
+	  gst-plugins-ugly: re-indent with GNU indent 2.2.12
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4182>
 
-	* README.md:
-	  Maintain build instructions at a single location
-	  Do not maintain similar build instructions within each gst-plugins-*
-	  subproject and the subproject/gstreamer subproject. Use the build
-	  instructions from the mono-repository and link to them via hyperlink.
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1743>
+2023-01-08 17:15:12 +0000  Tim-Philipp Müller <tim@centricular.com>
 
-2022-02-21 11:37:26 -0500  Xavier Claessens <xavier.claessens@collabora.com>
+	* ext/mpeg2dec/gstmpeg2dec.c:
+	  mpeg2dec: drop use of GSlice
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
 
-	* ext/amrnb/meson.build:
-	* ext/x264/meson.build:
-	  devenv: Add some missing GStreamer specific env variables
-	  This should make "meson devenv" closer to what "gst-env.py" sets.
-	  - GST_VALIDATE_SCENARIOS_PATH
-	  - GST_VALIDATE_APPS_DIR
-	  - GST_OMX_CONFIG_DIR
-	  - GST_ENCODING_TARGET_PATH
-	  - GST_PRESET_PATH
-	  - GST_PLUGIN_SCANNER
-	  - GST_PTP_HELPER
-	  - _GI_OVERRIDES_PATH
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1768>
-
-2022-02-21 23:29:23 +0100  Ruben Gonzalez <rgonzalez@fluendo.com>
+2023-01-08 17:15:00 +0000  Tim-Philipp Müller <tim@centricular.com>
 
-	* docs/gst_plugins_cache.json:
 	* ext/x264/gstx264enc.c:
-	  x264enc: Update long-name and description
-	  Update x264enc long-name to be more than just the name. Then the
-	  description also was updated to be longer than the long-name, and
-	  similar to the plugin description.
-	  Finally, as I am here, H264 was replaced by H.264 and x264 is only a
-	  plugin (not plugins).
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1771>
+	  x264enc: drop use of GSlice
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
 
-2021-10-01 20:27:28 +0900  Seungha Yang <seungha@centricular.com>
-
-	* meson.build:
-	  meson: Do hard build error for some MSVC warnings
-	  Handle various MSVC warnings as errors for development version.
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1006>
-
-2022-02-04 11:15:47 +0000  Tim-Philipp Müller <tim@centricular.com>
+2023-01-23 23:04:53 +0000  Tim-Philipp Müller <tim@centricular.com>
 
 	* meson.build:
 	  Back to development
-	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1635>
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3775>
 
-=== release 1.20.0 ===
+=== release 1.22.0 ===

gst-plugins-ugly-1.22.7.tar.xz/NEWS -> gst-plugins-ugly-1.24.5.tar.xz/NEWS Changed

@@ -1,1776 +1,1465 @@
-GStreamer 1.22 Release Notes
+GStreamer 1.24 Release Notes
 
-GStreamer 1.22.0 was originally released on 23 January 2023.
+GStreamer 1.24.0 was originally released on 4 March 2024.
 
-The latest bug-fix release in the stable 1.22 series is 1.22.7 and was
-released on 13 November 2023.
+The latest bug-fix release in the stable 1.24 series is 1.24.5 and was released on 20 June 2024.
 
-See https://gstreamer.freedesktop.org/releases/1.22/ for the latest
-version of this document.
-
-Last updated: Monday 13 November 2023, 10:00 UTC (log)
-
-Introduction
-
-The GStreamer team is proud to announce a new major feature release in
-the stable 1.x API series of your favourite cross-platform multimedia
-framework!
-
-As always, this release is again packed with many new features, bug
-fixes and other improvements.
-
-Highlights
-
--   AV1 video codec support improvements
--   New HLS, DASH and Microsoft Smooth Streaming adaptive streaming
-    clients
--   Qt6 support for rendering video inside a QML scene
--   Minimal builds optimised for binary size, including only the
-    individual elements needed
--   Playbin3, Decodebin3, UriDecodebin3, Parsebin enhancements and
-    stabilisation
--   WebRTC simulcast support and support for Google Congestion Control
--   WebRTC-based media server ingestion/egress (WHIP/WHEP) support
--   New easy to use batteries-included WebRTC sender plugin
--   Easy RTP sender timestamp reconstruction for RTP and RTSP
--   ONVIF timed metadata support
--   New fragmented MP4 muxer and non-fragmented MP4 muxer
--   New plugins for Amazon AWS storage and audio transcription services
--   New gtk4paintablesink and gtkwaylandsink renderers
--   New videocolorscale element that can convert and scale in one go for
-    better performance
--   High bit-depth video improvements
--   Touchscreen event support in navigation API
--   Rust plugins now shipped in macOS and Windows/MSVC binary packages
--   H.264/H.265 timestamp correction elements for PTS/DTS reconstruction
-    before muxers
--   Improved design for DMA buffer sharing and modifier handling for
-    hardware-accelerated video decoders/encoders/filters and
-    capturing/rendering on Linux
--   Video4Linux2 hardware accelerated decoder improvements
--   CUDA integration and Direct3D11 integration and plugin improvements
--   New H.264 / AVC, H.265 / HEVC and AV1 hardware-accelerated video
-    encoders for AMD GPUs using the Advanced Media Framework (AMF) SDK
--   applemedia: H.265 / HEVC video encoding + decoding support
--   androidmedia: H.265 / HEVC video encoding support
--   New “force-live” property for audiomixer, compositor, glvideomixer,
-    d3d11compositor etc.
--   Lots of new plugins, features, performance improvements and bug
-    fixes
-
-Major new features and changes
-
-AV1 video codec support improvements
-
-AV1 is a royalty free next-generation video codec by AOMedia and a free
-alternative to H.265/HEVC.
-
-While supported in earlier versions of GStreamer already, this release
-saw a lot of improvements across the board:
-
--   Support for hardware encoding and decoding via VAAPI/VA, AMF, D3D11,
-    NVCODEC, QSV and Intel MediaSDK. Hardware codecs for AV1 are slowly
-    becoming available in embedded systems and desktop GPUs (AMD, Intel,
-    NVIDIA), and these can now be used via GStreamer.
-
--   New AV1 RTP payloader and depayloader elements.
-
--   New encoder settings in the AOM reference encoder-based av1enc
-    element.
-
--   Various improvements in the AV1 parser and in the MP4/Matroska/WebM
-    muxers/demuxers.
-
--   dav1d and rav1e based software decoder/encoder elements shipped as
-    part of the binaries.
-
--   AV1 parser improvements and various bugfixes all over the place.
-
-Touchscreen event support in Navigation API
-
-The Navigation API supports the sending of key press events and mouse
-events through a GStreamer pipeline. Typically these will be picked up
-by a video sink on which these events happen and then the event is
-transmitted into the pipeline so it can be handled by elements inside
-the pipeline if it wasn’t handled by the application.
-
-This has traditionally been used for DVD menu support, but can also be
-used to forward such inputs to source elements that render a web page
-using a browser engine such as WebKit or Chromium.
-
-This API has now gained support for touchscreen events, and this has
-been implemented in various plugins such as the GTK, Qt, XV, and x11
-video sinks as well as the wpevideosrc element.
-
-GStreamer CUDA integration
-
--   New gst-cuda library
--   integration with D3D11 and NVIDIA dGPU NVMM elements
--   new cudaconvertscale element
-
-GStreamer Direct3D11 integration
-
--   New gst-d3d11 public library
-    -   gst-d3d11 library is not integrated with GStreamer documentation
-        system yet. Please refer to the examples
--   d3d11screencapture: Add Windows Graphics Capture API based capture
-    mode, including Win32 application window capturing
--   d3d11videosink and d3d11convert can support flip/rotation and crop
-    meta
--   d3d11videosink: New emit-present property and present signal so that
-    applications can overlay an image on Direct3D11 swapchain’s
-    backbuffer via Direct3D/Direct2D APIs. See also C++ and Rust
-    examples
--   d3d11compositor supports YUV blending/composing without intermediate
-    RGB(A) conversion to improve performance
--   Direct3D11 video decoders are promoted to GST_RANK_PRIMARY or
-    higher, except for the MPEG2 decoder
-
-H.264/H.265 timestamp correction elements
-
--   Muxers are often picky and need proper PTS/DTS timestamps set on the
-    input buffers, but that can be a problem if the encoded input media
-    stream comes from a source that doesn’t provide proper signalling of
-    DTS, such as is often the case for RTP, RTSP and WebRTC streams or
-    Matroska container files. Theoretically parsers should be able to
-    fix this up, but it would probably require fairly invasive changes
-    in the parsers, so two new elements h264timestamper and
-    h265timestamper bridge the gap in the meantime and can reconstruct
-    missing PTS/DTS.
-
-Easy sender timestamp reconstruction for RTP and RTSP
-
--   it was always possible to reconstruct and retrieve the original RTP
-    sender timestamps in GStreamer, but required a fair bit of
-    understanding of the internal mechanisms and the right property
-    configuration and clock setup.
-
--   rtspsrc and rtpjitterbuffer gained a new
-    “add-reference-timestamp-meta” property that if set puts the
-    original absolute reconstructed sender timestamps on the output
-    buffers via a meta. This is particularly useful if the sender is
-    synced to an NTP clock or PTP clock. The original sender timestamps
-    are either based on the RTCP NTP times, NTP RTP header extensions
-    (RFC6051) or RFC7273-style clock signalling.
-
-Qt6 support
-
--   new qml6glsink element for Qt6 similar to the existing Qt5 element.
-    Matching source and overlay elements will hopefully follow in the
-    near future.
-
-OpenGL + Video library enhancements
-
--   Support for new video formats (NV12_4L4, NV12_16L32S, NV12_8L128,
-    NV12_10BE_8L128) and dmabuf import in more formats (Y410, Y212_LE,
-    Y212_BE, Y210, NV21, NV61)
-
--   Improved support for tiled formats with arbitrary tile dimensions,
-    as needed by certain hardware decoders/encoders
-
--   glvideomixer: New “crop-left,”crop-right, “crop-top” and
-    “crop-bottom” pad properties for cropping inputs
-
--   OpenGL support for gst_video_sample_convert():
-
-    -   Used for video snapshotting and thumbnailing, to convert buffers
-        retrieved from appsinks or sink “last-sample” properties in
-        JPG/PNG thumbnails.
-    -   This function can now take samples and buffers backed by GL
-        textures as input and will automatically plug a gldownload
-        element in that case.
-
-High bit-depth support (10, 12, 16 bits per component value) improvements
-
--   compositor can now handle any supported input format and also mix
-    high-bitdepth (10-16 bit) formats (naively)
-
--   videoflip has gained support for higher bit depth formats.
-
--   vp9enc, vp9dec now support 12-bit formats and also 10-bit 4:4:4
-
-WebRTC
-
--   Allow insertion of bandwidth estimation elements e.g. for Google
-    Congestion Control (GCC) support
-
--   Initial support for sending or receiving simulcast streams
-
--   Support for asynchronous host resolution for STUN/TURN servers
-
--   GstWebRTCICE was split into base classes and implementation to make
-    it possible to plug custom ICE implementations
-
--   webrtcsink: batteries-included WebRTC sender (Rust)
-
--   whipsink: WebRTC HTTP ingest (WHIP) to a MediaServer (Rust)
-
--   whepsrc: WebRTC HTTP egress (WHEP) from a MediaServer (Rust)
-
--   Many other improvements and bug fixes
-
-New HLS, DASH and MSS adaptive streaming clients
-
-A new set of “adaptive demuxers” to support HLS, DASH and MSS adaptive
-streaming protocols has been added. They provide improved performance,
-new features and better stream compatibility compared to the previous
-elements. These new elements require a “streams-aware” pipeline such as
-playbin3, uridecodebin3 or urisourcebin.
-
-The previous elements’ design prevented implementing several use-cases
-and fixing long-standing issues. The new elements were re-designed from
-scratch to tackle those:
-
--   Scheduling Only 3 threads are present, regardless of the number of
-    streams selected. One in charge of downloading fragments and
-    manifests, one in charge of outputting parsed data downstream, and
-    one in charge of scheduling. This improves performance, resource
-    usage and latency.
-
--   Better download control The elements now directly control the
-    scheduling and download of manifests and fragments using libsoup
-    directly instead of depending on external elements for downloading.
-
--   Stream selection, only the selected streams are downloaded. This
-    improves bandwith usage. Switching stream is done in such a way to
-    ensure there are no gaps, meaning the new stream will be switched to
-    only once enough data for it has been downloaded.
-
--   Internal parsing, the downloaded streams are parsed internally. This
-    allows the element to fully respect the various specifications and
-    offer accurate buffering, seeking and playback. This is especially
-    important for HLS streams which require parsing for proper
-    positioning of streams.
-
--   Buffering and adaptive rate switching, the new elements handle
-    buffering internally which allows them to have a more accurate
-    visibility of which bandwith variant to switch to.
-
-Playbin3, Decodebin3, UriDecodebin3, Parsebin improvements
-
-The “new” playback elements introduced in 1.18 (playbin3 and its various
-components) have been refactored to allow more use-cases and improve
-performance. They are no longer considered experimental, so applications
-using the legacy playback elements (playbin and (uri)decodebin) can
-migrate to the new components to benefit from these improvements.
-
--   Gapless The “gapless” feature allows files and streams to be
-    fetched, buffered and decoded in order to provide a “gapless”
-    output. This feature has been refactored extensively in the new
-    components:
-    -   A single (uri)decodebin3 (and therefore a single set of
-        decoders) is used. This improves memory and cpu usage, since on
-        identical codecs a single decoder will be used.
-    -   The “next” stream to play will be pre-rolled “just-in-time”
-        thanks to the buffering improvements in urisourcebin (see below)
-    -   This feature is now handled at the uridecodebin3 level.
-        Applications that wish to have a “gapless” stream and process it
-        (instead of just outputting it, for example for transcoding,
-        retransmission, …) can now use uridecodebin3 directly. Note that
-        a streamsynchronizer element is required in that case.
--   Buffering improvements The urisourcebin element is in charge of
-    fetching and (optionally) buffering/downloading the stream. It has
-    been extended and improved:
-    -   When the parse-streams property is used (by default in
-        uridecodebin3 and playbin3), compatible streams will be demuxed
-        and parsed (via parsebin) and buffering will be done on the
-        elementary streams. This provides a more accurate handling of
-        buffering. Previously buffering was done on a best-effort basis
-        and was mostly wrong (i.e. downloading more than needed).
-    -   Applications can use urisourcebin with this property as a
-        convenient way of getting elementary streams from a given URI.
-    -   Elements can handle buffering themselves (such as the new
-        adaptive demuxers) by answering the GST_QUERY_BUFFERING query.
-        In that case urisourcebin will not handle it.
--   Stream Selection Efficient stream selection was previously only
-    possible within decodebin3. The downside is that this meant that
-    upstream elements had to provide all the streams from which to chose
-    from, which is inefficient. With the addition of the
-    GST_QUERY_SELECTABLE query, this can now be handled by elements
-    upstream (i.e. sources)
-    -   Elements that can handle stream selection internally (such as
-        the new adaptive demuxer elements) answer that query, and handle
-        the stream selection events themselves.
-    -   In this case, decodebin3 will always process all streams that
-        are provided to it.
--   Instant URI switching This new feature allows switching URIs
-    “instantly” in playbin3 (and uridecodebin3) without having to change
-    states. This mimics switching channels on a television.
-    -   If compatible, decoders will be re-used, providing lower
-        latency/cpu/memory than by switching states.
-    -   This is enabled by setting the instant-uri property to true,
-        setting the URI to switch to immediately, and then disabling the
-        instant-uri property again afterwards.
--   playbin3, decodebin3, uridecodebin3, parsebin, and urisourcebin are
-    no longer experimental
-    -   They were originally marked as ‘technology preview’ but have
-        since seen extensive usage in production settings, so are
-        considered ready for general use now.
-
-Fraunhofer AAC audio encoder HE-AAC and AAC-LD profile support
-
--   fdkaacenc:
-    -   Support for encoding to HE-AACv1 and HE-AACv2 profile
-    -   Support for encoding to AAC Low Delay (LD) profile
-    -   Advanced bitrate control options via new “rate-control”,
-        “vbr-preset”, “peak-bitrate”, and “afterburner” properties
-
-RTP rapid synchronization support in the RTP stack (RFC6051)
-
-RTP provides several mechanisms how streams can be synchronized relative
-to each other, and how absolute sender times for RTP packets can be
-obtained. One of these mechanisms is via RTCP, which has the
-disadvantage that the synchronization information is only distributed
-out-of-band and usually some time after the start.
-
-GStreamer’s RTP stack, specifically the rtpbin, rtpsession and
-rtpjitterbuffer elements, now also have support for retrieving and
-sending the same synchronization information in-band via RTP header
-extensions according to RFC6051 (Rapid Synchronisation of RTP Flows).
-Only 64-bit timestamps are supported currently.
-
-This provides per packet synchronization information from the very
-beginning of a stream and allows accurate inter-stream, and (depending
-on setup) inter-device, synchronization at the receiver side.
-
-ONVIF XML Timed Metadata support
-
-The ONVIF standard implemented by various security cameras also
-specifies a format for timed metadata that is transmitted together with
-the audio/video streams, usually over RTSP.
-
-Support for this timed metadata is implemented in the MP4 demuxer now as
-well as the new fragmented MP4 muxer and the new non-fragmented MP4
-muxer from the GStreamer Rust plugins. Additionally, the new onvif
-plugin ‒ which is part of the GStreamer Rust plugins ‒ provides general
-elements for handling the metadata and e.g. overlaying certain parts of
-it over a video stream.
-
-As part of this support for absolute UTC times was also implemented
-according to the requirements of the ONVIF standards in the
-corresponding elements.
-
-MP3 gapless playback support
-
-While MP3 can probably considered a legacy format at this point, a new
-feature was added with this release.
-
-When playing back plain MP3 files, i.e. outside a container format,
-switches between files can now be completely gapless if the required
-metadata is provided inside the file. There is no standardized metadata
-for this, but the LAME MP3 encoder writes metadata that can be parsed by
-the mpegaudioparse element now and forwarded to decoders for ensuring
-removal of padding samples at the front and end of MP3 files.
-
-“force-live” property for audio + video aggregators
-
-This is a quality of life fix for playout and streaming applications
-where it is common to have audio and video mixer elements that should
-operate in live mode from the start and produce output continuously.
-
-Often one would start a pipeline without any inputs hooked up to these
-mixers in the beginning, and up until now there was no way to easily
-force these elements into live mode from the start. One would have to
-add an initial live video or audio test source as dummy input to achieve
-this.
-
-The new “force-live” property makes these audio and video aggregators
-start in live mode without the need for any dummy inputs, which is
-useful for scenarios where inputs are only added after starting the
-pipeline.
-
-This new property should usually be used in connection with the
-“min-upstream-latency” property, i.e. you should always set a non-0
-minimum upstream latency then.
-
-This is now supported in all GstAudioAggregator and GstVideoAggregator
-subclasses such as audiomixer, audiointerleave, compositor,
-glvideomixer, d3d11compositor, etc.
-
-New elements and plugins
-
--   new cudaconvertscale element that can convert and scale in one pass
-
--   new gtkwaylandsink element based on gtksink, but similar to
-    waylandsink and uses Wayland APIs directly instead of rendering with
-    Gtk/Cairo primitives. This approach is only compatible with Gtk3,
-    and like gtksink this element only supports Gtk3.
-
--   new h264timestamper and h265timestamper elements to reconstruct
-    missing pts/dts from inputs that might not provide them such as
-    e.g. RTP/RTSP/WebRTC inputs (see above)
-
--   mfaacdec, mfmp3dec: Windows MediaFoundation AAC and MP3 decoders
-
--   new msdkav1enc AV1 video encoder element
-
--   new nvcudah264enc, nvcudah265enc, nvd3d11h264enc, and nvd3d11h265enc
-    NVIDIA GPU encoder elements to support zero-copy encoding, via CUDA
-    and Direct3D11 APIs, respectively
-
--   new nvautogpuh264enc and nvautogpuh265enc NVIDIA GPU encoder
-    elements: The auto GPU elements will automatically select a target
-    GPU instance in case multiple NVIDIA desktop GPUs are present, also
-    taking into account the input memory. On Windows CUDA or Direct3D11
-    mode will be determined by the elements automatically as well. Those
-    new elements are useful if target GPU and/or API mode (either CUDA
-    or Direct3D11 in case of Windows) is undeterminable from the encoder
-    point of view at the time when pipeline is configured, and therefore
-    lazy target GPU and/or API selection are required in order to avoid
-    unnecessary memory copy operations.
-
--   new nvav1dec AV1 NVIDIA desktop GPU decoder element
-
--   new qml6glsink element to render video with Qt6
-
--   qsv: New Intel OneVPL/MediaSDK (a.k.a Intel Quick Sync) based
-    decoder and encoder elements, with gst-d3d11 (on Windows) and gst-va
-    (on Linux) integration
-
-    -   Support multi-GPU environment, for example, concurrent video
-        encoding using Intel iGPU and dGPU in a single pipeline
-    -   H.264 / H.265 / VP9 and JPEG decoders
-    -   H.264 / H.265 / VP9 / AV1 / JPEG encoders with dynamic encoding
-        bitrate update
-    -   New plugin does not require external SDK for building on Windows
-
--   vulkanoverlaycompositor: new vulkan overlay compositor element to
-    overlay upstream GstVideoOverlayCompositonMeta onto the video
-    stream.
-
--   vulkanshaderspv: performs operations with SPIRV shaders in Vulkan
-
--   win32ipcvideosink, win32ipcvideosrc: new shared memory videosrc/sink
-    elements for Windows
-
--   wicjpegdec, wicpngdec: Windows Imaging Component (WIC) based JPEG
-    and PNG decoder elements.
-
--   Many exciting new Rust elements, see Rust section below
-
-New element features and additions
-
--   audioconvert: Dithering now uses a slightly slower, less biased PRNG
-    which results in better quality output. Also dithering can now be
-    enabled via the new “dithering-threshold” property for target bit
-    depths of more than 20 bits.
-
--   av1enc: Add “keyframe-max-dist” property for controlling max
-    distance between keyframes, as well as “enc-pass”, “keyframe-mode”,
-    “lag-in-frames” and “usage-profile” properties.
-
--   cccombiner: new “output-padding” property
-
--   decklink: Add support for 4k DCI, 8k/UHD2 and 8k DCI modes
-
--   dvbsubenc: Support for >SD resolutions is working correctly now.
-
--   fdkaacenc: Add HE-AAC / HE-AACv2 profile support
+See https://gstreamer.freedesktop.org/releases/1.24/ for the latest version of this document.
 
--   glvideomixer: New “crop-left,”crop-right, “crop-top” and
-    “crop-bottom” pad properties for cropping inputs
+Last updated: Wednesday 20 June 2024, 22:35 UTC (log)
 
--   gssink: new ‘content-type’ property. Useful when one wants to upload
-    a video as video/mp4 instead of ’video/quicktime` for example.
+## Introduction
 
--   jpegparse: Rewritten using the common parser library
+The GStreamer team is proud to announce a new major feature release in the stable 1.x API series of your favourite
+cross-platform multimedia framework!
+
+As always, this release is again packed with many new features, bug fixes and other improvements.
+
+## Highlights
+
+-   New Discourse forum and Matrix chat space
+-   New Analytics and Machine Learning abstractions and elements
+-   Playbin3 and decodebin3 are now stable and the default in gst-play-1.0, GstPlay/GstPlayer
+-   The va plugin is now preferred over gst-vaapi and has higher ranks
+-   GstMeta serialization/deserialization and other GstMeta improvements
+-   New GstMeta for SMPTE ST-291M HANC/VANC Ancillary Data
+-   New unixfd plugin for efficient 1:N inter-process communication on Linux
+-   cudaipc source and sink for zero-copy CUDA memory sharing between processes
+-   New intersink and intersrc elements for 1:N pipeline decoupling within the same process
+-   Qt5 + Qt6 QML integration improvements including qml6glsrc, qml6glmixer, qml6gloverlay, and qml6d3d11sink elements
+-   DRM Modifier Support for dmabufs on Linux
+-   OpenGL, Vulkan and CUDA integration enhancements
+-   Vulkan H.264 and H.265 video decoders
+-   RTP stack improvements including new RFC7273 modes and more correct header extension handling in depayloaders
+-   WebRTC improvements such as support for ICE consent freshness, and a new webrtcsrc element to complement webrtcsink
+-   WebRTC signallers and webrtcsink implementations for LiveKit and AWS Kinesis Video Streams
+-   WHIP server source and client sink, and a WHEP source
+-   Precision Time Protocol (PTP) clock support for Windows and other additions
+-   Low-Latency HLS (LL-HLS) support and many other HLS and DASH enhancements
+-   New W3C Media Source Extensions library
+-   Countless closed caption handling improvements including new cea608mux and cea608tocea708 elements
+-   Translation support for awstranscriber
+-   Bayer 10/12/14/16-bit depth support
+-   MPEG-TS support for asynchronous KLV demuxing and segment seeking, plus various new muxer features
+-   Capture source and sink for AJA capture and playout cards
+-   SVT-AV1 and VA-API AV1 encoders, stateless AV1 video decoder
+-   New uvcsink element for exporting streams as UVC camera
+-   DirectWrite text rendering plugin for windows
+-   Direct3D12-based video decoding, conversion, composition, and rendering
+-   AMD Advanced Media Framework AV1 + H.265 video encoders with 10-bit and HDR support
+-   AVX/AVX2 support and NEON support on macOS on Apple ARM64 CPUs via new liborc
+-   GStreamer C# bindings have been updated
+-   Rust bindings improvements and many new and improved Rust plugins
+-   Rust plugins now shipped in packages for all major platforms including Android and iOS
+-   Lots of new plugins, features, performance improvements and bug fixes
+
+## Major new features and changes
+
+### Discourse forum and Matrix chat space
+
+-   The new Discourse forum and Matrix chat space are now our preferred communication channels for support and developer chat.
+
+-   The mailing lists and IRC channel are on their way to being phased out, but Discourse can be used via e-mail as well.
+
+-   For release announcements please subscribe to the News + Announcements category on Discourse, although we will continue to
+    also send announcements to the mailing list for the time being.
+
+### Playbin3, decodebin3 now stable and default
+
+-   After a year of stability, testing and more improvements, playbin3, and its various components (uridecodebin3, decodebin3
+    and urisourcebin), are now the recommended playback components.
+
+-   Some playback components have now switched to defaulting to playbin3: gst-play-1.0 and the GstPlay / GstPlayer libraries.
+    Application developers are strongly recommended to switch to using those components instead of the legacy playbin and
+    (uri)decodebin.
+
+Improvements in this cycle:
+
+-   Better support missing/faulty decoders, attempt to use another one or gracefully un-select the stream.
+
+-   Many fixes for more complex gapless and instant-switching scenarios
+
+-   Lower latency for live pipelines
+
+-   Fix for “chained” streams (ex: Ogg, or PMT update in MPEG-TS)
+
+-   Fixes for hardware-accelerated playback with subtitles (provided the sink can handle offloading composition). This was also
+    partly due to a historical confusion between subtitle “decoders” (which decode the format to text and “parsers” (which only
+    do timing detection and optional seeking).
+
+### GstMeta serialization/deserialization and other GstMeta improvements
+
+-   GstMeta serialization/deserialization allows metas to be transmitted or stored. This is used by the unixfd and cudaipc
+    plugins for inter-process communication (IPC). Implemented so far for GstCustomMeta, GstVideoMeta, GstAudioMeta and
+    GstReferenceTimestampMeta.
+
+-   Simplified GstCustomMeta registration with gst_meta_register_custom_simple() for the simple case where tags and transform
+    functions are not needed.
+
+-   GstMetaClearFunction clears the content of the meta. This will be called by buffer pools when a pooled buffer is returned to
+    the pool.
+
+-   Add gst_meta_info_new() and gst_meta_info_register() to register a GstMeta in two steps for easier extensibility.
+
+### New unixfd plugin for efficient 1:N inter-process communication on Linux
+
+-   unixfdsink and unixfdsrc are elements that, inspired by shmsink andn shmsrc, send UNIX file descriptors (e.g. memfd, dmabuf)
+    from one sink to multiple source elements in other processes on Linux.
+
+-   The unixfdsink proposes a memfd/shm allocator to upstream elements which allows for example videotestsrc to write directly
+    into memory that can be transfered to other processes without copying.
+
+### New GstMeta for SMPTE ST-291M HANC/VANC Ancillary Data
+
+-   Previously only various specific GstMeta for ancillary data were provided, such as GstVideoCaptionMeta and GstVideoAFDMeta.
+    The new GstAncillaryMeta allows passing arbitrary ancillary data between elements, including custom and non-standard
+    ancillary data. See GstAncillaryMeta for details.
+
+-   Add with gst_buffer_add_ancillary_meta() and retrieve with gst_buffer_get_ancillary_meta() or
+    gst_buffer_iterate_ancillary_meta().
+
+-   Supported by the newly added AJA sink and source elements
+
+### DSD audio support
+
+-   DSD audio is a non-PCM raw audio format representation and the GstAudio library gained support for this in form of new
+    GstDsdInfo and GstDsdFormat API.
+
+-   Support for DSD audio has been implemented in alsasink as well as the GstAudioSink and GstAudioRingBuffer base classes, and
+    the gst-libav plugin to enable FFmpeg-based DSD elements and functionality.
+
+### Analytics and Machine Learning
+
+-   A new library, GstAnalytics, has been added. It defines a GstAnalyticsRelationMeta that can efficiently hold a large number
+    of observations from a data analysis process, for example from machine learning. It also contains a matrix of the
+    relationship between those observations.
+
+-   Three types of metadata are already defined in the library: object detection, classification and tracking.
+
+-   A new objectdetectionoverlay element has been merged that draws the bounding boxes and the classes from the object detection
+    and classification metadata types.
+
+-   The onnxinference element has been split into two parts. The first part works with the ONNX Runtime library to do the actual
+    inference, while the second part called ssdobjectdetector interprets the produced tensor. This new element creates
+    GstAnalyticsRelationMeta.
+
+-   The onnxinference element now accepts video frames without transformation if the module declares that it accepts the “Image”
+    type and the format is something that GStreamer knows.
+
+-   In the next release, tensor decoders such as ssdobjectdetector will live outside of the ONNX plugin so they can be used with
+    other machine learning acceleration frameworks.
+
+### Qt5 + Qt6 QML integration improvements
+
+-   The Qt5 qmlglsink, qmlgloverlay, qmlglmixer received support for directly consuming BGRA and YV12 video frames without a
+    prior glcolorconvert.
+
+-   New qml6glsrc, qml6glmixer, and qml6gloverlay elements as Qt6 counterparts to the existing Qt5 elements, also with support
+    for directly consuming BGRA and YV12 video frames without a prior glcolorconvert.
+
+-   qml6d3d11sink is a new Direct3D11 Qt6 QML sink for Windows as an alternative to the existing qml6glsink.
+
+### DRM Modifier Support for dmabufs on Linux
+
+The Linux dmabuf subsystem provides buffer sharing across different hardware device drivers and subsystems, and is used
+extensively by the DRM subsystem to exchange buffers between processes, contexts, and library APIs within the same process, and
+also to exchange buffers with other subsystems such as Linux Media.
+
+In GStreamer, it’s used on the capture side (v4l2src, pipewire), hardware-accelerated video decoders and encoders, OpenGL
+integration, Wayland renderers, etc.
+
+GStreamer has had support for dmabufs for a long time and was able to negotiate “zero-copy” paths between different components,
+however it only supported and assumed simple linear formats and was not able to negotiate complex non-linear formats. This meant
+that dmabuf support actually had to be disabled in many scenarios to avoid “garbled video”.
+
+With GStreamer 1.24 there is now full DRM modifier support and complex non-linear formats can be supported and negotiated
+between components.
+
+This is achieved with an extra drm_format field in video/x-raw(memory:DMABuf), format=(string)DMA_DRM caps, e.g.
+drm-format=(string)NV12:0x0x0100000000000001.
+
+See the GStreamer DMA buffers design documentation for more details.
+
+This is used in the VA-API va plugin, waylandsink, the MSDK plugin, and the OpenGL integration. Video4Linux support is expected
+to land in one of the next minor releases.
+
+New API has been added for easy handling of these new caps:
+
+-   GstVideoInfoDmaDrm plus associated functions, similar to GstVideoInfo, including conversion to and from GstVideoInfo with
+    gst_video_info_dma_drm_from_video_info() and gst_video_info_dma_drm_to_video_info()
+
+-   GST_VIDEO_DMA_DRM_CAPS_MAKE
+
+-   GST_VIDEO_FORMAT_DMA_DRM
+
+### OpenGL integration enhancements
+
+-   When using EGL, if both OpenGL ES and OpenGL are available, OpenGL ES is preferred over OpenGL. OpenGL ES supports some
+    necessary features required for dmabuf support. This does not apply if an external library/application chooses an OpenGL API
+    first.
+
+-   Improved support for dmabuf use cases. The glupload element now supports the new and improved dmabuf negotiation with
+    explicit modifiers.
+
+-   Base classes for mixing with OpenGL are now public API. GstGLBaseMixer and GstGLMixer are exposed matching the existing
+    filter-based GstGLBaseFilter and GstGLFilter base classes. The new OpenGL mixer base classes are based on
+    GstVideoAggregator.
+
+-   Add support for a ‘surfaceless’ EGL context using EGL_MESA_platform_surfaceless.
+
+-   Expose Vivante Framebuffer build-related files (pkg-config, gir) as public API
+
+-   Add support for more video formats:
+
+    -   A420 8/10/12/16-bit.
+    -   A422 8/10/12/16-bit.
+    -   A444 8/10/12/16-bit.
+    -   I420 10/12 bit.
+    -   RBGA.
+
+-   Add support for tiled video formats
+
+    -   NV12_16L32S (Mediatek format)
+    -   NV12_4L4 (Verisilicon Hantro format)
+
+-   glcolorconvert now has API for retrieving shader strings for:
+
+    -   swizzling (reordering components).
+    -   YUV->RGB conversion.
+    -   RGB->YUV conversion.
+
+-   Add more helpers for information about a particular video and/or GL format e.g. number of components, bytes used, or pixel
+    ordering.
+
+-   glvideomixer has new sink pad properties sizing-policy, xalign, yalign matching compositor.
+
+-   GstGLBufferPool now has a configuration option for allowing a number of buffers to be always outstanding allowing for
+    reducing the potential synchronisation delay when reusing OpenGL memory backed buffers.
+
+### Vulkan integration enhancements
+
+-   Add support for the Vulkan H.264 and H.265 decoders.
+
+-   Add support for timeline semaphores.
+
+-   Optionally use newer Vulkan functions for format selection.
+
+-   Add support for GPU-assisted validation.
+
+-   Vulkan/Wayland: add support for xdg_wm_base protocol for creating a visible debug window. Required as the previous wl_shell
+    interface is being removed from compositors.
+
+### CUDA / NVCODEC integration and feature additions
+
+-   New cudaipcsrc and cudaipcsink elements for zero-copy CUDA memory sharing between processes
+
+-   New nvJPEG library based nvjpegenc JPEG encoder element
+
+-   The NVIDIA desktop GPU decoder nvh264sldec, nvh265sldec, nvvp8sldec, and nvvp9sldec elements were renamed to nvh264dec,
+    nvh265dec, nvvp8dec, and nvvp9dec, respectively.
+
+-   GStreamer NVIDIA H.264 and H.265 encoders except for nvh264enc and nvh265enc gained support for CEA708 Closed Caption
+    inserting.
+
+-   OpenGL memory support is added to nv{cuda,autogpu}h264enc and nv{cuda,autogpu}h265enc elements
+
+-   CUDA stream integration: As of 1.24, CUDA stream synchronization is an application’s responsibility, and GStreamer will not
+    execute unnecessary synchronization operations. If an application needs direct access to CUDA memory via GST_MAP_CUDA map
+    flag, cuStreamSynchronize() or gst_cuda_memory_sync() call is required unless application-side CUDA operation is executed
+    with the GstCudaMemory’s associated CUDA stream.
+
+### RTP stack improvements
+
+-   New rtppassthroughpay element which just passes RTP packets through unchanged, but appears like an RTP payloader element.
+    This is useful for relaying an RTP stream as-is through gst-rtsp-server, which expects an RTP payloader with certain
+    properties at the end of an RTSP media sub-pipeline.
+
+-   New “timeout-inactive-rtp-sources” property on rtpbin, sdpdemux and rtpsession to allow applications to disable automatic
+    timeout of sources from which no data has been received for a while.
+
+-   rtpvp8pay, rtpvp9pay: expose “picture-id” as a property, and add a “picture-id-offset” property to the VP9 payloader to
+    bring it in line with the VP8 payloader.
+
+-   rtpjitterbuffer has seen improved media clock handling (clock equality and clock setting/resetting), as well as two new
+    properties that allow reconstruction of absolute PTP timestamps without actually syncing to the PTP clock, which can be
+    useful in scenarios where one wants to reconstruct the absolute PTP clock timestamps on a machine that doesn’t have access
+    to the network of the PTP clock provider. The two new properties are:
+
+    -   “rfc7273-use-system-clock”: allows the jitter buffer to assume that the system clock is synced sufficiently close to the
+        media clock used by an RFC7273 stream. By default the property is disabled and the jitter buffer will create a media
+        clock and try to sync to it, but this is only required to determine in which wraparound period from the media clock’s
+        Epoch the current RTP timestamps refer to (and thus to reconstruct absolute time stamps from them). If the property is
+        enabled the wraparound period and current offset from the Epoch will be determined based on the local system clock,
+        which means that no direct network connection to the media clock provider is needed to reconstruct absolute timestamps.
+        There is also no start-up delay, because there’s no clock sync that needs to be established first.
+
+    -   “rfc7273-reference-timestamp-meta-only”: If this property is enabled then the jitter buffer will do the normal timestamp
+        calculations for the output buffers according to the configured mode instead of making use of the RFC7273 media clock
+        for that. It will still calculate RFC7273 media clock timestamps, but only attach them to the output buffers in form of
+        a clock reference meta.
+
+-   RTP payloaders and depayloaders now have an “extensions” property for retrieving the list of currently enabled RTP header
+    extensions.
+
+-   rtpbin and webrtcbin no longer blindly set properties on the jitter buffer assuming it’s a standard rtpjitterbuffer, but
+    instead check if the property is available first, to better support non-standard jitterbuffers or even an identity element
+    in lieu of a jitter buffer.
+
+-   RTP header extension handling fixes for depayloaders that aggregate multiple input buffers into a single output buffer.
+    Before, only the last RTP input buffer was checked for header extensions. Now the depayloader remembers all RTP packets
+    pushed before an output buffer is produced and checks all RTP input buffers for header extensions.
+
+    -   Affected depayloaders: rtph264depay, rtph265depay, rtpvp8depay, rtpvp9depay, rtpxqtdepay, rtpasfdepay, rtpmp4gdepay,
+        rtpsbcdepay, rtpvorbisdepay, rtpmp4vdepay, rtptheoradepay, rtpsv3vdepay, rtpmp4adepay, rtpklvdepay, rtpjpegdepay,
+        rtpj2kdepay, rtph263pdepay, rtph263depay, rtph261depay. rtpgstdepay.
+
+### WebRTC improvements
+
+-   Add support for ICE consent freshness (RFC 7675). This requires libnice >= 0.1.22.
+
+-   Advertise the local side of an end-of-candidates with an empty candidate string.
+
+-   Add the number of Data Channels opened and closed to webrtcbin’s statistics.
+
+-   Various improvements and feature additions in the Rust webrtc plugin, which provides webrtcsrc and webrtcsink elements as
+    well as specific elements for different WebRTC signalling protocols. See the Rust plugins section below for more details.
+
+### Adaptive Streaming improvements and Low-Latency HLS (LL-HLS) support
+
+-   hlsdemux2 now supports Low-Latency HLS (LL-HLS)
+
+-   hlsdemux2 asynchronous playlist download and update improves responsiveness and bandwith usage.
+
+-   hlsdemux2 handles fallback variant URLs.
+
+-   hlsdemux2 is more responsive and accurate when handling seeks.
+
+-   dashdemux2 and hlsdemux2 have a new “start-bitrate” property, improving the decision for which initial stream variant that
+    will be used.
+
+-   dashdemux2, hlsdemux2, mssdemux2 have received many improvements regarding seeking, along with support for “early-seek”
+    which allows playback to start immediately from the requested position without any previous download.
+
+-   dashdemux2, hlsdemux2, mssdemux2 better handle errors on or near the live edge.
+
+-   dashsink can now use the dashmp4mux muxer from the Rust plugins and will also produce better and RFC 6381-compatible codec
+    strings. The “suggested-presentation-delay” property allows to set the suggested presentation delay in the MPD.
+
+-   No development took place on the legacy demuxers (dashdemux, hlsdemux, mssdemux). Application developers are reminded to use
+    the new demuxers instead. They are automatically picked up when using urisourcebin, uridecodebin3 or playbin3.
+
+### W3C Media Source Extensions library
+
+-   A new GStreamer library (mse) implementing the W3C Media Source Extensions specification was added.
+
+-   Applications can embed this library along with GStreamer in order to integrate software that uses the Media Source APIs
+    without relying on a web browser engine. Typically an application consuming this library will wrap the C API with JavaScript
+    bindings that match the Media Source API so their existing code can integrate with this library.
+
+### Closed Caption handling improvements
+
+-   ccconverter supports converting between the two CEA-608 fields.
+
+-   New cea608mux element for muxing multiple CEA-608 streams together.
+
+-   Various improvements and feature additions in the Rust-based closed caption elements. Check out the Rust plugins section
+    below for more details.
+
+### Precision Time Protocol (PTP) clock improvements
+
+-   Many fixes and compatibility/interoperability improvements.
+
+-   Better support for running on devices with multiple network interfaces.
+
+-   Allow sync to master clock on same host.
+
+-   PTP clock support is now also available on Windows.
+
+-   The standalone ptp-helper binary has been rewritten in Rust for portability and security. This works on Linux, Android,
+    Windows, macOS, FreeBSD, NetBSD, OpenBSD, DragonFlyBSD, Solaris and Illumos. Newly supported compared to the C version is
+    Windows. Compared to the C version various error paths are handled more correctly and a couple of memory leaks are fixed.
+    Otherwise it should work identically. The minimum required Rust version for compiling this is 1.48, i.e. the version
+    currently in Debian oldstable. On Windows, Rust 1.54 is needed at least.
+
+-   New ptp-helper Meson build option so PTP support can be disabled or required.
+
+-   gst_ptp_init_full() allows for a more fine-grained and extensible configuration and initialization of the GStreamer PTP
+    subsystem, including TTL configuration.
+
+### Bayer 10/12/14/16-bit depth support
+
+-   bayer2rgb and rgb2bayer now support bayer with 10/12/14/16 bit depths
+
+-   v4l2src and videotestsrc now support bayer with 10/12/14/16 bit depths
+
+-   imagefreeze gained bayer support as well
+
+### MPEG-TS improvements
+
+-   mpegtsdemux gained support for
+    -   segment seeking for seamless non-flushing looping, and
+    -   synchronous KLV
+-   mpegtsmux now
+    -   allows attaching PCR to non-PES streams
+    -   allows setting of the PES stream number for AAC audio and AVC video streams via a new “stream-number” property on the
+        muxer sink pads. Currently the PES stream number is hard-coded to zero for these stream types.
+    -   allows writing arbitrary Opus channel mapping families and up to 255 channels
+    -   separate handling of DVB and ATSC AC3 descriptors
+
+## New elements and plugins
+
+-   analyticsoverlay visualises object-detection metas on a video stream.
+
+-   autovideoflip and autodeinterlace are two new auto elements.
+
+-   AJA source and sink elements plus device provider for AJA capture and playout cards, including support for HANC/VANC
+    ancillary data.
+
+-   New cea608mux element for muxing multiple CEA-608 streams together.
+
+-   The codec2json plugin adds av12json, h2642json, h2652json and vp82json elements which convert AV1, H.264, H.265 and VP8
+    frame parameters into human readable JSON data, which is useful for debugging and testing purposes.
+
+-   New lc3 plugin with a decoder and encoder for the Bluetooth LC3 audio codec.
+
+-   New onnxinference element to run ONNX inference models on video buffers.
+
+-   New rtppassthroughpay element which just passes RTP packets through unchanged, but appears like an RTP payloader element.
+    This is mostly useful for medias that simply pass through an existing RTP stream in gst-rtsp-server.
+
+-   Qt6: qml6glsrc, qml6glmixer, qml6gloverlay, and qml6d3d11sink
+
+-   New SVT-AV1 encoder plugin, imported from SVT-AV1 but with many fixes.
+
+-   Many exciting new Rust elements, see Rust section below.
+
+-   New DirectWrite text rendering and Direct3D12 plugins (see Windows section below).
+
+-   New vaav1enc element for encoding video in AV1 (See VA-API section)
+
+-   New uvcsink element for exporting streams as UVC camera
+
+## New element features and additions
+
+-   alphacombine supports I420_10LE now for 10-bit WebM/alpha support.
+
+-   The amfcodec for hardware-accelerated video encoding using the Advanced Media Framework (AMF) SDK for AMD GPUs gained some
+    new features:
+
+    -   10-bit and HDR support for H.265 / HEVC and AV1 video encoders
+    -   B-frame support in the H.264 encoder
+    -   Initial support of pre-analysis and pre-encoding
+    -   Initial support of Smart Access Video for optimal distribution amongst multiple AMD hardware instances.
+
+-   appsink: new “propose-allocation” signal so applications can provide a buffer pool or allocators to the upstream elements,
+    as well as “max-time” and “max-buffers” properties to configure the maximum size of the appsink-internal queue in addition
+    to the existing “max-bytes” property.
+
+-   autovideoconvert exposes colorspace and scaler elements for well know elements
+
+-   avtp: add AVTP Raw Video Format payload and de-payload support.
+
+-   cacasink’s output driver can now be selected via the “driver” property.
+
+-   camerabin: various fixes and stability improvements
+
+-   clocksync: “QoS” property to optionally send QoS events upstream like a synchronising sink would.
+
+-   cutter: can add GstAudioLevelMeta on output buffers, which can be enabled via the new “audio-level-meta” property.
+
+-   dashdemux2 has a new “start-bitrate” property.
+
+-   dashsink can now use the dashmp4mux muxer from the Rust plugins and will also produce better and RFC 6381-compatible codec
+    strings. The “suggested-presentation-delay” property allows to set the suggested presentation delay in the MPD.
+
+-   deinterlace: Add support for 10/12/16-bit planar YUV formats
+
+-   The dvdspu subpicture overlay now implements GstVideoOverlayComposition support to make it work better with hardware
+    decoders where the video data should ideally stay on the GPU/VPU and the overlay blitting be delegated to the renderer.
+
+-   encodebin now automatically autoplugs timestamper elements such as h264timestamper or h265timestamper, based on new
+    “Timestamper” element factory type and rank.
+
+-   New fakevideodec element (see debugging section below).
+
+-   filesink: “file-mode” property to allow the ability to specify rb+ file mode, which overwrites an existing file. This is
+    useful in combination with splitmuxsink so that files can be pre-allocated which can be useful to reduce disk fragmentation
+    over time.
+
+-   flvmux: add “enforce-increasing-timestamps” property to allow disabling a hack that was added back in the day because
+    librtmp as used in rtmpsink would get confused by timestamps going backwards, but this is no longer required with rtmpsink2.
+    If set to true (still the default, for backwards compatibility), flvmux will modify buffers timestamps to ensure they are
+    always strictly increasing, inside one stream and also between the audio and video streams.
+
+-   giostreamsink: Add a property to close the stream on stop().
+
+-   h264parse improved its AU boundary detection.
+
+-   h264parse, h265parse, mpegvideoparse now support multiple unregistered user data SEI messages.
+
+-   insertbin is now a registered element and available via the registry, so can be constructed via parse-launch and not just
+    via the insertbin API.
+
+-   jack: libjack is now loaded dynamically at runtime instead of linking it at build time. This means the plugin can be shipped
+    on Windows and macOS and will work if there’s a user-installed JACK server/library setup.
+
+-   jpegparse now has a rank so it will be autoplugged if needed.
+
+-   kmssink: Add auto-detection for NXP i.MX8M Plus LCDIFv3, ST STM32 LTDC, and Texas Instruments TIDSS display controllers.
+
+-   matroskademux and matroskamux gained support for more raw video formats, namely RGBA64_LE, BGRA64_LE, GRAY10_BE32, GRAY16_LE
+
+-   mpg123audiodec’s rank was changed from MARGINAL to PRIMARY so it’s now higher than avdec_mp3, as it works better with
+    “freeformat” MP3s.
 
 -   msdk:
 
-    -   new msdkav1enc AV1 video encoder element
-    -   msdk decoders: Add support for Scaler Format Converter (SFC) on
-        supported Intel platforms for hardware accelerated conversion
-        and scaling
-    -   msdk encoders: support import of dmabuf, va memory and D3D11
-        memory
-    -   msdk encoders: add properties for low delay bitrate control and
-        max frame sizes for I/P frames
-    -   msdkh264enc, msdkh265enc: more properties to control intra
-        refresh
-    -   note that on systems with multi GPUs the Windows D3D11
-        integration might only work reliably if the Intel GPU is the
-        primary GPU
+    -   DRM modifier support on Linux
 
--   mxfdemux: Add support for Canon XF-HEVC
+    -   only expose codecs and capabilities actually supported by the platform
 
--   openaptx: Support the freeaptx library
+    -   msdkvpp video post-processing:
 
--   qroverlay:
+        -   new “hdr-tone-mapping” property to enable HDR-to-SDR tone mapping
+        -   new compute scaling mode
 
-    -   new “qrcode-case-sensitive” property allows encoding case
-        sensitive strings like wifi SSIDs or passwords.
-    -   added the ability to pick up data to render from an
-        upstream-provided custom GstQROverlay meta
+    -   Decoders sport D3D11 and VA integration, and the VP9 decoder supports certain resolution changes.
 
--   qtdemux: Add support for ONVIF XML Timed MetaData and AVC-Intra
-    video
+    -   Encoders:
 
--   rfbsrc now supports the uri handler interface, so applications can
-    use RFB/VNC sources in uridecodebin(3) and playbin, with
-    e.g. rfb://:password@10.1.2.3:5903?shared=1
+        -   msdkh264enc, msdkh265enc**: “pic-timing-sei” property to insert pic timing SEI
+        -   msdkh264enc, msdkh265enc**: Add properties to allow different max/min-qp values for I/P/B frames
+        -   msdkh264enc: Added BGRx format DMABuf support
+        -   Advertise special image formats in low power mode
 
--   rtponviftimestamp: Add support for using reference timestamps
+-   mxfdemux gained support for FFV1 demuxing
 
--   rtpvp9depay now has the same keyframe-related properties as
-    rtpvp8depay and rtph264depay: “request-keyframe” and
-    “wait-for-keyframe”
+-   opusenc, opusdec now support decoding and encoding more than 8 channels, and can also handle unknown/unpositioned channel
+    layouts.
 
--   rtspsrc: Various RTSP servers are using invalid URL operations for
-    constructing the control URL. Until GStreamer 1.16 these worked
-    correctly because GStreamer was just appending strings itself to
-    construct the control URL, but starting version 1.18 the correct URL
-    operations were used. With GStreamer 1.22, rtspsrc now first tries
-    with the correct control URL and if that fails it will retry with
-    the wrongly constructed control URL to restore support for such
-    servers.
+-   The oss plugin gained a device provider for audio device discovery
 
--   rtspsrc and rtpjitterbuffer gained a new
-    “add-reference-timestamp-meta” property that makes them put the
-    unmodified original sender timestamp on output buffers for NTP or
-    PTP clock synced senders
+-   pcapparse learned how to handle the Linux “cooked” capture encapsulation v2
 
--   srtsrc, srtsink: new “auto-reconnect” property to make it possible
-    to disable automatic reconnects (in caller mode) and make the
-    elements post an error immediately instead; also stats improvements
+-   Intel Quick Sync plugin improvements:
 
--   srtsrc: new “keep-listening” property to avoid EOS on disconnect and
-    keep the source running while it waits for a new connection.
+    -   qsvh264enc gained more encoding options
+    -   qsvh265dec now supports GBR decoding and HEVC RExt profiles
 
--   videocodectestsink: added YUV 4:2:2 support
+-   qtdemux now adds audio clipping meta when playing gapless m4a content, supports CENC sample grouping, as well as the SpeedHQ
+    video codec.
 
--   wasapi2src: Add support for process loopback capture
+-   ristsrc gained support for dynamic payloads via the new “caps” and “encoding-name” properties. These can be used to make the
+    ristsrc receive other payload types than MPEG-TS.
 
--   wpesrc: Add support for modifiers in key/touch/pointer events
+-   rtmp2src: a new “no-eof-is-error” property was added: There is currently no way for applications to know if the stream has
+    been properly terminated by the server or if the network connection was disconnected, as an EOS is sent in both cases. With
+    the property set, connection errors will be reported as errors, allowing applications to distinguish between both scenarios.
 
-Plugin and library moves
+-   rtspsrc: new “extra-http-request-headers” property for adding custom http request headers when using http tunnelling.
 
--   The xingmux plugin has been moved from gst-plugins-ugly into
-    gst-plugins-good.
+-   sdpdemux now supports SDP source filters as per RFC 4570; audio-only or video-only streaming can be selected via the new
+    “media” property, and RTCP feedback can be disabled via the “rtcp-mode” property.
 
--   The various Windows directshow plugins in gst-plugins-bad have been
-    unified into a single directshow plugin.
+-   splitmuxsrc uses natural ordering to sort globbed filenames now, i.e. 0, 1, 2, 10, 11 instead of 0, 1, 10, 11, 2, …
 
-Plugin removals
+-   srt: Add more fields to the statistics to see how many packets were retransmitted and how many were dropped.
 
--   The dxgiscreencapsrc element has been removed, use
-    d3d11screencapturesrc instead
+-   switchbin: many improvements, especially for caps handling and passthrough.
 
-Miscellaneous API additions
+-   taginject: a “scope” property was added to allow injection of global tags in addition to the current default which is stream
+    tags.
 
--   GST_AUDIO_FORMAT_INFO_IS_VALID_RAW() and
-    GST_VIDEO_FORMAT_INFO_IS_VALID_RAW() can be used to check if a
-    GstAudioFormatInfo or GstVideoFormatInfo has been initialised to a
-    valid raw format.
+-   timeoverlay: add buffer-count and buffer-offset time modes.
 
--   Video SEI meta: new GstVideoSEIUserDataUnregisteredMeta to carry
-    H.264 and H.265 metadata from SEI User Data Unregistered messages.
+-   udpsrc: new “multicast-source” property to support IGMPv3 Source Specific Muliticast (SSM) as per RFC 4604.
 
--   vulkan: Expose gst_vulkan_result_to_string()
+-   videoconvertscale, videoconvert: add a “converter-config” property to allow fine-tuning conversion parameters that are not
+    exposed directly as property.
 
-Miscellaneous performance, latency and memory optimisations
+-   videoflip: many orientation tag handling fixes and improvements
 
--   liborc 0.4.33 adds support for aarch64 (64-bit ARM) architecture
-    (not enabled by default on Windows yet though) and improvements for
-    32-bit ARM and should greatly enhance performance for certain
-    operations that use ORC.
+-   videorate: add “drop-out-of-segment” property to force dropping of out-of-segment buffers.
 
--   as always there have been plenty of performance, latency and memory
-    optimisations all over the place.
+-   volume now supports arbitrarily-large positive gains via a new “volume-full-range” property (it was not possibly to just
+    allow a bigger maximum value for the existing “volume” property for GstController-related backwards-compatibility reasons).
 
-Miscellaneous other changes and enhancements
+-   waylandsink, gtkwaylandsink: improved frame scheduling reducing frame drops and improve throughput.
 
--   the audio/video decoder base classes will not consider decoding
-    errors a hard error by default anymore but will continue trying to
-    decode. Previously more than 10 consecutive errors were considered a
-    hard error but this caused various partially broken streams to fail.
-    The threshold is configurable via the “max-errors” property.
+-   webpenc now has support for animated WebP which can be enabled via the new “animated” property. By default it will just
+    output a stand-alone WebP image for each input buffer, just like before.
 
--   compatibility of the GStreamer PTP clock implementation with
-    different PTP server implementations was improved, and
-    synchronization is achieved successfully in various scenarios that
-    failed before.
+-   wpe: added a WebProcess crash handler; gained WPEWebKit 2.0 API support.
 
-Tracing framework and debugging improvements
+-   x264enc gained support for 8-bit monochrome video (GRAY8).
 
-New tracers
+-   ximagesrc gained navigation support (mouse and keyboard events).
 
--   buffer-lateness: Records lateness of buffers and the reported
-    latency for each pad in a CSV file. Comes with a script for
-    visualisation.
+-   y4mdec now parses extended headers to support high bit depth video.
 
--   pipeline-snapshot: Creates a .dot file of all pipelines in the
-    application whenever requested via SIGUSR1 (on UNIX systems)
+## Plugin and library moves
 
--   queue-levels: Records queue levels for each queue in a CSV file.
-    Comes with a script for visualisation.
+-   The AMR-NB and AMR-WB plugins have been moved from -ugly to -good.
 
-Debug logging system improvements
+## Plugin and element removals
 
--   new log macros GST_LOG_ID, GST_DEBUG_ID, GST_INFO_ID,
-    GST_WARNING_ID, GST_ERROR_ID, and GST_TRACE_ID allow passing a
-    string identifier instead of a GObject. This makes it easier to log
-    non-gobject-based items and also has performance benefits.
+-   The entire gst-omx package and plugin has been retired. See the OMX section below for more details.
 
-Tools
+-   The RealServer RTSP extension, RDT handling and PNM source have been removed from the realmedia plugin.
 
--   gst-play-1.0 gained a --no-position command line option to suppress
-    position/duration queries, which can be useful to reduce debug log
-    noise.
+-   The kate subtitle plugin has been removed.
 
-GStreamer FFMPEG wrapper
+## Miscellaneous API additions
 
--   Fixed bitrate management and timestamp inaccuracies for video
-    encoders
+GStreamer Core
 
--   Fix synchronization issues and errors created by the (wrong)
-    forwarding of upstream segment events by ffmpeg demuxers.
+-   gst_pipeline_get_configured_latency() and gst_pipeline_is_live() convenience functions to query liveness and configured
+    latency of a pipeline.
 
--   Clipping meta support for gapless mp3 playback
+-   Plugins can now provide status info messages for plugins that will be displayed in gst-inspect-1.0 and is useful for dynamic
+    plugins that register features at runtime. They are now able to provide information to the user why features might not be
+    available. This is now used in the amfcodec, nvcodec, qsv, and va plugins.
 
-GStreamer RTSP server
+-   GST_OBJECT_AUTO_LOCK() and GST_PAD_STREAM_AUTO_LOCK() are g_autoptr(GMutexLocker)-based helpers for GstPad and GstObject
+    that unlock the mutex automatically when the helper goes out of scope. This is not portable so should not be used in
+    GStreamer code that needs to be portable to e.g. Windows with MSVC.
 
--   Add RFC5576 Source-specific media attribute to the SDP media for
-    signalling the CNAME
+-   gst_clear_context(), gst_clear_promise(), gst_clear_sample()
 
--   Add support for adjusting request response on pipeline errors
+-   gst_util_ceil_log2() and gst_util_simplify_fraction() utility functions
 
-    -   Give the application the possibility to adjust the error code
-        when responding to a request. For that purpose the pipeline’s
-        bus messages are emitted to subscribers through a
-        “handle-message” signal. The subscribers can then check those
-        messages for errors and adjust the response error code by
-        overriding the virtual method
-        GstRTSPClientClass::adjust_error_code().
+-   New TAG_CONTAINER_SPECIFIC_TRACK_ID tag for container specific track ID as used in an HTML5 context, plus basic support in
+    matroskademux, qtdemux, dashdemux and dashdemux2
 
--   Add gst_rtsp_context_set_token() method to make it possible to set
-    the RTSPToken on some RTSPContext from bindings such as the Python
-    bindings.
+-   New utility functions to create a stream-id without a pad for elements:
 
--   rtspclientsink gained a “publish-clock-mode” property to configure
-    whether the pipeline clock should be published according to RFC7273
-    (RTP Clock Source Signalling), similar to the same API on
-    GstRTSPMedia.
+    -   gst_element_decorate_stream_id()
+    -   gst_element_decorate_stream_id_printf_valist()
+    -   gst_element_decorate_stream_id_printf()
 
-GStreamer VA-API support
+-   GstQueueArray gained API for sorting and sorted insertion
 
--   Development activity has shifted towards the new va plugin, with
-    gstreamer-vaapi now basically in maintenance-only mode. Most of the
-    below refers to the va plugin (not gstreamer-vaapi).
+-   Add strict GstStructure serialisation with gst_structure_serialize_full() in combination with GST_SERIALIZE_FLAG_STRICT
+    which only succeeds if the result can later be fully deserialised again.
 
--   new gst-va library for GStreamer VA-API integration
+-   GstBaseSrc enhancements: the “automatic-eos” property can be used to do the equivalent to gst_base_src_set_automatic_eos().
+    gst_base_src_push_segment() sends a segment event right away which can be useful for subclasses like appsrc which have their
+    own internal queuing.
 
--   vajpegdec: new JPEG decoder
+-   GstBaseSink gained a new custom GST_BASE_SINK_FLOW_DROPPED flow return which can be used by subclasses from the virtual
+    ::render method to signal to the base class that a frame is not being rendered. This is used in e.g. waylandsink and ensures
+    that elements such as fpsdisplaysink will correctly report the rate of frames rendered and dropped.
 
--   vah264enc, vah265enc: new H.264/H.265 encoders
+GstDiscoverer
 
--   vah264lpenc, vah265lpenc: new low power mode encoders
+-   New “load-serialized-info” signal to retrieve a serialized GstDiscovererInfo
 
--   vah265enc: Add extended formats support such as 10/12 bits, 4:2:2
-    and 4:4:4
+GstSDP
 
--   Support encoder reconfiguration
+-   Add gst_sdp_message_remove_media()
 
--   vacompositor: Add new compositor element using the VA-API VPP
-    interface
+Video Library
 
--   vapostproc:
+DRM Modifier Support for dmabufs on Linux
 
-    -   new “scale-method” property
-    -   Process HDR caps if supported
-    -   parse video orientation from tags
+See section above.
 
--   vaapipostproc: Enable the use of DMA-Buf import and export
-    (gstreamer-vaapi)
+List of Video Formats for Passthrough
 
-GStreamer Video4Linux2 support
+New helper API was added to get a list of all supported video formats, including DMA_DRM formats, and can be used to advertise
+all supported formats for passthrough purposes:
 
--   Added support for Mediatek Stateless CODEC (VP8, H.264, VP9)
+-   GST_VIDEO_FORMATS_ANY_STR, GST_VIDEO_FORMATS_ANY
+-   gst_video_formats_any() which can be used by bindings or for code that prefers GstVideoFormat values instead of strings.
 
--   Stateless H.264 interlaced decoder support
+New Video Formats
 
--   Stateless H.265 decoder support
+-   12-bit and 16-bit A420 / A422 / A444 (YUV with alpha channel) variants:
 
--   Stateful decoder support for driver resolution change events
+    -   A444_12BE, A444_12LE
+    -   A422_12BE, A422_12LE
+    -   A420_12BE, A420_12LE
+    -   A444_16BE, A444_16LE
+    -   A422_16BE, A422_16LE
+    -   A420_16BE, A420_16LE
 
--   Stateful decoding support fixes for NXP/Amphion driver
+-   8-bit A422 / A444 (YUV with alpha channel) variant:
 
--   Support for hardware crop in v4l2src
+    -   A422
+    -   A444
 
--   Conformance test improvement for stateful decoders
+-   Planar 16-bit 4:4:4 RGB formats:
 
--   Fixes for Raspberry Pi CODEC
+    -   GBR_16BE
+    -   GBR_16LE
 
-GStreamer OMX
+-   RBGA, intended to be used by hardware decoders where VUYA is only supported 4:4:4 decoding surface but the stream is encoded
+    with GBR color space, such as HEVC and VP9 GBR streams for example.
 
--   There were no changes in this module
+-   Two tiled Mediatek 10-bit formats:
 
-GStreamer Editing Services and NLE
+    -   MT2110T
+    -   MT2110R
 
--   Handle compositors that are bins around the actual compositor
-    implementation (like glvideomixers which wraps several elements)
+-   Tiled 10-bit NV12 format NV12_10LE40_4L4 (Verisilicon Hantro)
 
--   Add a mode to disable timeline editing API so the user can be in
-    full control of its layout (meaning that the user is responsible for
-    ensuring its validity/coherency)
+## Miscellaneous performance, latency and memory optimisations
 
--   Add a new fade-in transition type
+-   liborc 0.4.35 (latest: 0.4.38) adds support for AVX/AVX2 and contains improvements for the SSE backend.
 
--   Add support for non-1/1 PAR source videos
+-   liborc 0.4.37 adds support for NEON on macOS on Apple ARM64 CPUs.
 
--   Fix frame accuracy when working with very low framerate streams
+-   Most direct use of the GLib GSLice allocator has been removed, as there is little evidence that it actually still provides
+    much advantage over the standard system allocator on Linux or Windows in 2024. There is strong evidence however that it
+    causes memory fragmentation for standard GStreamer workloads such as RTSP/RTP/WebRTC streaming.
 
-GStreamer validate
+-   As always there have been plenty of performance, latency and memory optimisations all over the place.
 
--   Clean up and stabilize API so we can now generate rust bindings
+## Tracing framework and debugging improvements
 
--   Enhance the appsrc-push action type allowing to find tune the
-    buffers more in details
+-   The gst-stats tool can now be passed a custom regular expression
 
--   Add an action type to verify currently configured pad caps
+-   gst-debug-viewer from the devtools module has seen minor improvements and fixes
 
--   Add a way to run checks from any thread after executing a ‘wait’
-    action. This is useful when waiting on a signal and want to check
-    the value of a property right when it is emited for example.
+New tracers
 
-GStreamer Python Bindings
+-   None in this release.
 
--   Add a Gst.init_python() function to be called from plugins which
-    will initialise everything needed for the GStreamer Python bindings
-    but not call Gst.init() again since this will have been called
-    already.
+Debug logging system improvements
 
--   Add support for the GstURIHandlerInterface that allows elements to
-    advertise what URI protocols they support.
+-   Nothing major in this cycle.
 
-GStreamer C# Bindings
+Fake video decoder
 
--   Fix AppSrc and AppSink constructors
+-   The new fakevideodec element does not decode the input bitstream, it only reads video width, height and framerate from the
+    caps and then pushes out raw video frames of the expected size in RGB format.
 
--   The C# bindings have yet to be updated to include new 1.22 API,
-    which requires improvements in various places in the bindings /
-    binding generator stack. See issue #1718 in GitLab for more
-    information and to track progress.
+-   It draws a snake moving from left to right in the middle of the frame, which is reasonably light weight and still provides
+    an idea about how smooth the rendering is.
 
-GStreamer Rust Bindings and Rust Plugins
+## Tools
 
-The GStreamer Rust bindings are released separately with a different
-release cadence that’s tied to gtk-rs, but the latest release has
-already been updated for the new GStreamer 1.22 API. Check the bindings
-release notes for details of the changes since 0.18, which was released
-around GStreamer 1.20.
+-   gst-launch-1.0 gained a new --prog-name command line option to set the program name, which will be used by GTK and GStreamer
+    to set the class or app-id.
 
-gst-plugins-rs, the module containing GStreamer plugins written in Rust,
-has also seen lots of activity with many new elements and plugins. A
-list of all Rust plugins and elements provided with the 0.9 release can
-be found in the repository.
+-   gst-play-1.0 now defaults to using playbin3, but can still be made to use the old playbin by passing the --use-playbin2
+    command line argument.
 
--   33% of GStreamer commits are now in Rust (bindings + plugins), and
-    the Rust plugins module is also where most of the new plugins are
-    added these days.
+## GStreamer FFmpeg wrapper
 
--   The Rust plugins are now shipped as part of the Windows MSVC + macOS
-    binary packages. See below for the list of shipped plugins and the
-    status of Rust support in cerbero.
+-   New avvideocompare element to compare two incoming video buffers using a specified comparison method (e.g. SSIM or PSNR).
 
--   The Rust plugins are also part of the documentation on the GStreamer
-    website now.
+-   Prefer using FFmpeg Musepack decoder/demuxer over musepackdec as they work better with decodebin3 and playbin3 which likes
+    to have parsers and decoders separate.
 
--   Rust plugins can be used from any programming language. To the
-    outside they look just like a plugin written in C or C++.
+-   Added codec mappings for AV1, MxPEG, FFVHuff video
 
-New Rust plugins and elements
+-   Added raw video format support for P010, VUYA, Y410, P012, Y212 and Y412.
 
--   rtpav1pay / rtpav1depay: RTP (de)payloader for the AV1 video codec
--   gtk4paintablesink: a GTK4 video sink that provides a GdkPaintable
-    for rendering a video in any place inside a GTK UI. Supports
-    zero-copy rendering via OpenGL on Linux and macOS.
--   ndi: source, sink and device provider for NewTek NDI protocol
--   onvif: Various elements for parsing, RTP (de)payloading, overlaying
-    of ONVIF timed metadata.
--   livesync: Element for converting a live stream into a continuous
-    stream without gaps and timestamp jumps while preserving live
-    latency requirements.
--   raptorq: Encoder/decoder elements for the RaptorQ FEC mechanism that
-    can be used for RTP streams (RFC6330).
+-   Newer, non-deprecated APIs are used by the plugin when built with FFmpeg 6.0 or newer.
 
-WebRTC elements
+-   The FFmpeg meson subproject wrap has been updated to v6.1
 
--   webrtcsink: a WebRTC sink (batteries included WebRTC sender with
-    specific signalling)
--   whipsink: WebRTC HTTP ingest (WHIP) to MediaServer
--   whepsrc: WebRTC HTTP egress (WHEP) from MediaServer
--   rtpgccbwe: RTP bandwidth estimator based on the Google Congestion
-    Control algorithm (GCC), used by webrtcsink
+-   Note: see Known Issues section below for known issues with FFmpeg 6.0.0 and the latest FFmpeg 7.x release
 
-Amazon AWS services
+## GStreamer RTSP server
 
--   awss3src / awss3sink: A source and sink element to talk to the
-    Amazon S3 object storage system.
--   awss3hlssink: A sink element to store HLS streams on Amazon S3.
--   awstranscriber: an element wrapping the AWS Transcriber service.
--   awstranscribeparse: an element parsing the packets of the AWS
-    Transcriber service.
+-   New “ensure-keyunit-on-start” property: While the suspend modes NONE and PAUSED provided a low startup latency for
+    connecting clients, it did not ensure that streams started on fresh data. With this new property it is possible to maintain
+    the low startup latency of those suspend modes while also ensuring that a stream starts on a key unit. Furthermore, by
+    setting the new “ensure-keyunit-on-start-timeout” property it is also possible to accept a key unit of a certain age, but
+    discard it if too much time has passed and instead force a new key unit.
 
-Video Effects (videofx)
+-   rtspclientsink: apply “port-range” property for RTCP port selection as well
 
--   roundedcorners: Element to make the corners of a video rounded via
-    the alpha channel.
--   colordetect: A pass-through filter able to detect the dominant
-    color(s) on incoming frames, using color-thief.
--   videocompare: Compare similarity of video frames. The element can
-    use different hashing algorithms like Blockhash, DSSIM, and others.
+## GStreamer VA-API support
 
-New MP4 muxer + Fragmented MP4 muxer
+GstVA
 
--   isofmp4mux, cmafmux, dashmp4mux, onviffmp4mux: New fragmented
-    MP4/ISOBMFF/CMAF muxer for generating e.g. DASH/HLS media fragments.
--   isomp4mux, onvifmp4mux: New non-fragmented, normal MP4 muxer.
+-   vah264dec, vah265dec, vavp8dec, vavp9dec, vampeg2dec and vaav1dec were promoted to rank PRIMARY+1 on Linux
 
-Both plugins provides elements that replace the existing qtmux/mp4mux
-element from gst-plugins-good. While not feature-equivalent yet, the new
-codebase and using separate elements for the fragment and non-fragmented
-case allows for easier extensability in the future.
+-   Improved support for dmabuf use cases. All va elements now negotiate the new and improved dmabuf capabilities with explicit
+    modifiers. This supports both import and export of dmabufs.
 
-Cerbero Rust support
+-   Added vaav1enc element, available in recent Intel and AMD GPUs
 
--   Starting this release, cerbero has support for building and shipping
-    Rust code on Linux, Windows (MSVC) and macOS. The Windows (MSVC) and
-    macOS binaries also ship the GStreamer Rust plugins in this release.
-    Only dynamic plugins are built and shipped currently.
+-   Added support for the experimental VA-Win32 backend. It needs at least libva 1.18
 
--   Preliminary support for Android, iOS and Windows (MinGW) exists but
-    more work is needed. Check the tracker issue for more details about
-    future work.
+-   Improved handling of multi-GPU systems. Still, sharing buffers among them is not advised.
 
--   The following plugins are included currently: audiofx, aws, cdg,
-    claxon, closedcaption, dav1d, fallbackswitch, ffv1, fmp4, gif,
-    hlssink3, hsv, json, livesync, lewton, mp4, ndi, onvif, rav1e,
-    regex, reqwest, raptorq, png, rtp, textahead, textwrap, threadshare,
-    togglerecord, tracers, uriplaylistbin, videofx, webrtc, webrtchttp.
+-   Bumped minimum libva version to 1.12
 
-Build and Dependencies
+-   Enhanced support for RadeonSI Mesa driver for 10bit decoding
 
--   meson 0.62 or newer is required
+-   Register elements only for allowed drivers (Intel and Mesa, for the moment)
 
--   GLib >= 2.62 is now required (but GLib >= 2.64 is strongly
-    recommended)
+GStreamer-VAAPI
 
--   libnice >= 0.1.21 is now required and contains important fixes for
-    GStreamer’s WebRTC stack.
+-   The new GstVA elements (see above) should be preferred when possible.
 
--   liborc >= 0.4.33 is recommended for 64-bit ARM support and 32-bit
-    ARM improvements
+-   Ranks of decoders were demoted to NONE so they won’t be used automatically by playbin and similar elements anymore.
 
--   onnx: OnnxRT >= 1.13.1 is now required
+-   Clean-ups and minimal fixes.
 
--   openaptx: can now be built against libfreeaptx
+-   gstreamer-vaapi should be considered deprecated and may be discontinued as soon as the va plugin is fully feature
+    equivalent. Users who rely on gstreamer-vaapi are encouraged to migrate and test the va elements at the earliest
+    opportunity.
 
--   opencv: allow building against any 4.x version
+## GStreamer Video4Linux2 support
 
--   shout: libshout >= 2.4.3 is now required
+-   New uvcsink element, based on v4l2sink allow streaming your pipeline as a UVC camera using Linux UVC Gadget driver.
 
--   gstreamer-vaapi’s Meson build options have been switched from a
-    custom combo type (yes/no/auto) to the built-in Meson feature type
-    (enabled/disabled/auto)
+-   v4l2src now supports 10/12/14/16-bit bayer formats.
 
--   The GStreamer Rust plugins module gst-plugins-rs is now considered
-    an essential part of the GStreamer plugin offering and packagers and
-    distributors are strongly encouraged to package and ship those
-    plugins alongside the existing plugin modules.
+-   Stateful decoders now pass too large encoded frames over multiple buffers.
 
--   we now make use of Meson’s install tags feature which allows
-    selective installation of installl components and might be useful
-    for packagers.
+-   AV1 Stateless video decoder.
 
-Monorepo build (gst-build)
+-   Stateless decoders now tested using Virtual driver (visl), making it possible to run the tests in the cloud based CI
 
--   new “orc-source” build option to allow build against a
-    system-installed liborc instead of forcing the use of orc as a
-    subproject.
+## GStreamer OMX
 
--   GStreamer command line tools can now be linked to the gstreamer-full
-    library if it’s built
+-   The gst-omx module has been removed. The OpenMAX standard is long dead and even the Raspberry Pi OS no longer supports it.
+    There has not been any development since 1.22 was released. Users of these elements should switch to the Video4Linux-based
+    video encoders and decoders which have been the standard on embedded Linux for quite some time now.
 
-Cerbero
+-   Hardware vendors which still use OpenMAX are known to have non-standard forks and it is recommended that they maintain it
+    while planning their move to the Video4Linux API.
 
-Cerbero is a meta build system used to build GStreamer plus dependencies
-on platforms where dependencies are not readily available, such as
-Windows, Android, iOS, and macOS.
+## GStreamer Editing Services and NLE
 
-General improvements
+-   Implement a gesvideoscale effect which gives user the ability to chooses where a clip has to be scaled in the chain of
+    effects. By default scaling is done in the compositor.
 
--   Rust support was added for all support configurations, controlled by
-    the rust variant; see above for more details
--   All pkgconfig files are now reliably relocatable without requiring
-    pkg-config --define-prefix. This also fixes statically linking with
-    GStreamer plugins using the corresponding pkgconfig files.
--   New documentation on how to build a custom GStreamer repository
-    using Cerbero, please see the README
--   HTTPS certificate checking is enabled for downloads on all platforms
-    now
--   Fetching now automatically retries on error for robustness against
-    transient errors
--   Support for building the new Qt6 plugin was added
--   pkgconfig files for various recipes were fixed
--   Several recipes were updated to newer versions
--   New plugins: adaptivedemux2 aes codectimestamper dav1d
--   New libraries: cuda webrtcnice
-
-macOS / iOS
-
--   Added support for running Cerbero on ARM64 macOS
--   GStreamer.framework and all libraries in it are now relocatable,
-    which means they use LC_RPATH entries to find dependencies instead
-    of using an absolute path. If you link to GStreamer using the
-    pkgconfig files, no action is necessary. However, if you use the
-    framework directly or link to the libraries inside the framework by
-    hand, then you need to pass -Wl,-rpath,<path_to_libdir> to the
-    linker.
--   Apple bitcode support was dropped, since Apple has deprecated it
--   macOS installer now correctly advertises support for both x86_64 and
-    arm64
--   macOS framework now ships the gst-rtsp-server-1.0 library
--   Various fixes were made to make static linking to gstreamer
-    libraries and plugins work correctly on macOS
--   When statically linking to the applemedia plugin using Xcode 13, you
-    will need to pass -fno-objc-msgsend-selector-stubs which works
-    around a backwards-incompatible change in Xcode 14. This is not
-    required for the rest of GStreamer at present, but will be in the
-    future.
--   macOS installer now shows the GStreamer logo correctly
+-   Add support for gessrc as sub-timeline element so third party can implement their own formatters and use their timelines as
+    sub-timelines. Before that, only timelines serialized as files on the filesystem could be loaded as sub-timelines (using
+    gesdemux).
 
-Windows
+-   Implement a new GESDiscovererManager singleton object making management of the discoverers used to discoverer media files
+    cleaner and allowing to expose the following APIs:
 
--   MSVC is now required by default on Windows, and the Visual Studio
-    variant is enabled by default
-    -   To build with MinGW, use the mingw variant
--   Visual Studio props files were updated for newer Visual Studio
-    versions
--   Visual Studio 2015 support was dropped
--   MSYS2 is now supported as the base instead of MSYS. Please see the
-    README for more details. Some advantages include:
-    -   Faster build times, since parallel make works
-    -   Faster bootstrap, since some tools are provided by MSYS2
-    -   Other speed-ups due to using MSYS2 tools instead of MSYS
--   Faster download by using powershell instead of hand-rolled Python
-    code
--   Many recipes were ported from Autotools to Meson, speeding up the
-    build
--   Universal Windows Platform is no longer supported, and binaries are
-    no longer shipped for it
--   New documentation on how to force a specific Visual Studio
-    installation in Cerbero, please see the README
--   New plugins: qsv wavpack directshow amfcodec wic win32ipc
--   New libraries: d3d11
+    -   load-serialize-info signal so GstDiscovererInfo can be serialized by users the way they like and load them without
+        requiring discovering the file when reloading a project.
+    -   source-setup signal so user can tweak source elements during discovery
 
-Windows MSI installer
+-   Expose GESFrameCompositionMeta in public API so user can implement their own effects targetting GES which take into account
+    that meta.
 
--   Universal Windows Platform prebuilt binaries are no longer available
+-   Expose audioconvert:mix-matrix property in audio sources
 
-Linux
+-   Port GESPipeline rendering to use encodebin2. This allows rendering timelines directly with a muxing sink (like hlssinkX
+    etc..) and leverage all new features of that new element.
 
--   Various fixes for RHEL/CentOS 7 support
--   Added support for running on Linux ARM64
+ges-launch
 
-Android
+-   Fix setting keyframes
 
--   Android support now requires Android API version 21 (Lollipop)
--   Support for Android Gradle plugin 7.2
+-   Add an ignore-eos option
 
-Platform-specific changes and improvements
+-   Allow overriding container profile so that the user can build encoding profiles following the media format of a specific
+    media file, for example, but ensuring it is muxed using a specific format
 
-Android
+-   Ensure sink elements are inside a GstBin and never in a GstPipeline
 
--   Android SDK 21 is required now as minimum SDK version
+-   Move +effect stack effects from source to last effect added, so it feels more natural to user as adding them at the
+    beginning of the chain while the syntax is +effect felt wrong
 
--   androidmedia: Add H.265 / HEVC video encoder mapping
+## GStreamer validate
 
--   Implement JNI_OnLoad() to register static plugins etc. automatically
-    in case GStreamer is loaded from Java using System.loadLibrary(),
-    which is also useful for the gst-full deployment scenario.
-
-Apple macOS and iOS
-
--   The GLib version shipped with the GStreamer binaries does not
-    initialize an NSApp and does not run a NSRunLoop on the main thread
-    anymore. This was a custom GLib patch and caused it to behave
-    different from the GLib shipped by Homebrew or anybody else.
-
-    The change was originally introduced because various macOS APIs
-    require a NSRunLoop to run on the main thread to function correctly
-    but as this change will never get merged into GLib and it was
-    reverted for 1.22. Applications that relied on this behaviour should
-    move to the new gst_macos_main() function, which also does not
-    require the usage of a GMainLoop.
-
-    See e.g. gst-play.c for an example for the usage of
-    gst_macos_main().
-
--   GStreamer.framework and all libraries in it are now relocatable,
-    which means they use LC_RPATH entries to find dependencies instead
-    of using an absolute path. If you link to GStreamer using the
-    pkgconfig files, no action is necessary. However, if you use the
-    framework directly or link to the libraries inside the framework by
-    hand, then you need to pass -Wl,-rpath,<path_to_libdir> to the
-    linker.
-
--   avfvideosrc: Allow specifying crop coordinates during screen capture
-
--   vtenc, vtdec: H.265 / HEVC video encoding + decoding support
-
--   osxaudiosrc: Support a device as both input and output
-
-    -   osxaudiodeviceprovider now probes devices more than once to
-        determine if the device can function as both an input AND and
-        output device. Previously, if the device provider detected that
-        a device had any output capabilities, it was treated solely as
-        an Audio/Sink. This caused issues for devices that have both
-        input and output capabilities (for example, USB interfaces for
-        professional audio have both input and output channels). Such
-        devicesare now listed as both an Audio/Sink as well as an
-        Audio/Source.
-
--   osxaudio: support hidden devices on macOS
-
-    -   These are devices that will not be shown in the macOS UIs and
-        that cannot be retrieved without having the specific UID of the
-        hidden device. There are cases when you might want to have a
-        hidden device, for example when having a virtual speaker that
-        forwards the data to a virtual hidden input device from which
-        you can then grab the audio. The blackhole project supports
-        these hidden devices and this change provides a way that if the
-        device id is a hidden device it will use it instead of checkinf
-        the hardware list of devices to understand if the device is
-        valid.
+-   In action types, add a way to avoid checking property value after setting it, in case elements do it async for example.
 
-Windows
+-   Add a vmethod to free GstValidateActionParameters to be more binding friendly.
 
--   win32ipcvideosink, win32ipcvideosrc: new shared memory videosrc/sink
-    elements
+-   Allow scenarios to define the pipeline state target in the metadata instead of assuming PLAYING state.
 
--   wasapi2: Add support for process loopback capture for a specified
-    PID (requires Windows 11/Windows Server 2022)
+-   Add support to run sub-pipelines/scenarios
 
--   The Windows universal UWP build is currently non-functional and will
-    need updating after the recent GLib upgrade. It is unclear if anyone
-    is using these binaries, so if you are please make yourself known.
+    -   Added support to forward buffers from appsink to appsrc
 
--   wicjpegdec, wicpngdec: Windows Imaging Component (WIC) based JPEG
-    and PNG decoder elements.
+-   Add a way to set pipeline base-time, start-time and force using the system clock.
 
--   mfaacdec, mfmp3dec: Windows MediaFoundation AAC and MP3 decoders
+-   Add a ‘fill-mode’ to the appsrc-push action type so we can create some type of streams easily using an appsrc, giving
+    control when writing scenarios without requiring files with the content.
 
--   The uninstalled development environment supports PowerShell 7 now
+-   Add a “select-streams” action type to test “stream aware” elements.
 
-Linux
+-   Add a way to wait for a property to reach a specified value before executing an action. For example it is possible to wait
+    for a pad to get some specific caps set before executing an action.
 
--   Improved design for DMA buffer sharing and modifier handling for
-    hardware-accelerated video decoders/encoders/filters and
-    capture/rendering on Linux and Linux-like system.
+-   validate: Add support to replace variables in deeply nested structures in particular for more complex action types where
+    some of the properties are inside structures.
 
--   kmssink
+-   Fixed compatibility with Python 3.12.
 
-    -   new “fd” property which allows an application to provide their
-        own opened DRM device fd handle to kmssink. That way an
-        application can lease multiple fd’s from a DRM master to display
-        on different CRTC outputs at the same time with multiple kmssink
-        instances, for example.
-    -   new “skip-vsync” property to achieve full framerate with legacy
-        emulation in drivers.
-    -   HDR10 infoframe support
+## GStreamer Python Bindings
 
--   va plugin and gstreamer-vaapi improvements (see above)
+gst-python is an extension of the regular GStreamer Python bindings based on gobject-introspection information and PyGObject,
+and provides “syntactic sugar” in form of overrides for various GStreamer APIs that makes them easier to use in Python and more
+pythonic; as well as support for APIs that aren’t available through the regular gobject-introspection based bindings, such as
+e.g. GStreamer’s fundamental GLib types such as Gst.Fraction, Gst.IntRange etc.
 
--   waylandsink: Add “rotate-method” property and “render-rectangle”
-    property
+-   Added a GstTagList override that makes a tag list act like a dict
 
--   new gtkwaylandsink element based on gtksink, but similar to
-    waylandsink and uses Wayland APIs directly instead of rendering with
-    Gtk/Cairo primitives. This approach is only compatible with Gtk3,
-    and like gtksink this element only supports Gtk3.
+-   Fix build and usage in Windows
 
-Documentation improvements
+-   Various fixes for Python >= 3.12
 
--   The GStreamer Rust plugins are now included and documented in the
-    plugin documentation.
+-   Rework libpython loading to be relocatable
 
-Possibly Breaking Changes
+-   Fix libpython dlopen on macOS
 
--   the Opus audio RTP payloader and depayloader no longer accept the
-    lower case encoding-format=multiopus but instead produce and accept
-    only the upper case variant encoding-format=MULTIOPUS, since those
-    should always be upper case in GStreamer (caps fields are always
-    case sensitive). This should hopefully only affect applications
-    where RTP caps are set manually and multi-channel audio (>= 3
-    channels) is used.
+## GStreamer C# Bindings
 
--   wpesrc: the URI handler protocols changed from wpe:// and web:// to
-    web+http://, web+https://, and web+file:// which means URIs are RFC
-    3986 compliant and the source can simply strip the prefix from the
-    protocol.
+-   The GStreamer C# bindings have been updated to a more recent version of GtkSharp and the bindings have been regenerated with
+    that version.
 
--   The Windows screen capture element dxgiscreencapsrc has been
-    removed, please use d3d11screencapturesrc instead.
+-   GStreamer API added in recent GStreamer releases is now available
 
--   On Android the minimum supported Android API version is now version
-    21 and has been increased from 16.
+-   GstRtspServer bindings have been added, plus an RTSP server example
 
--   On macOS, the GLib version shipped with the GStreamer binaries will
-    no longer initialize an NSApp or run an NSRunLoop on the main
-    thread. See macOS/iOS section above for details.
+## GStreamer Rust Bindings and Rust Plugins
 
--   decklink: The decklink plugin is now using the 12.2.2 version of the
-    SDK and will not work with drivers older than version 12.
+The GStreamer Rust bindings and plugins are released separately with a different release cadence that’s tied to the twice-a-year
+GNOME release cycle.
 
--   On iOS Apple Bitcode support was removed from the binaries. This
-    feature is deprecated since XCode 14 and not used on the App Store
-    anymore.
+The latest release of the bindings (0.22) has already been updated for the new GStreamer 1.24 APIs, and works with any GStreamer
+version starting at 1.14.
 
--   The MP4/Matroska/WebM muxers now require the “stream-format” to be
-    provided as part of the AV1 caps as only the original “obu-stream”
-    format is supported in these containers and not the “annexb” format.
+gst-plugins-rs, the module containing GStreamer plugins written in Rust, has also seen lots of activity with many new elements
+and plugins. The GStreamer 1.24 binaries track the 0.12 release series of gst-plugins-rs, and fixes from newer versions will be
+backported as needed to the 0.12 brach for future 1.24.x bugfix releases.
 
-Known Issues
+Rust plugins can be used from any programming language. To applications they look just like a plugin written in C or C++.
 
--   The Windows UWP build in Cerbero needs fixing after the recent GLib
-    upgrade (see above)
+### WebRTC
 
--   The C# bindings have not been updated to include new 1.22 API yet
-    (see above)
+-   New element webrtcsrc that can act as a recvonly WebRTC client. Just like the opposite direction, webrtcsink, this can
+    support various different WebRTC signalling protocols. Some are included with the plugin and provide their own element
+    factory for easier usage but it is also possible for applications to provide new signalling protocol implementations.
 
-Statistics
+-   webrtcsink now exposes the signaller as property and allows implementing a custom signaller by connecting signal handlers to
+    the default signaller.
 
--   4072 commits
+-   A new signaller and webrtcsink implementation for Janus’ VideoRoom implementation. The corresponding webrtcsrc signaller
+    implementation is currently in a merge request in GitLab.
 
--   2224 Merge Requests
+-   New whepsrc element that can receive WHEP WebRTC streams. This is currently not based on webrtcsrc but in the future a new
+    element around webrtcsrc will be added.
 
--   716 Issues
+-   New whipserversrc element around webrtcsrc for ingesting WHIP WebRTC streams in GStreamer.
 
--   200+ Contributors
+-   New whipclientsink element around webrtcsink for publishing WHIP WebRTC streams from GStreamer. This deprecates the old
+    whipsink element.
 
--   ~33% of all commits and Merge Requests were in Rust modules
+-   A new signaller and webrtcsink implementation for LiveKit. The corresponding webrtcsrc signaller implementation was merged
+    into the git repository recently.
 
--   4747 files changed
+-   A new signaller and webrtcsink implementation for AWS Kinesis Video Streams
 
--   469633 lines added
+-   webrtcsink has a new payloader-setup signal to allow the application more fine grained control over the RTP payloader
+    configuration, similar to the already existing encoder-setup signal for encoders.
 
--   209842 lines deleted
+-   webrtcsrc gained support for a custom navigation event protocol over the data channel, which is compatible with the
+    navigation event protocol supported by webrtcsink.
 
--   259791 lines added (net)
+-   webrtcsink supports encoded streams as input. Using encoded streams will disable webrtcsinks congestion control changing any
+    encoded stream parameters.
 
-Contributors
+-   webrtcsink and webrtcsrc have a new signal ‘request-encoded-filter’ to allow transformations of the encoded stream. This can
+    be used, for example, for the same use-cases as the WebRTC Insertable Streams API.
 
-Ádám Balázs, Adam Doupe, Adrian Fiergolski, Adrian Perez de Castro, Alba
-Mendez, Aleix Conchillo Flaqué, Aleksandr Slobodeniuk, Alicia Boya
-García, Alireza Miryazdi, Andoni Morales Alastruey, Andrew Pritchard,
-Arun Raghavan, A. Wilcox, Bastian Krause, Bastien Nocera, Benjamin
-Gaignard, Bill Hofmann, Bo Elmgreen, Boyuan Zhang, Brad Hards, Branko
-Subasic, Bruce Liang, Bunio FH, byran77, Camilo Celis Guzman, Carlos
-Falgueras García, Carlos Rafael Giani, Célestin Marot, Christian Wick,
-Christopher Obbard, Christoph Reiter, Chris Wiggins, Chun-wei Fan, Colin
-Kinloch, Corentin Damman, Corentin Noël, Damian Hobson-Garcia, Daniel
-Almeida, Daniel Morin, Daniel Stone, Daniels Umanovskis, Danny Smith,
-David Svensson Fors, Devin Anderson, Diogo Goncalves, Dmitry Osipenko,
-Dongil Park, Doug Nazar, Edward Hervey, ekwange, Eli Schwartz, Elliot
-Chen, Enrique Ocaña González, Eric Knapp, Erwann Gouesbet, Evgeny
-Pavlov, Fabian Orccon, Fabrice Fontaine, Fan F He, F. Duncanh, Filip
-Hanes, Florian Zwoch, François Laignel, Fuga Kato, George Kiagiadakis,
-Guillaume Desmottes, Gu Yanjie, Haihao Xiang, Haihua Hu, Havard Graff,
-Heiko Becker, He Junyan, Henry Hoegelow, Hiero32, Hoonhee Lee, Hosang
-Lee, Hou Qi, Hugo Svirak, Ignacio Casal Quinteiro, Ignazio Pillai, Igor
-V. Kovalenko, Jacek Skiba, Jakub Adam, James Cowgill, James Hilliard,
-Jan Alexander Steffens (heftig), Jan Lorenz, Jan Schmidt, Jianhui Dai,
-jinsl00000, Johan Sternerup, Jonas Bonn, Jonas Danielsson, Jordan
-Petridis, Joseph Donofry, Jose Quaresma, Julian Bouzas, Junsoo Park,
-Justin Chadwell, Khem Raj, Krystian Wojtas, László Károlyi, Linus
-Svensson, Loïc Le Page, Ludvig Rappe, Marc Leeman, Marek Olejnik, Marek
-Vasut, Marijn Suijten, Mark Nauwelaerts, Martin Dørum, Martin Reboredo,
-Mart Raudsepp, Mathieu Duponchelle, Matt Crane, Matthew Waters, Matthias
-Clasen, Matthias Fuchs, Mengkejiergeli Ba, MGlolenstine, Michael Gruner,
-Michiel Konstapel, Mikhail Fludkov, Ming Qian, Mingyang Ma, Myles
-Inglis, Nicolas Dufresne, Nirbheek Chauhan, Olivier Crête, Pablo Marcos
-Oltra, Patricia Muscalu, Patrick Griffis, Paweł Stawicki, Peter
-Stensson, Philippe Normand, Philipp Zabel, Pierre Bourré, Piotr
-Brzeziński, Rabindra Harlalka, Rafael Caricio, Rafael Sobral, Rafał
-Dzięgiel, Raul Tambre, Robert Mader, Robert Rosengren, Rodrigo
-Bernardes, Rouven Czerwinski, Ruben Gonzalez, Sam Van Den Berge,
-Sanchayan Maity, Sangchul Lee, Sebastian Dröge, Sebastian Fricke,
-Sebastian Groß, Sebastian Mueller, Sebastian Wick, Sergei Kovalev,
-Seungha Yang, Seungmin Kim, sezanzeb, Sherrill Lin, Shingo Kitagawa,
-Stéphane Cerveau, Talha Khan, Taruntej Kanakamalla, Thibault Saunier,
-Tim Mooney, Tim-Philipp Müller, Tomasz Andrzejak, Tom Schuring, Tong Wu,
-toor, Tristan Matthews, Tulio Beloqui, U. Artie Eoff, Víctor Manuel
-Jáquez Leal, Vincent Cheah Beng Keat, Vivia Nikolaidou, Vivienne
-Watermeier, WANG Xuerui, Wojciech Kapsa, Wonchul Lee, Wu Tong, Xabier
-Rodriguez Calvar, Xavier Claessens, Yatin Mann, Yeongjin Jeong, Zebediah
-Figura, Zhao Zhili, Zhiyuaniu, مهدي شينون (Mehdi Chinoune),
-
-… and many others who have contributed bug reports, translations, sent
-suggestions or helped testing.
-
-Stable 1.22 branch
-
-After the 1.22.0 release there will be several 1.22.x bug-fix releases
-which will contain bug fixes which have been deemed suitable for a
-stable branch, but no new features or intrusive changes will be added to
-a bug-fix release usually. The 1.22.x bug-fix releases will be made from
-the git 1.22 branch, which will be a stable branch.
-
-1.22.0
-
-1.22.0 was originally released on 23 January 2023.
-
-1.22.1
-
-The first 1.22 bug-fix release (1.22.1) was released on 04 March 2023.
-
-This release only contains bugfixes and it should be safe to update from
-1.22.0.
-
-Highlighted bugfixes in 1.22.1
-
--   audio channel-mix: allow up to 64 channels (instead of up to 63
-    channels)
--   avfvideosrc: Don’t wait on main thread for permissions request
--   avvidenc: avoid generating inaccurate output timestamps, especially
-    with variable framerate streams
--   AV1 video codec caps signalling improvements in various elements
--   codectimestamper: Fix timestamping on sequence update
--   d3d11overlaycompositor: fix texture width and height
--   d3d11videosink: Fix rendering on external handle
--   dashdemux2: fix seek operation taking a log time to finish for some
-    streams
--   nvencoder: Fix B-frame encoding on Linux and min buffers in auto GPU
-    mode
--   playbin3: fixing buffering for live pipelines
--   playbin: fix potential deadlock when stopping stream with subtitles
-    visible
--   redenc: fix setting of extension ID for twcc
--   rtspsrc: improved compatibility with more broken RTSP servers
--   v4l2h264dec: Fix Raspberry Pi4 will not play video in application
--   vtdec: fix jittery playback of H.264 Level 4.1 movies in macOS
--   vtdec: Fix non-deterministic frame output after flushing seeks
--   vtenc: fix handling of interlaced ProRes on Apple M1 hardware
--   vtenc: don’t advertise ARGB/RGBA64 input caps on M1 Pro/Max with
-    macOS <13
--   wasapi2src: Fix loopback capture on Windows 10 Anniversary Update
--   tools: better handling of non-ASCII command line arguments on
-    Windows
--   gst-libav: fix build against newer ffmpeg versions
--   gst-python: Use arch-specific install dir for gi overrides
--   cerbero: Fix setuptools site.py breakage in Python 3.11
--   macOS packages: Fix broken binaries on macos < 11.0
--   various bug fixes, memory leak fixes, and other stability and
-    reliability improvements
+-   gstwebrtc-api: JavaScript API for interacting with the default signalling protocol used by webrtcsink / webrtcsrc.
 
-gstreamer
+… and various other smaller improvements!
 
--   buffer: fix copy meta reference debug log formatting
--   bin: Don’t unlock unlocked mutex in gst_bin_remove_func()
--   pad: Don’t leak user_data in gst_pad_start_task()
--   aggregator: Always lock aggpad around update_time_level
--   inputselector: Avoid potential deadlock when shutting down,
-    e.g. playbin with subtitles
--   multiqueue: Handle use-interleave latency live pipelines, fixing
-    buffering for live pipelines in playbin3
--   GstBaseSrc: fix transfer annotation for fixate() virtual method
--   GstBaseSrc, GstPushSrc: add nullable annotations to virtual methods
--   tools: Make sure UTF-8 encoded command line arguments on Windows
+### RTSP
 
-gst-plugins-base
+-   New rtspsrc2 element. Only a subset of RTSP features are implemented so far:
+    -   RTSP 1.0 support
+    -   TCP, UDP, UDP-Multicast lower transports
+    -   RTCP SR, RTCP RR, RTCP-based A/V sync
+        -   Tested for correctness in multicast cases too
+    -   Lower transport selection and order (NEW!)
+        -   The existing rtspsrc has a hard-coded order list for lower transports
+    -   Many advanced features are not implemented yet, such as non-live support. See the README for the current status.
 
--   alsasink: Fix stall when going from PLAYING to NULL (stucked at
-    PAUSED) with uac1 gadget
--   appsrc: Don’t chain up BaseSrc::negotiate()
--   audio: channel-mix: Fix channel count limit to be able to equal 64
--   gldisplay: Mark gst_gl_display_create_context() other_context
-    parameter as nullable
--   gldisplay: Remove unused code
--   gstglwindow_x11.c: Fix colormap leak
--   gl/cocoa: Return a strong ref to the parent GstGLContext
--   rtspconnection: Annotate RTSP message and RTSP events parameters
-    correctly
--   sdp, typefind: Fix some annotations
--   sdp: gstmikey: gst_mikey_message_to_caps: extract ROC from first
-    crypto session
--   subparse: Properly forward segment seqnum
--   uridecodebin: Set source element to READY before querying it
--   uritranscodebin: Fix unref of NULL
--   gst-play-1.0: Don’t force accurate seeking
+### GTK4
 
-gst-plugins-good
+-   Support for rendering GL textures on X11/EGL, X11/GLX, Wayland, macOS, and WGL/EGL on Windows.
 
--   adaptivedemux2: Fix buffering threshold initialization
--   dashdemux2: the seek operation takes a log time to finish for some
-    streams
--   glvideomixer: Keep a reference to the underlying pad
--   qtdemux: Don’t emit GstSegment correcting start time when in MSE
-    mode
--   qtdemux: Handle moov atom length=0 case by reading until the end
--   qtdemux, qtmux: Drop av1C version 0 parsing and implement version 1
-    parsing/writing
--   qtmux: Fix assertion on caps update
--   redenc: fix setting of extension ID for twcc
--   rtspsrc: Use the correct vfunc for the push-backchannel-sample
-    action signal
--   rtpssrcdemux: set different stream-id on each src pad
--   udpsrc: GstSocketTimestampMessage only for SCM_TIMESTAMPNS
--   v4l2h264dec: Fix Raspberry Pi4 will not play video in application
+-   Create a window for testing purposes when running in gst-launch-1.0 or if GST_GTK4_WINDOW=1 is set.
 
-gst-plugins-bad
+-   New background-color property for setting the color of the background of the frame and the borders, if any. This also allows
+    setting a fully transparent background.
 
--   aom: Include stream-format and alignment in the AV1 caps
--   av1parser, h265parser: Fix some code defects
--   av1parser: Don’t consider unknown metadata OBUs a bitstream error
--   avfvideosrc: Don’t wait on main thread for permissions request
--   ccconverter: don’t debug a potentially freed filter caps
--   codectimestamper: Fix timestamping on sequence update
--   codecparsers: {h264, h265}bitwriter: Remove redundant condition
-    checks
--   codecs: decoders: fail early if no input caps have been provided for
-    all new decoder base classes
--   closedcaption: Don’t leak caps event
--   curlhttpsrc: Add curl anyauth option
--   d3d11overlaycompositor: fix texture width and height
--   d3d11videosink: Fix rendering on external handle
--   h265parse: Always set profile on src caps
--   msdkav1enc: fix the category for msdkav1enc debug
--   nvcodec: improve error reporting on plugin init
--   nvencoder: Fix b-frame encoding on Linux
--   nvencoder: Fix min buffers parameter of allocation query in auto GPU
-    mode
--   nvvp9dec: Fix return value
--   qsvav1enc, amfav1enc: Set stream-format on caps
--   vtdec: Jittery playback of H.264 Level 4.1 movies in macOS (both
-    x86_64 and arm64)
--   vtdec: Fix DPB size calculations not taking values from SPS into
-    account
--   vtdec: Fix not waiting for async frames when flushing
--   vtenc: Disable ARGB/RGBA64 caps on M1 Pro/Max with macOS <13
--   vtenc: Fix checking for certain CPU variants when running in VMs
--   vtenc: Disable HW acceleration for interlaced ProRes
--   va: Avoid the array index overflow when filling 8x8 scaling list.
--   va: Fix some code defects
--   vah265enc: Use helper to update properties.
--   vulkan: memory: Flush non coherent memory after write.
--   wasapi2src: Fix loopback capture on Windows 10 Anniversary Update
--   webrtcbin: small stats improvements
--   win32ipcutils: Add missing include
--   wpe: Logging fixes for the WebExtension
+-   New scale-filter property for defining how to scale the frames.
 
-gst-plugins-ugly
+-   Add Python example application to the repository.
 
--   mpegpsdemux: Ignore DTS if PTS < DTS
+-   Various bugfixes, including support for the new GTK 4.14 GL renderer. The plugin needs to be built with at least the
+    gtk_v4_10 feature to work with the new GTK 4.14 GL renderer, and will work best if built with the gtk_v4_14 feature.
 
-gst-libav
+### Closed Caption
 
--   avauddec, avviddec: Free packet side data after usage
--   avviddec: change
-    AV_CODEC_CAP_AUTO_THREADS->AV_CODEC_CAP_OTHER_THREADS to fix build
-    against newer ffmpeg versions
--   Memory leak in ’ av_packet_add_side_data’ in
-    /lib/x86_64-linux-gnu/libavcodec.so reading the file
-    clock_odd_size_RLE_g1597902.avi
--   avvidenc: Don’t take ffmpeg timestamps verbatim but only use them to
-    calculate DTS
+-   Add cea608tocea708 element for upconverting CEA-608 captions to their CEA-708 representation.
 
-gst-rtsp-server
+-   Add support for translations within transcriberbin.
 
--   No changes
+-   awstranscriber supports translating the transcribed text into different languages, including multiple languages at the same
+    time.
 
-gstreamer-vaapi
+-   awstranscriber is using the new HTTP/2-based API now instead of the WebSocket-based one.
 
--   vaapi: Skip plugin pc file for shared plugins
+### Other new elements
 
-gstreamer-sharp
+-   New awss3putobjectsink that works similar to awss3sink but with a different upload strategy.
 
--   No changes
+-   New hlscmafsink element for writing HLS streams with CMAF/ISOBMFF fragments.
 
-gst-omx
+-   New inter plugin with new intersink and intersrc elements that allow to 1:N connect different pipelines in the same process.
+    This is implemented around the appsrc / appsink-based StreamProducer API that is provided as part of the GStreamer Rust
+    bindings, and is also used inside webrtcsrc and webrtcsink.
 
--   No changes
+-   New livesync element that allows maintaining a contiguous live stream without gaps from a potentially unstable source.
 
-gst-python
+-   New isomp4mux non-fragmented MP4 muxer element.
 
--   gst-python: Use arch-specific install dir for gi overrides
+### Other improvements
 
-gst-editing-services
+-   audiornnoise
+    -   Attach audio level meta to output buffers.
+    -   Add voice detection threshold property
+-   fmp4mux
+    -   Add support for CMAF-style chunking, e.g. low-latency / LL HLS and DASH
+    -   Add support for muxing Opus, VP8, VP9 and AV1 streams
+    -   Add ‘offset-to-zero’ property and make media/track timescales configurable
+-   hlssink3
+    -   Allow adding EXT-X-PROGRAM-DATE-TIME tag to the manifest.
+    -   Allow generating I-frame-only playlist
+-   ndi
+    -   Closed Caption support in ndisrc / ndisink
+    -   Zero-copy output support in ndisrc for raw video and audio
+-   spotifyaudiosrc: Support configurable bitrate
 
--   No changes
+For a full list of changes in the Rust plugins see the gst-plugins-rs ChangeLog between versions 0.9 (shipped with GStreamer
+1.22) and 0.12 (shipped with GStreamer 1.24).
 
-gst-validate + gst-integration-testsuites
+## Cerbero Rust support
 
--   validate:scenario: sink refs when building
--   tests: Fix known issue definition location for unit tests and how we
-    handle them in validate launcher
--   tests: mark elements_srtp.test_play test as flaky
--   Fix gstreamer-validate-1.0 dependency name
--   validate-scenario: fix g-i warning in annotation
--   validate: Fix gst_validate_execute_action annotation
+-   As of GStreamer 1.24, the GStreamer Rust plugins are shipped as part of our binary packages on all major platforms. This
+    includes Android and iOS now in addition to macOS and Windows/MSVC.
 
-gst-examples
+## Build and Dependencies
 
--   webrtc examples: Use webrtc.gstreamer.net
--   webrtc_sendrecv.py: Various fixes
+-   Meson >= 1.1 is now required for all modules
 
-Development build environment
+-   The GLib requirement has been bumped to >= 2.64
 
--   gst-env: Handle installing python modules to dist-packages
--   meson: Allow sysdeps to be forced as fallback subprojects
--   meson: Switch dav1d wrap to a tarball and update to dav1d 1.1.0
-
-Cerbero build tool and packaging changes in 1.22.1
-
--   macos: Fix broken binaries on macos < 11.0
--   orc: Update pthread_jit_write_protect fix for macOS/iOS
--   dav1d: Update to 1.1.0
--   libsrtp: update to v2.5.0
--   rustup: Update to 1.25.2
--   rust: Update to 1.67; cargo-c to 0.9.16
--   cerbero: Don’t error out if bindir already exists
--   Fix setuptools site.py breakage in Python 3.11, bump
-    gobject-introspection, bump windows image
--   cerbero: Retry if cargo update fails on macOS
--   gst-plugins-rs: Build glib/gio bindings with 2.74 API support
-
-Contributors to 1.22.1
-
-Alessandro Bono, Arun Raghavan, Bart Van Severen, Carlos Falgueras
-García, Célestin Marot, David Svensson Fors, Edward Hervey, Enrique
-Ocaña González, Frank Dana, Guillaume Desmottes, He Junyan, James
-Hilliard, Jan Alexander Steffens (heftig), Jan Schmidt, Jordan Petridis,
-Mathieu Duponchelle, Matthew Waters, medithe, Mengkejiergeli Ba, Nicolas
-Beland, Nirbheek Chauhan, Patricia Muscalu, Pawel Stawicki, Philippe
-Normand, Piotr Brzeziński, Rajneesh Soni, Robert Rosengren, Sanchayan
-Maity, Sebastian Dröge, Seungha Yang, Simon Himmelbauer, Thibault
-Saunier, Tim-Philipp Müller, Tristan van Berkom, U. Artie Eoff, Víctor
-Manuel Jáquez Leal, Vivia Nikolaidou, Xuchen Yang, Yinhang Liu,
-
-… and many others who have contributed bug reports, translations, sent
-suggestions or helped testing. Thank you all!
-
-List of merge requests and issues fixed in 1.22.1
-
--   List of Merge Requests applied in 1.22.1
--   List of Issues fixed in 1.22.1
-
-1.22.2
-
-The second 1.22 bug-fix release (1.22.2) was released on 11 April 2023.
-
-This release only contains bugfixes and it should be safe to update from
-1.22.x.
-
-Highlighted bugfixes in 1.22.2
-
--   avdec_h264: fix decoder deadlocks with FFmpeg 6.0
--   rtspsrc: fix regression with URI protocols in OPTIONS requests for
-    RTSP over TLS
--   rtspsrc: improved control url handling compatibility for broken
-    servers
--   decklink: fix 10 bit RGB (r210) format auto detection for capture
-    and fix playout if video caps are configured before audio caps
--   d3d11videosink: Fix tearing in case of fullscreen mode
--   playbin: fix deadlock when stopping stream with subtitles visible
-    (even more)
--   typefinding: fix regression not detecting application/dash+xml in
-    some corner cases
--   osxvideosink: fix broken aspect ratio and frame drawing region
--   decodebin3, parsebin: Improve elementary stream handling when
-    decoders are not present and fix hang when removing a failing stream
--   urisourcebin: Propagate sticky events from parsebin, so that the
-    STREAM_START event with the GstStream info is always available when
-    pads get exposed
--   v4l2: Add support for YVU420M format; mark JPEG content as parsed
--   h264decoder, h265decoder: DPB bumping process and latency reporting
-    fixes
--   Opus: Fix reading of extended channel config in MPEG-TS and fix
-    missing sample rate when remuxing from RTP to Matroska
--   zxing: add support for building against zxing-c++ 2.0
--   cerbero: Fix packaging of Rust plugins on Android; fix modern Gentoo
-    distro detection
--   various bug fixes, memory leak fixes, and other stability and
-    reliability improvements
+-   liborc >= 0.4.38 is strongly recommended
 
-gstreamer
+-   libnice >= 0.1.22 is strongly recommended, as it is required for WebRTC ICE consent freshness (RFC 7675).
 
--   datetime: Return G_MAXFLOAT instead of G_MAXDOUBLE for no timezone
-    offset
--   inputselector: Wake up streaming thread before PLAYING_TO_PAUSED
-    transition
--   tools: fix potential crash when passing command-line options on
-    Windows
+-   gst-libav was updated for FFmpeg API deprecations and removals
 
-gst-plugins-base
+-   libwebpmux is required for the animated WebP support
 
--   alsasink: Fix for being stuck in stop_streaming_threads state
--   decodebin3: fix hang when removing a failing stream
--   gl: wayland: cleanup on close
--   parsebin: Improve elementary stream handling
--   playbin: fix deadlock when stopping stream with subtitles visible
-    even more
--   sdp: Skip source-specific caps fields when creating an SDP media
-    from caps
--   urisourcebin: Propagate sticky events from parsebin
--   urisourcebin: Activate pad before transferring sticky events
--   typefinding: fix failure to recognize application/dash+xml in some
-    cases
+-   The wpe plugin gained support for the WPEWebKit 2.0 API version
 
-gst-plugins-good
+-   Bumped minimum libva version to 1.12 for the va plugin.
 
--   osxvideosink: fix broken aspect ratio and frame drawing region
--   qtdemux: Fix seek adjustment with SNAP_AFTER flag
--   rtpopusdepay, matroskamux: Fix invalid rate while muxing Opus in
-    Matroska
--   rtpmanager: twcc: Fix duplicate packet handling
--   rtsp: url: fix incorrect request URI scheme for TLS transport
-    methods (regression)
--   rtspsrc: Consider “451: Parameter Not Understood” when handling
-    broken control urls
--   rtspsrc: fix behavior change with URI protocols in OPTIONS requests
--   rtspsrc: Skip PTs with caps incompatible to the global caps
--   rtpjpegdepay: fix logic error when checking if an end of image (EOI)
-    tag is present
--   v4l2: Add support for YVU420M format
--   v4l2: mark JPEG as parsed
+-   zxing: added support for the zxing-c++ 2.0 API
 
-gst-plugins-bad
+-   The ptp-helper for Precision Time Protocol (PTP) support in GStreamer core has been rewritten in Rust, and the minimum
+    required Rust version for building this is 1.48, i.e. the version currently in Debian oldstable. On Windows, at least Rust
+    1.54 is needed. There is a new ptp-helper Meson feature option that can be used to make sure everything needed for PTP
+    support is available (if set to ptp-helper=enabled). cargo is not required for building.
 
--   cea708overlay: fix HCR interpretation
--   d3d11bufferpool: Fix invalid access in debug print loop
--   d3d11compositor: Fix composition error on release_pad()
--   d3d11converter: Fix conversion backend selection
--   d3d11videosink: Fix tearing in case of fullscreen mode -
-    d3d11bufferpool: Fix invalid access in debug print loop
--   d3d11window: fix memory leak
--   decklink: fix 10 bit RGB (r210) format auto detection
--   decklinkaudiosink: Fix playback when video caps is configured before
-    audio
--   decklinkvideosrc: RGB 4:4:4 doesn’t work after GStreamer upgrade
-    (regression)
--   decklinkvideosrc: unable to show HDMI stream that Blackmagic’s Media
-    Express is able to see
--   debugqroverlay: fix string leak
--   gtkwaylandsink: Destroy GstWlWindow when parent GtkWindow is
-    destroyed
--   gtkwaylandsink: Fix crash when rendering after the window is closed
--   ksvideo, directshow: Fix reference leaks in device providers
--   h264decoder: Fix DPB bumping process
--   h264decoder, h265decoder: Latency reporting related fixes
--   h264parse: Validate VUI framerate
--   jpegparse: reset parse state when the SOI is not the first marker
--   nvencoder: Fix CQP option setting
--   nvh264encoder: Fix template caps to include progressive mode as well
--   openjpegdec: allow multithread decoding only in subframe mode
--   tsdemux: Fix reading of extended Opus channel configuration
--   vulkan: fix validation layer issues
--   vulkanoverlaycompositor: fix potential use after free
--   vulkanswapper: correctly handle force-aspect-ratio=false
--   wasapi2: Fix potential crash on device activation failure
--   webrtc: Fix segfault traversing ice transports
--   webrtc: patch leak caused by early return
--   zxing: add support for zxing-c++ 2.0
+-   gst-plugins-rs requires Rust 1.70 or newer.
 
-gst-plugins-ugly
+-   Link to libsoup at build time in all cases on non-Linux, and only load it dynamically on Linux systems where we might need
+    to support a mix of applications with dependencies that might be using libsoup2 or libsoup3. A “soup-version” meson build
+    option was added to prefer a specific version. Distros should make sure that libsoup is still a package dependency, since
+    it’s still required at runtime for the soup and adaptivedemux2 plugins to function.
 
--   No changes
+-   libjack is now dynamically loaded at runtime by the JACK audio plugin, and no longer a hard build dependency. However, it
+    still is a runtime dependency, so distros should make sure it remains a package dependency.
 
-gst-libav
+Monorepo build (née gst-build)
 
--   avdec_h264 pipeline freeze with FFmpeg6
--   avdeinterlace, avmux: fix element reference leak
--   avviddec: Drop decoder stream lock when calling send_packet
+-   There is now a top-level meson build option to enable/require webrtc
 
-gst-rtsp-server
+-   The FFmpeg subproject wrap was udpated to 6.1
 
--   rtsp-server: fix deadlock on shutdown with non-live pipeline if
-    media isn’t playing/prerolled yet and eos-shutdown is enabled for
-    the media
+-   A libvpx wrap was added (for VP8/VP9 software encoding/decoding)
 
-gstreamer-vaapi
+gstreamer-full
 
--   No changes
+-   Add full static build support, including on Windows: Allow a project to use gstreamer-full as a static library and link to
+    create a binary without dependencies. Introduce the meson build option gst-full-target-type to select the build type:
+    dynamic (default) or static.
 
-gstreamer-sharp
+-   Registers all full features in a plugin now to offer the possibility to get information at the plugin level and get it from
+    the registry. All the full features are now registered in a fullstaticfeatures meta plugin instead of having a NULL plugin.
 
--   No changes
+Development environment
 
-gst-omx
+-   add VSCode IDE integration
 
--   No changes
+-   gst-env.py: Output a setting for the prompt with --only-environment
 
-gst-python
+### Cerbero
 
--   No changes
+Cerbero is a meta build system used to build GStreamer plus dependencies on platforms where dependencies are not readily
+available, such as Windows, Android, iOS, and macOS.
 
-gst-editing-services
+General improvements
 
--   No changes
+-   New plugins have been added
+    -   codecalpha dvbsubenc rtponvif switchbin videosignal isac ivfparse inter rtspsrc2
+-   JACK plugin is now available all platforms (previously only Linux), and will be loaded if the JACK library is found at
+    plugin load time
+-   Several recipes were ported to meson, leading to faster builds and better MSVC support
+    -   ffmpeg, gperf, lame, libvpx, ogg, opencore-amr, sbc, speex, tiff, webrtc-audio-processing
+    -   For more information, please see the tracker issue
+-   Some recipes are now outdated or unnecessary and have been removed:
+    -   intltool, libkate
+-   Various recipe updated to their latest versions
+-   Rust toolchain updated to 1.76.0 (latest as of writing)
+-   Rust plugins are now stripped and debug info split out correctly, reducing their size
+-   Fix several spurious build issues, especially with the Rust toolchain
+-   CMake is picked up from the system if available
+-   Cerbero will no longer OOM or consume excessive resources on low-end systems
+-   Python recipes have been moved from setuptools to virtualenv
+-   Fixed support for Python 3.12+
+
+macOS
+
+-   Minimum macOS version has been increased to 10.13 (High Sierra)
+    -   Released 5½ years ago, >95% marketshare
+-   Fix macOS app store validation issue caused by absolute RPATHs
+-   Rosetta is automatically installed if required for universal builds on Apple Silicon
+-   The official macOS binaries now also include static libs for the GStreamer Rust plugins
+
+iOS
+
+-   Minimum iOS version has been increased to 12.0
+-   The iOS binary packages now include the GStreamer Rust plugins
+-   tremor and ivorbisdec plugins are no longer shipped on iOS
 
-gst-validate + gst-integration-testsuites
+Windows
 
--   No changes
+-   New features shipped with the official binaries:
+    -   plugins: dwrite d3d12 (MSVC) soundtouch (MSVC) taglib (MSVC) webrtcdsp (MSVC)
+    -   plugin features: d3d11-wgc (Windows Graphics Capture Support)
+    -   libraries: gstdxva-1.0
+-   New qml6 plugin can be built on Windows with the qt6 variant enabled
+    -   Similar to qt5, this plugin cannot be included in the official binaries
+-   GLib process handling helpers for Windows are now shipped
+-   Windows 11 SDK is now required for builds
+    -   Visual Studio 2019 and newer ship this SDK
+-   MSYS is now deprecated for building Windows binaries, will be removed in the next release
+    -   MSYS2 is required, and the bootstrap script tools/bootstrap-windows.ps can install it for you
+-   Windows bootstrap script tools/bootstrap-windows.ps1 is much more interactive and user-friendly now
+-   Fixed Pango crash on 32-bit Windows
+-   WiX packaging now works with cross-windows builds from linux
 
-gst-examples
+Linux
 
--   No changes
+-   Linux packages will now also include static libs for the GStreamer Rust plugins
+-   Add Python support for multiarch distributions
+-   Build fixes for various recipes on multiarch distributions
+-   Use arch-specific libdir correctly on multiarch distributions
+-   gst-omx was removed from gstreamer, and hence is no longer shipped
+-   Fixed Gentoo support
+-   Added support for RHEL 9
+-   Added support for Rocky Linux
+-   Added support for Manjaro Linux
 
-Development build environment
+Android
+
+-   Android NDK has been updated to r25c
+    -   Only the Clang toolchain is used from the NDK now (both target and host)
+    -   gnustl has been completely removed
+-   The Android binary packages now include the GStreamer Rust plugins
+-   tremor and ivorbisdec plugins are no longer shipped on Android
+-   openh264 plugin no longer enables ASM optimizations on Android x86 due to relocation errors
+
+## Platform-specific changes and improvements
+
+### Android
+
+-   Add NDK implementation of Android MediaCodec. This reduces the amount of Java <-> native calls, which should reduce
+    overhead.
+
+-   Add support for AV1 to the androidmedia video encoder and decoder.
+
+### Apple macOS and iOS
+
+-   osxaudio: audio clock improvements (interpolate based on system time)
+
+-   Set activation policy in gst_macos_main() and in osxvideosink and glimagesink. Setting the policy to
+    NSApplicationActivationPolicyAccessory by default makes sure that we can activate windows programmatically or by clicking on
+    them. Without that, windows would disappear if you clicked outside them and there would be no way to bring them to front
+    again. This change also allows osxvideosink to receive navigation events correctly.
+
+### Windows
+
+-   New DirectWrite text rendering plugin with dwriteclockoverlay, dwritetimeoverlay, dwritetextoverlay, dwritesubtitlemux, and
+    dwritesubtitleoverlay elements, including closed caption overlay support in dwritetextoverlay.
+
+-   PTP clock support is now also available on Windows
+
+-   qml6d3d11sink is a new Direct3D11 Qt6 QML sink for Windows as an alternative to the existing qml6glsink.
+
+-   wasapi2 audio plugin:
+
+    -   Added an option to monitor a loopback device’s mute state
+    -   Allows process loopback capture on Windows 10
+
+-   win32ipc supports zero-copy rendering now through a shared bufferpool.
+
+-   Add a Windows-specific plugin loader implementation (gst-plugin-scanner), so plugin loading during registry updates happens
+    in an external process on Windows as well.
+
+-   gst_video_convert_sample() which is often used for thumbnailing gained a D3D11 specific conversion path.
+
+-   d3d11 plugin:
+
+    -   d3d11mpeg2dec element is promoted to PRIMARY + 1 rank
+    -   Added d3d11ipcsrc and d3d11ipcsink elements for zero-copy GPU memory sharing between multiple processes.
+    -   Added HLSL shader pre-compilation (at binary build time) support in MSVC build.
+    -   d3d11videosink and d3d11convert elements support 3D transform, MSAA (MultiSampling Anti-Aliasing) and anisotropic
+        sampling method.
+    -   Added support for more raw video formats by using compute shader. A list of supported raw video formats can be found in
+        the d3d11videosink plugin documentation.
+    -   Added d3d11overlay element for applications to be able to draw/blend an overlay image on Direct3D11 texture directly.
+
+-   New Direct3D12 plugin: From a video decoding, conversion, composition, and rendering point of view, this new d3d12 plugin is
+    feature-wise near equivalent to the d3d11 plugin. Notable differences between d3d12 and d3d11 are:
+    -   The GStreamer Direct3D12 integration layer is not exposed as a GStreamer API yet. Thus, other plugins such as amfcodec,
+        nvcodec, qsv, and dwrite are not integrated with d3d12 yet.
+    -   H.264 video encoding support via d3d12h264enc element.
+        -   Direct3D12 video encoding API requires Windows 11 or DirectX 12 Agility SDK
+    -   IPC, overlay, and deinterlace elements are not implemented in d3d12
+    -   Windows Graphics Capture API based screen capturing is not implemented in d3d12
+    -   In this release, MSVC is the only officially supported toolchain for the d3d12 plugin build.
+    -   All d3d12 elements are zero ranked for now. Users will need to adjust rank of each d3d12 element via GST_PLUGIN_RANK
+        environment or appropriate plugin feature APIs if they want these elements autoplugged.
+
+## Documentation improvements
+
+-   hotdoc has been updated to the latest version, and the theme has also been updated, which should fix various usability
+    issues.
+
+## Possibly Breaking Changes
+
+-   gst_plugin_feature_check_version() has been updated to fix unexpected version check behaviour for git versions. It would
+    return TRUE if the plugin version is for a git development version (e.g. 1.24.0.1) and the check is for the “next” micro
+    version (1.24.1). Now it no longer does this. This just brings the runtime version check in line with the build time version
+    checks which have been fixed some time ago.
+
+-   GstAudioAggregator and subclasses such as audiomixer now sync property values to output timestamp, which is what
+    GstVideoAggregator has been doing already since 2019, and which makes sense, as the properties need to change at every
+    output frame based on the output time because they may change even though the input frame is not changing.
+
+-   rtpac3depay now outputs audio/x-ac3 instead of audio/ac3 as that is the canonical media format in GStreamer. audio/ac3 is
+    still sometimes accepted as input format for backwards compatibility (e.g. in rtpac3pay or ac3parse), but shouldn’t be
+    output.
+
+-   timecodestamper: The “drop-frame” property now defaults to TRUE
 
--   git: prevent CRLF line ending conversion for patches to fix pango
-    subproject patching issues on Windows
+-   The NVIDIA desktop GPU decoders nvh264sldec, nvh265sldec, nvvp8sldec and nvvp9sldec were renamed to nvh264dec, nvh265dec,
+    nvvp8dec and nvvp9dec, respectively.
 
-Cerbero build tool and packaging changes in 1.22.2
+## Known Issues
 
--   build: retry rust build on SIGBUS errors too
--   Fix packaging of rust plugins on Android
--   Modern Gentoo distro adaptation
--   sbc: update to 2.0
--   speex: update to 1.2.1
-
-Contributors to 1.22.2
-
-Adrien De Coninck, Albert Sjölund, Alexande B, Antonio Rojas, Arun
-Raghavan, Bart Van Severen, Carlo Cabrera, Colin Kinloch, Edward Hervey,
-Guillaume Desmottes, Haihua Hu, He Junyan, Ilie Halip, Jordan Petridis,
-Josef Kolář, Lily Foster, Mathieu Duponchelle, Matt Feury, Matthew
-Waters, Maxim P. Dementyev, Michael Tretter, Nicolas Dufresne, Nirbheek
-Chauhan, Piotr Brzeziński, Robert Rosengren, Rouven Czerwinski,
-Sebastian Dröge, Seungha Yang, Shengqi Yu, Stéphane Cerveau, Talha Khan,
-Thibault Saunier, Tim-Philipp Müller, Víctor Manuel Jáquez Leal, Vivia
-Nikolaidou, Wang Chuan, Wojciech Kapsa,
-
-… and many others who have contributed bug reports, translations, sent
-suggestions or helped testing. Thank you all!
-
-List of merge requests and issues fixed in 1.22.2
-
--   List of Merge Requests applied in 1.22.2
--   List of Issues fixed in 1.22.2
-
-1.22.3
-
-The third 1.22 bug-fix release (1.22.3) was released on 19 May 2023.
-
-This release only contains bugfixes and it should be safe to update from
-1.22.x.
-
-Highlighted bugfixes in 1.22.3
-
--   avdec: fix occasional video decoder deadlock on seeking with FFmpeg
-    6.0
--   decodebin3: fix regression handling input streams without CAPS or
-    TIME segment such as e.g. udpsrc or `pushfilesrc
--   bluez: a2dpsink: fix Bluetooth SIG Certification test failures
--   osxvideosink: fix deadlock upon closing output window
--   qtdemux: fix edit list handling regression and AV1 codec box parsing
--   qtmux: fix extraction of CEA608 closed caption data from S334-1A
-    packets
--   rtspsrc: Fix handling of * control path
--   splitmux: timestamp handling improvements
--   v4l2videodec: Rework dynamic resolution change handling (needed for
-    IMX6 mainline codec)
--   videoflip: fix regression with automatically rotating video based on
-    tags
--   d3d11: many d3d11videosink and d3d11compositor fixes
--   webrtc, rtp: numerous data race fixes and stability fixes
--   various bug fixes, memory leak fixes, and other stability and
-    reliability improvements
+-   There are known issues with FFmpeg version 6.0.0 due to opaque passing being broken in that version. This affects at least
+    avdec_h264, but may affect other decoders as well. Versions before 6.0.0, and 6.0.1 or higher are not affected.
+
+-   gst-libav currently doesn’t build against the latest FFmpeg 7.0 release. This is being worked on and tracked in this “libav:
+    Fix compatibility with ffmpeg 7” Merge Request.
+
+## Statistics
+
+-   4643 commits
+
+-   2405 Merge Requests
+
+-   850 Issues
+
+-   290+ Contributors
+
+-   ~25% of all commits and Merge Requests were in Rust modules
+
+-   4747 files changed
+
+-   469633 lines added
+
+-   209842 lines deleted
+
+-   259791 lines added (net)
+
+## Contributors
+
+Aaron Boxer, Aaron Huang, Acky Xu, adixonn, Adrian Fiergolski, Adrien De Coninck, Akihiro Sagawa, Albert Sjölund, Alessandro
+Bono, Alexande B, Alexander Slobodeniuk, Alicia Boya García, amindfv, Amir Naghdinezhad, anaghdin, Anders Hellerup Madsen,
+Andoni Morales Alastruey, Antonio Kevo, Antonio Rojas, Arnaud Rebillout, Arnaud Vrac, Arun Raghavan, badcel, Balló György, Bart
+Van Severen, Bastien Nocera, Benjamin Gaignard, Bilal Elmoussaoui, Brad Hards, Camilo Celis Guzman, Carlo Cabrera, Carlos
+Falgueras García, Carlos Rafael Giani, Célestin Marot, Chao Guo, Charlie Blevins, Cheah, Vincent Beng Keat, Chris Degawa, Chris
+Spencer, Christian Curtis Veng, Christian Meissl, Christopher Degawa, Chris Wiggins, Cidana-Developers, Colin Kinloch, Damian
+Hobson-Garcia, Daniel Almeida, Daniel Knobe, Daniel Moberg, Daniel Morin, Daniel Pendse, Daniel Stone, Daniel Ulery, Dan
+Searles, Dario Marino Saccavino, Dave Patrick Caberto, David Craven, David Revay, David Rosca, David Svensson Fors, Detlev
+Casanova, Diego Nieto, Dominique Leroux, Dongyun Seo, Doug Nazar, Edward Hervey, Ekwang Lee, elenril, Elliot Chen, Enrique Ocaña
+González, Erik Fröbrant, Eva Pace, Evgeny Pavlov, Fabian Orccon, Felix Yan, Fernando Jiménez Moreno, Florian Zwoch, François
+Laignel, Frank Dana, Georges Basile Stavracas Neto, Guillaume Desmottes, Guillaume Gomez, Gwyn Ciesla, Haihua Hu, Hassene Tmar,
+hassount, Heiko Becker, He Junyan, hguermaz, Hiero32, Hosang Lee, Hou Qi, Hugo Svirak, Hugues Fruchet, Hu Qian, Hyung Song,
+Ignazio Pillai, Ilie Halip, Itamar Marom, Ivan Molodetskikh, Ivan Tishchenko, JackZhouVCD, Jacob Johnsson, jainl28patel, Jakub
+Adam, James Cowgill, James Hilliard, James Oliver, Jan Alexander Steffens (heftig), Jan Beich, Jan Schmidt, Jan Vermaete, Jayson
+Reis, Jeff Wilson, Jeongki Kim, Jeri Li, Jimmi Holst Christensen, Jimmy Ohn, Jochen Henneberg, Johan Adam Nilsson, Johan
+Sternerup, John King, Jonas Danielsson, Jonas K Danielsson, Jonas Kvinge, Jordan Petridis, Jordan Yelloz, Josef Kolář, Juan
+Navarro, Julien Isorce, Jun Zhao, Jurijs Satcs, Kalev Lember, Kamal Mostafa, kelvinhu325, Kevin Song, Khem Raj, Kourosh
+Firoozbakht, Leander Schulten, Leif Andersen, L. E. Segovia, Lieven Paulissen, lijing0010, Lily Foster, Link Mauve, Li Yuanheng,
+Loïc Le Page, Loïc Molinari, Lukas Geiger, Luke McGartland, Maksym Khomenko, Ma, Mingyang, Manel J, Marcin Kolny, Marc Leeman,
+Marc Solsona, Marc Wiblishauser, Marek Vasut, Marijn Suijten, Mark Hymers, Markus Ebner, Martin Nordholts, Martin Robinson, Mart
+Raudsepp, Marvin Schmidt, Mathieu Duponchelle, Matt Feury, Matthew Waters, Matthias Fuchs, Matthieu Volat, Maxim P. Dementyev,
+medithe, Mengkejiergeli Ba, Michael Bunk, Michael Gruner, Michael Grzeschik, Michael Olbrich, Michael Tretter, Michiel
+Westerbeek, Mihail Ivanchev, Ming Qian, Nader Mahdi, naglis, Nick Steel, Nicolas Beland, Nicolas Dufresne, Nirbheek Chauhan,
+Olivier Babasse, Olivier Blin, Olivier Crête, Omar Khlif, Onur Demiralay, Patricia Muscalu, Paul Fee, Pawel Stawicki, Peter
+Stensson, Philippe Normand, Philipp Zabel, PhoenixWorthVCD, Piotr Brzeziński, Priit Laes, Qian Hu, Rabindra Harlalka, Rafał
+Dzięgiel, Rahul T R, rajneeshksoni, Ratchanan Srirattanamet, renjun wang, Rhythm Narula, Robert Ayrapetyan, Robert Mader, Robert
+Rosengren, Robin Gustavsson, Roman Lebedev, Rouven Czerwinski, Ruben Gonzalez, Ruslan Khamidullin, Ryan Pavlik, Sanchayan Maity,
+Sangchul Lee, Scott Kanowitz, Scott Moreau, SeaDve, Sean DuBois, Sebastian Dröge, Sebastian Szczepaniak, Sergey Radionov,
+Seungha Yang, Shengqi Yu, Simon Himmelbauer, Slava Andrejev, Slawomir Pawlowski, soak, Stefan Brüns, Stéphane Cerveau, Stephan
+Seitz, Stijn Last, Talha Khan, Taruntej Kanakamalla, Jin Chung Teng, Théo Maillart, Thibault Saunier, Thomas Schneider,
+Tim-Philipp Müller, Tobias Rapp, Tong Wu, Tristan van Berkom, ttrigui, U. Artie Eoff, utzcoz, Víctor Manuel Jáquez Leal, Vivia
+Nikolaidou, Wang Chuan, William Manley, Wojciech Kapsa, Xabier Rodriguez Calvar, Xavier Claessens, Xuchen Yang, Yatin Maan,
+Yinhang Liu, Yorick Smilda, Yuri Fedoseev, Gang Zhao, Jack Zhou, …
+
+… and many others who have contributed bug reports, translations, sent suggestions or helped testing. Thank you all!
+
+Stable 1.24 branch
+
+After the 1.24.0 release there will be several 1.24.x bug-fix releases which will contain bug fixes which have been deemed
+suitable for a stable branch, but no new features or intrusive changes will be added to a bug-fix release usually. The 1.24.x
+bug-fix releases will be made from the git 1.24 branch, which will be a stable branch.
+
+1.24.0
+
+GStreamer 1.24.0 was released on 4 March 2024.
+
+1.24.1
+
+The first 1.24 bug-fix release (1.24.1) was released on 21 March 2024.
+
+This release only contains bugfixes and it should be safe to update from 1.24.0.
+
+Highlighted bugfixes in 1.24.1
+
+-   Fix instant-EOS regression in audio sinks in some cases when volume is 0
+-   rtspsrc: server compatibility improvements and ONVIF trick mode fixes
+-   rtsp-server: fix issues if RTSP media was set to be both shared and reusable
+-   (uri)decodebin3 and playbin3 fixes
+-   adaptivdemux2/hlsdemux2: Fix issues with failure updating playlists
+-   mpeg123 audio decoder fixes
+-   v4l2codecs: DMA_DRM caps support for decoders
+-   va: various AV1 / H.264 / H.265 video encoder fixes
+-   vtdec: fix potential deadlock regression with ProRes playback
+-   gst-libav: fixes for video decoder frame handling, interlaced mode detection
+-   avenc_aac: support for 7.1 and 16 channel modes
+-   glimagesink: Fix the sink not always respecting preferred size on macOS
+-   gtk4paintablesink: Fix scaling of texture position
+-   webrtc: Allow resolution and framerate changes, and many other improvements
+-   webrtc: Add new LiveKit source element
+-   Fix usability of binary packages on arm64 iOS
+-   various bug fixes, memory leak fixes, and other stability and reliability improvements
 
 gstreamer
 
--   tracing: Initialize tracing infrastructure even if the debug system
-    is not compiled in
--   parse-launch: fix missing unref of looked-up child element
--   gstutils: Add category and object to most logging messages
+-   registry, ptp: Canonicalize the library path returned by dladdr
+-   segment: Don’t use g_return_val_if_fail() in gst_segment_to_running_time_full()
+-   uri: Sort uri protocol sources/sinks by feature name to break a feature rank tie
+-   ptp: Initialize expected DELAY_REQ seqnum to an invalid value
+-   ptp: Don’t install test executable
+-   gst-inspect: fix –exists for plugins with versions other than GStreamer’s version, like the Rust plugins
+-   identity: Don’t refuse seeks unless single-segment=true
 
 gst-plugins-base
 
--   allocators: Fix fdmem unit test with recent GLib versions
--   audiotestsrc: Initialize all samples in wave=ticks mode
--   decodebin3: Handle input streams without CAPS or TIME segment such
-    as e.g. udpsrc or pushfilesrc
--   decodebin3: fix regression handling streams without caps
--   decodebin3: fix random hang when remove failing stream
--   uridecodebin3: Ensure atomic urisourcebin state change
--   glvideoflip: fix leaked caps
--   glcontext_wgl: fix missing unref
--   playsink: Fix volume leak
+-   audiobasesink: Don’t wait on gap events
+-   audioencoder: Avoid using temporarily mapped memory as base for input buffers
+-   decodebin3: Be more specific when sending missing plugin messages
+-   decodebin3: Fix re-usability issues
+-   decodebin3: Provide clear error message if no decoders present
+-   playbin3: Remove un-needed URI NULL check
+-   uridecodebin3: Don’t hold lock when posting messages or signals
+-   uridecodebin3: Handle potential double redirection errors
+-   glimagesink: Fix the sink not always respecting preferred size on macOS
+-   glupload: Do not propose allocators with sysmem, fixes warning when playing VP9 with alpha
+-   shmallocator: fix build on Illumos
+-   meson: Fix the condition to skip theoradec test
 
 gst-plugins-good
 
--   adaptivedemux2: fix critical when using an unsupported URI
--   dashdemux2: mpdclient: fix divide by 0 if segment has no duration
--   imagesequencesrc: Properly set default location
--   multifile: error out if no filename was set
--   osxvideosink: fix deadlock upon closing output window
--   rtpmanager: rtpsession: data race leading to critical warnings
--   rtpmanager: rtpsession: race conditions leading to critical warnings
--   rtspsrc: Fix handling of * control path
--   splitmuxsink: Catch invalid DTS to avoid running into problems later
--   splitmuxsrc: Make PTS contiguous by preference
--   qtdemux: emit no-more-pads after pruning old pads
--   Revert “qtdemux: fix conditions for end of segment in reverse
-    playback” to fix edit list regression
--   qtdemux: Fix av1C parsing
--   qtmux: Fix extraction of CEA608 data from S334-1A packets
--   qtwindow: unref caps in destructor
--   v4l2: device provider: Fix GMainLoop leak
--   v4l2: videodec: Rework dynamic resolution change handling
--   v4l2: videodec: Prefer acquired caps over anything downstream
--   videoflip: fix setting of method property at construction time
--   videoflip 1.22.2 not rotating video when extracting frames
+-   adaptivdemux/hlsdemux2: Fix issues with failure updating playlists
+-   mpg123audiodec: Correctly handle the case of clipping all decoded samples
+-   mpg123audiodec: gst_audio_decoder_allocate_output_buffer: assertion ‘size > 0’ failed
+-   qt: Fix description in meson build options
+-   qtdemux: Do not set channel-mask to zero
+-   rtspsrc: remove ‘deprecated’ flag from the ‘push-backchannel-sample’ signal
+-   rtspsrc: Consider 503 Service Not Available when handling broken control urls
+-   rtspsrc, rtponviftimestamp: ONVIF mode fixes
+-   rtspsrc: Don’t invoke close when stopping if we’ve started cleanup, fixing potential crash on shutdown
+-   rtpgstpay: Delay pushing of event packets until the next buffer
 
 gst-plugins-bad
 
--   a2dpsink: Fails many tests in Bluetooth SIG Certification
--   avdtputil: Use int instead of int range for fixed bitpool values
--   ccconverter: reintroduce frame count reset on cycle completion
--   ccconverter: integer overflow & crashing
--   codectimestamper: remove PC file generation from plugin’s own
-    meson.build
--   cudamemory: Fix for semi planar YUV memory size decision
--   d3d11compositor: Reconfigure resource only when output caps is
-    changed
--   d3d11compositor: Skip zero alpha input
--   d3d11convert: Fix for runtime property update
--   d3d11memory: Don’t clear wrapped texture memory
--   d3d11videosink: Fix for ignored initial render rectangle
--   d3d11videosink: fix race condition in window unprepare
--   d3d11videosink: Enhancement for initial window size decision
--   d3d11videosink: Don’t clear prepared buffer on unlock_stop()
--   dashdemux: mpdclient: fix divide by 0 if segment has no duration
--   dtlstransport: Keep strong ref of dtls encoder/decoder
--   GstPlay: avoid getting property of playbin2 if subtitle_sid is null
--   GstPlay: fix critical log when using playbin3
--   h264decoder: Drop nonexisting picture silently without error
--   dtmf: element classification improvements
--   mfvideoenc: Allow only even resolution numbers
--   sctpenc: Fix potential shutdown deadlock
--   srtpdec: fix “srtp-key” check
--   tests: disable dtls test if openssl is not present
--   tsdemux: Set number of channels to 2 for dual mono Opus
--   va: Various fixes for defects found with MSVC
--   wasapi2: Allows process loopback capture on Windows 10
--   webrtcdatachannel: Bind to parent webrtcbin using a weak reference
--   webrtcbin: Fix potential deadlock when closing before any data was
-    sent
--   webrtc: Plug leaks of resolved ICE addresses
--   webrtc: do not tear down data channel before data is flushed
+-   asio: Fix {input,output}-channels property handling
+-   cuda,d3d11,d3d12bufferpool: Disable preallocation
+-   d3d11device: Fix adapter LUID comparison in wrapped device mode
+-   d3d12device: Fix IDXGIFactory2 leak
+-   d3d12: Fix SDK debug layer activation
+-   dvbsubenc: Fix bottom field size calculation
+-   dvdspu: avoid null dereference
+-   GstPlay: Fix a critical warning in error callback
+-   v4l2codecs: decoders: Add DMA_DRM caps support
+-   vaav1enc: Init the output_frame_num when resetting gf group
+-   vah264enc, vah265enc, vaav1enc: fix potential crash on devices without rate control
+-   vah265enc: checking surface alignment
+-   videoparsers: Don’t verbosely warn about CEA_708_PROCESS_EM_DATA_FLAG not being set
+-   vtdec: Fix a deadlock during ProRes playback, handle non-linked gracefully
 
 gst-plugins-ugly
 
--   mpegpsdemux: Rework gap sending
+-   No changes
+
+GStreamer Rust plugins
+
+-   gtk4paintablesink: Fix scaling of texture position
+-   janusvrwebrtcsink: Handle 64 bit numerical room ids
+-   janusvrwebrtcsink: Don’t include deprecated audio/video fields in publish messages
+-   janusvrwebrtcsink: Handle various other messages to avoid printing errors
+-   livekitwebrtc: Fix shutdown behaviour
+-   rtpgccbwe: Don’t forward buffer lists with buffers from different SSRCs to avoid breaking assumptions in rtpsession
+-   sccparse: Ignore invalid timecodes during seeking
+-   webrtcsink: Don’t try parsing audio caps as video caps
+-   webrtc: Allow resolution and framerate changes
+-   webrtcsrc: Make producer-peer-id optional
+-   livekitwebrtcsrc: Add new LiveKit source element
+-   regex: Add support for configuring regex behaviour
+-   spotifyaudiosrc: Document how to use with non-Facebook accounts
+-   webrtcsrc: Add do-retransmission property
 
 gst-libav
 
--   avviddec: Temporarily unlock stream lock while flushing buffers
--   Random freeze and deadlock in ffmpegviddec flush and get_buffer
-    while seeking
+-   avcodecmap: Increase max AAC channels to 16
+-   avviddec: Fix how we get back the codec frame
+-   avviddec: Fix interlaced mode detection
+-   avviddec: Double check if AV_CODEC_FLAG_COPY_OPAQUE port is safe for our scenario
 
 gst-rtsp-server
 
--   No changes
+-   media: gst_rtsp_media_set_reusable() and gst_rtsp_media_set_shared() have become incompatible
+-   rtsp-stream: clear sockets when leaving bin
 
 gstreamer-vaapi
 
@@ -1790,8 +1479,7 @@
 
 gst-editing-services
 
--   ges: base-xml-formatter: Don’t pass non-GObject pointers to
-    GST_DEBUG_OBJECT
+-   ges: Fix critical warning
 
 gst-validate + gst-integration-testsuites
 
@@ -1805,140 +1493,125 @@
 
 -   No changes
 
-Cerbero build tool and packaging changes in 1.22.3
+Cerbero build tool and packaging changes in 1.24.1
+
+-   gstreamer: Enable ptp helper explicitly
+-   gst-plugins-bad: Package new insertbin plugin
+-   gst-plugins-rs: Adjust parallel architecture build blocks
+-   libnice: update to 0.1.22
+-   pixman: Bump to 0.43.4
+-   orc: disable JIT code generation on arm64 on iOS again, fixing crashes
 
--   glib: Ship Windows process spawning helpers
--   recipes: add recipe for libltc for timecodestamper element
--   Add support for RHEL9 and Rocky Linux
-
-Contributors to 1.22.3
-
-Aleksandr Slobodeniuk, Antonio Kevo, Arun Raghavan, Carlos Rafael Giani,
-Daniel Moberg, Edward Hervey, Elliot Chen, François Laignel, Guillaume
-Desmottes, Haihua Hu, Jan Alexander Steffens (heftig), Jan Beich, Jan
-Schmidt, Johan Sternerup, John King, Jordan Petridis, Juan Navarro, Lily
-Foster, Martin Nordholts, Mathieu Duponchelle, Matthew Waters, Matthias
-Fuchs, Michael Olbrich, Mihail Ivanchev, Nick Steel, Nicolas Dufresne,
-Nirbheek Chauhan, Patricia Muscalu, Philippe Normand, Piotr Brzeziński,
-Sanchayan Maity, Sebastian Dröge, Seungha Yang, Stéphane Cerveau,
-Thibault Saunier, Tim-Philipp Müller, Xabier Rodriguez Calvar,
-
-… and many others who have contributed bug reports, translations, sent
-suggestions or helped testing. Thank you all!
-
-List of merge requests and issues fixed in 1.22.3
-
--   List of Merge Requests applied in 1.22.3
--   List of Issues fixed in 1.22.3
-
-1.22.4
-
-The fourth 1.22 bug-fix release (1.22.4) was released on 20 June 2023.
-
-This release only contains bugfixes and security fixes and it should be
-safe to update from 1.22.x.
-
-Highlighted bugfixes in 1.22.4
-
--   Security fixes for flacparse, dvdspu, and subparse
--   d3d11videosink: Fix error on pause and play
--   decklink: Correctly handle SDK strings on macOS and free strings
-    after usage on Linux
--   filesink: Fix buffered mode writing of buffer lists and buffers with
-    multiple memories
--   gldownload: handle passthrough without a critical
--   h265parse: Fix framerate handling regression
--   oggdemux: vp8 fixes
--   mp4mux, qtmux, qtdemux: Opus audio mapping fixes
--   pngdec: Fix wrong colours output from 16bit RGB images
--   ptp clock: Work around ptpd bug in default configuration
--   srtpdec: fix critical warnings on shutdown
--   v4l2src: fix support for bayer format
--   v4l2videoenc: support force-keyframe event in v4l2 encoder
--   vtenc: apply DTS offset to ensure DTS <= PTS
--   gst-python: allow more functions to be called before gst_init()
--   cerbero: fix vaapi variant; add qt6 build on windows; ensure errors
-    on unguarded use of new APIs, require macOS 10.13
--   packages: ship codecalpha, rtponvif, dvbsubenc, switchbin,
-    videosignal plugins; fix pango crash on 32-bit windows
--   various bug fixes, memory leak fixes, and other stability and
-    reliability improvements
+Contributors to 1.24.1
+
+Alexander Slobodeniuk, Antonio Larrosa, Edward Hervey, Elizabeth Figura, François Laignel, Guillaume Desmottes, He Junyan, Jan
+Schmidt, Jordan Yelloz, L. E. Segovia, Mark Nauwelaerts, Mathieu Duponchelle, Michael Tretter, Mikhail Rudenko, Nicolas
+Dufresne, Nirbheek Chauhan, Philippe Normand, Piotr Brzeziński, Robert Mader, Ruijing Dong, Sebastian Dröge, Seungha Yang,
+Thomas Goodwin, Thomas Klausner, Tim-Philipp Müller, Xi Ruoyao,
+
+… and many others who have contributed bug reports, translations, sent suggestions or helped testing. Thank you all!
+
+List of merge requests and issues fixed in 1.24.1
+
+-   List of Merge Requests applied in 1.24.1
+-   List of Issues fixed in 1.24.1
+
+1.24.2
+
+The second 1.24 bug-fix release (1.24.2) was released on 9 April 2024.
+
+This release only contains bugfixes and it should be safe to update from 1.24.x.
+
+Highlighted bugfixes in 1.24.2
+
+-   H.264 parsing regression fixes
+-   WavPack typefinding improvements
+-   Video4linux fixes and improvements
+-   Android build and runtime fixes
+-   macOS OpenGL memory leak and robustness fixes
+-   Qt/QML video sink fixes
+-   Package new analytics and mse libraries in binary packages
+-   Windows MSVC binary packages: fix libvpx avx/avx2/avx512 instruction set detection
+-   various bug fixes, memory leak fixes, and other stability and reliability improvements
 
 gstreamer
 
--   filesink: Fix buffered mode writing of buffer lists and buffers with
-    multiple memories
--   basesink: Clear EOS flag on STREAM-START event
--   typefindhelper: downgrade bogus error level debug log message
--   ptp: Correctly parse clock ID from the commandline parameters in the
-    helper
--   ptp: Work around bug in ptpd in default configuration
+-   clock: Block futex_time64 usage on Android API level < 30
+-   basesrc: Clear submitted buffer lists consistently with buffers
+-   ptpclock: fix double free of domain data during deinit
+-   clocksync: Proxy allocation queries
+-   inputselector: fix possible clock leak on shutdown
+-   typefind: Handle WavPack block sizes > 131072
 
 gst-plugins-base
 
--   alsasink: Fix stall for transition from PAUSED to READY with USB
-    speakerphone.
--   appsink: unref buffer in prev sample early so buffers from v4l2 can
-    be released properly
--   basetextoverlay: Fix typo in “text-y” property description
--   gldownload: handle passthrough without a critical
--   glfilter: add parent meta to output buffer for input buffer
--   oggdemux: vp8: Push headers downstream and detect keyframe packets
--   opus: Fix potential crash when getting unexpected channel position
--   streamsynchronizer: reset eos on STREAM_START
--   subparse: Look for the closing > of a tag after the opening <
--   video: convertframe: Add D3D11 specific conversion path
--   videometa: Only validate the alignment only when it contains some
-    info
--   video-blend: Fix linking error with C++
+-   glcolorconvert: Ensure glcolorconvert does not miss supported RGB formats
+-   gl/macos: a couple of race/reference count fixes
+-   pbutils: descriptions: Don’t warn on MPEG-1 audio caps without layer field
+-   encodebin: Add the parser before timestamper to tosync list
+-   videorate: Reset last_ts when a new segment is received
 
 gst-plugins-good
 
--   flacparse: Avoid integer overflow in available data check for image
-    tags
--   flvmux: use the correct timestamp to calculate wait times
--   isomp4: Fix (E)AC-3 channel count handling
--   jpegdec: fixes related to interlaced jpeg
--   pngdec: Fix wrong colours output from 16bit RGB images
--   qtmux, qtdemux: fix byte order for opus extension
--   rtspsrc: Do not try send dropped get/set parameter
--   qt5, qt6: Add more meson options and eliminate all automagic
--   qt: glrenderer: don’t attempt to use QWindow from non-Qt main thread
--   qml6glsink: Support building on win32
--   v4l2src: fix support for bayer format
--   v4l2: Change to query only up to
-    V4L2_CID_PRIVATE_BASE+V4L2_CID_MAX_CTRLS
--   v4l2videodec: treat MPEG-1 format as MPEG-2
--   v4l2videoenc: support force keyframe event in v4l2 encoder
--   tests: rtpbin_buffer_list: fix possible unaligned write/read on
-    32-bit ARM
+-   qml6glsink: fix destruction of underlying texture
+-   qt/qt6: Fixup for dummy textures
+-   rtpjitterbuffer: Don’t use estimated_dts to do default skew adjustment
+-   rtpjitterbuffer: Use an extended RTP timestamp for the clock-base
+-   rtpmp4adepay: Set duration on outgoing buffers
+-   tests: rtpred: fix out-of-bound writes
+-   v4l2: allocator: Fix unref log/trace on memory release
+-   v4l2: Also set max_width/max_width if enum framesize fail
+-   v4l2: enforce a pixel aspect ratio of 1/1 if no data are available
+-   v4l2: fix error in calculating padding bottom for tile format
+-   v4l2src: need maintain the caps order in caps compare when fixate
+-   vpxenc: Include vpx error details in errors and warnings
 
 gst-plugins-bad
 
--   asfmux: fix possible unaligned write on 32-bit ARM
--   d3d11videosink: Fix error on pause and play
--   d3dvideosink: Fix navigation event leak
--   decklink: Correctly handle SDK strings on macOS and free strings
-    after usage on Linux
--   dvdspu: Make sure enough data is allocated for the available data
--   fdkaacdec: Support up to 5 rear channels
--   h265parse: Fix framerate handling
--   kmssink: Add STM32 LTDC and NXP i.MX8M Plus LCDIFv3 auto-detection
--   sdpdemux: ensure that only one srcpad is created per stream
--   srtpdec: fix critical warnings on shutdown
--   testsrcbin: Remove spurious caps unref
--   va: map the mbbrc to correct enum value in get_property()
--   vtenc: apply DTS offset to ensure DTS <= PTS
--   vtdec: time glitches on h264 playback
--   waylandsink: Emit “map” signal boarder surface is ready
+-   h264parse: element hangs with some video streams (regression)
+-   h264parse: Revert “AU boundary detection changes”
+-   alphadecodebin: Explicitly pass 64 bit integers as such through varargs
+-   atdec: Set a channel mask for channel counts greater than 2
+-   ccconverter: Fix caps leak and remove unnecessary code
+-   d3d11videosink: disconnect signals before releasing the window
+-   d3d11: meson: Add support for MinGW DirectXMath package and update directxmath wrap to 3.1.9
+-   d3d11: meson: Disable library build if DirectXMath header was not found
+-   dwrite: Fix crash on device update
+-   GstPlay: Update video_snapshot to support playbin3
+-   jpegparse: avi1 tag can be progressive
+-   jpegparse: turn some bus warnings into object ones
+-   qsvdecoder: Release too old frames
+-   ristsrc: Only free caps if needed
+-   va: av1enc: Correct the reference number and improve the reference setting
+-   va: {vp9, av1}enc: Avoid reopen encoder or renegotiate
+-   videoparsers: Demote CC warning message
+-   vkbufferpool: correct usage flags type
+-   vkh26xdec: a couple decoding fixes
+-   vtdec: Fix caps criticals during negotiation
+-   wpe: avoid crash with G_DEBUG=fatal_criticals and static build
+-   Sink missing floating references
 
 gst-plugins-ugly
 
 -   No changes
 
+GStreamer Rust plugins
+
+-   aws: use fixed BehaviorVersion
+-   aws: improve error message logs
+-   fmp4: Update to dash-mpd 0.16
+-   fmp4mux: Require gstreamer-pbutils 1.20 for the examples
+-   onvifmetadataparse: Reset state in PAUSED->READY after pad deactivation, fixing occasional deadlock on shutdown
+-   reqwest: Update to reqwest 0.12
+-   webrtcsink: set perfect-timestamp=true on audio encoders
+-   webrtcsink: improve panic message on unexpected caps during discovery
+-   webrtchttp: Update to reqwest 0.12
+-   webrtc: fix inconsistencies in documentation of object names
+-   Fix clippy warnings after upgrade to Rust 1.77
+
 gst-libav
 
--   No changes
+-   avviddec: Fix AVPacket leak
 
 gst-rtsp-server
 
@@ -1946,7 +1619,7 @@
 
 gstreamer-vaapi
 
--   vaapidecodebin: don’t load vaapipostproc if not available
+-   No changes
 
 gstreamer-sharp
 
@@ -1958,11 +1631,11 @@
 
 gst-python
 
--   python: More functions can be called before gst_init()
+-   No changes
 
 gst-editing-services
 
--   ges: launcher: Never put sinks in a GstPipeline
+-   ges: frame-composition-meta: Stop using keyword ‘operator’ for field in C++
 
 gst-validate + gst-integration-testsuites
 
@@ -1970,159 +1643,147 @@
 
 gst-examples
 
--   No changes
+-   webrtc examples: set perfect-timestamp=true on opusenc for better Chrome interoperability
 
 Development build environment
 
--   No changes
+-   flac: Add subproject wrap and allow falling back to it in the flac plugin
+-   libnice: bump subproject wrap to v0.1.22 (needed for ICE consent freshness support in gstwebrtc)
 
-Cerbero build tool and packaging changes in 1.22.4
+Cerbero build tool and packaging changes in 1.24.2
 
--   Ship codecalpha, rtponvif, dvbsubenc, switchbin, videosignal plugins
--   pango: Fix crash on Windows 32bit build
--   qml6: Add support for building the qml6 plugin on Windows and bump
-    meson to 1.1.1
--   vaapi: update vaapi variant/recipe for meson option changes
--   packages: Put libass in the same category as assrender
--   cerbero: Don’t extract if already extracted in fetch
--   darwin: Ensure errors on unguarded use of new APIs, require macOS
-    10.13
-
-Contributors to 1.22.4
-
-Andoni Morales Alastruey, Arun Raghavan, Colin Kinloch, Daniel Morin,
-Edward Hervey, ekwange, Elliot Chen, François Laignel, Guillaume
-Desmottes, Haihua Hu, He Junyan, Hou Qi, Jan Alexander Steffens
-(heftig), Jochen Henneberg, Jordan Petridis, Kevin Song, Maksym
-Khomenko, Marek Vasut, Mathieu Duponchelle, Matthew Waters,
-Mengkejiergeli Ba, Michael Olbrich, Nicolas Beland, Nicolas Dufresne,
-Nirbheek Chauhan, Philippe Normand, Piotr Brzeziński, Sebastian Dröge,
-Seungha Yang, Thibault Saunier, Tim-Philipp Müller, Víctor Manuel Jáquez
-Leal, William Manley, Xavier Claessens, Yuri Fedoseev,
-
-… and many others who have contributed bug reports, translations, sent
-suggestions or helped testing. Thank you all!
-
-List of merge requests and issues fixed in 1.22.4
-
--   List of Merge Requests applied in 1.22.4
--   List of Issues fixed in 1.22.4
-
-1.22.5
-
-The fifth 1.22 bug-fix release (1.22.5) was released on 20 July 2023.
-
-This release only contains bugfixes and security fixes and it should be
-safe to update from 1.22.x.
-
-Highlighted bugfixes in 1.22.5
-
--   Security fixes for the RealMedia demuxer
--   vaapi decoders, postproc: Disable DMAbuf from caps negotiation to
-    fix garbled video in some cases
--   decodebin3, playbin3, parsebin fixes, especially for stream
-    reconfiguration
--   hlsdemux2: fix early seeking; don’t pass referer when updating
-    playlists; webvtt fixes
--   gtk: Fix critical caused by pointer movement when stream is getting
-    ready
--   qt6: Set sampler filtering method, fixes bad quality with qml6glsink
-    and gstqt6d3d11
--   v4l2src: handle resolution change when buffers are copied
--   videoflip: update orientation tag in auto mode
--   video timecode: Add support for framerates lower than 1fps and
-    accept 119.88 (120/1.001) fps
--   webrtcsink: fixes for x264enc and NVIDIA encoders
--   cerbero: Pull ninja from system if possible, avoid spurious
-    bootstrap of cmake
--   packages: Recipe updates for ffmpeg, libsoup, orc
--   various bug fixes, memory leak fixes, and other stability and
-    reliability improvements
+-   glib: Block futex_time64 usage on Android API level < 30
+-   libvpx: Fix build with Python 3.8
+-   libvpx: Fix errors with avx* instruction set detection for x86* builds and MSVC
+-   openjpeg: Update to 2.5.2
+-   directxmath: Update to 3.1.9
+-   gst-plugins-rs: Fix superstripping for ELF breaking all plugins
+-   Rust-based plugin initialization hangs on Android with GStreamer 1.24.0
+
+Contributors to 1.24.2
+
+Alexander Slobodeniuk, Arnaud Vrac, Chao Guo, Chris Spencer, Daniel Morin, Edward Hervey, Elizabeth Figura, Elliot Chen, eri,
+François Laignel, Guillaume Desmottes, Haihua Hu, He Junyan, Hou Qi, Jan Schmidt, Jochen Henneberg, L. E. Segovia, Martin
+Nordholts, Matthew Waters, Nicolas Dufresne, Philippe Normand, Philipp Zabel, Piotr Brzeziński, Robert Guziolowski, Robert
+Mader, Ruben Gonzalez, Sebastian Dröge, Seungha Yang, Taruntej Kanakamalla, Thibault Saunier, Tim Blechmann, Tim-Philipp Müller,
+Víctor Manuel Jáquez Leal, Wojciech Kapsa, Xavier Claessens,
+
+… and many others who have contributed bug reports, translations, sent suggestions or helped testing. Thank you all!
+
+List of merge requests and issues fixed in 1.24.2
+
+-   List of Merge Requests applied in 1.24.2
+-   List of Issues fixed in 1.24.2
+
+1.24.3
+
+The third 1.24 bug-fix release (1.24.3) was released on 30 April 2024.
+
+This release only contains bugfixes and it should be safe to update from 1.24.x.
+
+Highlighted bugfixes in 1.24.3
+
+-   EXIF image tag parsing security fixes
+-   Subtitle handling improvements in parsebin
+-   Fix issues with HLS streams that contain VTT subtitles
+-   Qt6 QML sink re-render and re-sizing fixes
+-   unixfd ipc plugin timestamp and segment handling fixes
+-   vah264enc, vah265enc: Do not touch the PTS of the output frame
+-   vah264dec and vapostproc fixes and improvements
+-   v4l2: multiple fixes and improvements, incl. for mediatek JPEG decoder and v4l2 loopback
+-   v4l2: fix hang after seek with some v4l2 decoders
+-   Wayland sink fixes
+-   ximagesink: fix regression on RPi/aarch64
+-   fmp4mux, mp4mux gained FLAC audio support
+-   D3D11, D3D12: reliablity improvements and memory leak fixes
+-   Media Foundation device provider fixes
+-   GTK4 paintable sink improvements including support for directly importing dmabufs with GTK 4.14
+-   WebRTC sink/source fixes and improvements
+-   AWS s3sink, s3src, s3hlssink now support path-style addressing
+-   MPEG-TS demuxer fixes
+-   Python bindings fixes
+-   various bug fixes, memory leak fixes, and other stability and reliability improvements
 
 gstreamer
 
--   taglist, plugins: fix compiler warnings with GLib >= 2.76
--   tracerutils: allow casting parameter types
--   inputselector: fix playing variable is never set
+-   ptp: Silence Rust compiler warning about some unused trait methods
 
 gst-plugins-base
 
--   appsink: add missing make_writable call
--   audioaggregator: Do not post message before being constructed
--   decodebin3: Prevent a critical warning when reassigning output slots
--   decodebin3: Fix slot input linking when the associated stream has
-    changed
--   decodebin3: Remove spurious input locking during parsebin
-    reconfiguration
--   urisourcebin: Set source element to READY before querying it
--   gl/viv-fb: meson build updates
--   plugins: fix compiler warnings with GLib >= 2.76
--   subtitleoverlay: fix mutex error if sink caps is not video
--   video: timecode: Add support for framerates lower than 1fps
--   video: accept timecode of 119.88 (120/1.001) FPS
--   video: cannot attach time code meta when frame rate is 119.88
-    (120000/1001)
--   videodecoder: fix copying buffer metas
+-   EXIF image tag parsing security fixes
+-   glcolorconvert: remove some dead code
+-   parsebin: Ensure non-time subtitle streams get “parsed”
+-   playbin3: Handle combiner update in case of errors
+-   ximagesink: initialize mask for XISelectEvents
 
 gst-plugins-good
 
--   adaptivedemux2: Fix early seeking
--   hlsdemux2: Ensure processed webvtt ends with empty new line
--   hlsdemux2: Don’t set a referer when updating playlists
--   matroska: demux: Strip signal byte when encrypted
--   rtspsrc: Fix crash when is-live=false
--   gtk: Fix critical caused by pointer movement when stream is getting
-    ready
--   qt6: Set sampler filtering method, fixes bad quality with qml6glsink
-    and gstqt6d3d11
--   qtdemux: opus: set entry as sampled
--   v4l2src: handle resolution change when buffers are copied
--   v4l2videodec: Fix handling of initial gaps
--   v4l2videodec: correctly register v4l2mpeg2dec
--   v4l2videoenc: replace custom QUERY_CAPS handling with getcaps
-    callback
--   videoflip: update orientation tag in auto mode
--   videoflip: fix critical when tag list is not writable
+-   adaptivedemux2: Playback hangs with VTT fragments
+-   adaptivedemux2: Avoid double usage of parsebin
+-   pulsedeviceprovider: Add compare_device_type_name function and missing lock
+-   qml6glsink: Notify that the returned QSGNode node has changes
+-   qml6glsink: video content resizes to new item size
+-   qtdemux: fix wrong full_range offset when parsing colr box
+-   soup: fix thread name
+-   v4l2: add multiplane y42b (yuv422m) support (for mediatek v4l2 jpeg decoder)
+-   v4l2: bufferpool: Drop writable check on output pool process
+-   v4l2: bufferpool: Ensure freshly created buffers are not marked as queued, fixing issues with v4l2sink on a v4l2loopback
+    device
+-   v4l2: bufferpool: queue back the buffer flagged LAST but empty, fixes hangs after seek with some decoders
+-   v4l2: silence valgrind warning
+-   vpx: Include vpx error details in errors and warnings
 
 gst-plugins-bad
 
--   d3d11bufferpool: Fix heavy CPU usage in case of fixed-size pool
--   jpegparser: jpegdecoder: Don’t pollute bus and comply with spec
--   plugins: fix compiler warnings with GLib >= 2.76
--   webrtcbin: Prevent critical warning when creating an additional data
-    channel
--   webrtcstats: Properly report IceCandidate type
+-   d3d11device: protect device_lock vs device_new
+-   d3d11decoder, d3d12decoder: Fix potential use after free
+-   d3d11videosink: Fix rendering on keyed mutex enabled handle
+-   d3d12decoder: Fix d3d12 resource copy
+-   d3d12encoder: Fix buffer pool leak
+-   d3d12videosink: HWND event handling related fixes
+-   d3d12vp9dec: Fix Intel GPU crash occurred when decoding VP9 SVC
+-   dvbsubenc: fixed some memory leaks and a crash
+-   GstPlay: fix read duration failure issue for some type rtsp streams which have valid duration
+-   mediafoundation: Fix device enumeration
+-   mediafoundation: Fix infinite loop in device provider
+-   tests: fix possible libscpp build failure in gst-plugins-bad
+-   tsdemux, tsparse: Fix Program equality check
+-   tsdemux: Disable smart program update
+-   unixfdsink: Take segment into account when converting timestamps
+-   va: videoformat: use video library to get DRM fourcc
+-   va: radeonsi: DRM RGB formats doesn’t look correctly mapped to VA formats
+-   vah264enc, vah265enc: Do not touch the PTS of output frame
+-   vaav1enc: Change the alignment of output to “tu”
+-   vaallocator: disable derived all together for Mesa <23.3
+-   waylandsink: free staged buffer when do gst_wl_window_finalize
+-   wlwindow: clear configure mutex and cond when finalize
+-   waylandsink: config buffer pool with query size when propose_allocation
+-   v4l2codecs: Don’t unref allocation query caps
 
 gst-plugins-ugly
 
--   rmdemux: add some integer overflow checks
+-   No changes
+
+GStreamer Rust plugins
 
-gst-plugins-rs
+Fixed:
 
--   fallbackswitch: Change the threshold for trailing buffers
--   fallbackswitch: Fix pad health calculation and notifies
--   fmp4mux: Fix draining in chunk mode if keyframes are too late
--   livesync: Wait for the end timestamp of the previous buffer before
-    looking at queue
--   livesync: Improve EOS handling
--   togglerecord: Clip segment before calculating timestamp/duration
--   togglerecord: Error out if main stream buffer has no valid running
-    time
--   webrtcsink: fix pipeline when input caps contain max-framerate
--   webrtcsink: Configure only 4 threads for x264enc
--   webrtcsink: Translate force-keyunit events to force-IDR action
-    signal for NVIDIA encoders
--   webrtcsink: Set config-interval=-1 and aggregate-mode=zero-latency
-    on rtph264pay and rtph265pay
--   webrtcsink: Set VP8/VP9 payloader based on payloader element factory
-    name
--   webrtcink: Use correct property types for nvvideoconvert
--   webrtc/signalling: fix race condition in message ordering
--   videofx: Minimize dependencies of the image crate
+-   hrtfrender: Use a bitmask instead of an int in the caps for the channel-mask
+-   rtpgccbwe: Don’t log an error when pushing a buffer list fails while stopping
+-   webrtcsink: Don’t panic in bitrate handling with unsupported encoders
+-   webrtcsink: Don’t panic if unsupported input caps are used
+-   webrtcsrc: Allow a None producer-id in request-encoded-filter signal
+
+Added:
+
+-   aws: New property to support path-style addressing
+-   fmp4mux / mp4mux: Support FLAC inside (f)MP4
+-   gtk4: Support directly importing dmabufs with GTK 4.14
+-   gtk4: Add force-aspect-ratio property similar to other video sinks
 
 gst-libav
 
--   No changes
+-   libav: guard some recently dropped APIs
 
 gst-rtsp-server
 
@@ -2130,7 +1791,7 @@
 
 gstreamer-vaapi
 
--   vaapidecode,vaapipostproc: Disable DMAbuf from caps negotiation.
+-   No changes
 
 gstreamer-sharp
 
@@ -2142,17 +1803,17 @@
 
 gst-python
 
--   No changes
+-   Fix Python library name fetching and library searching on Windows
+-   Don’t link to python for the gi overrides module
 
 gst-editing-services
 
--   ges: some fixes for 32-bit systems
--   ges, nle: Avoid setting state or sending query when constructing
-    objects
+-   ges-launcher: Fix for forcing container profiles
 
-gst-validate + gst-integration-testsuites
+gst-devtools, gst-validate + gst-integration-testsuites
 
--   No changes
+-   debug-viewer: Fix plugin loading machinery
+-   validate/flvdemux: Stop spamming audio/video on test
 
 gst-examples
 
@@ -2162,146 +1823,112 @@
 
 -   No changes
 
-Cerbero build tool and packaging changes in 1.22.5
+Cerbero build tool and packaging changes in 1.24.3
+
+-   No changes
+
+Contributors to 1.24.3
+
+Alexander Slobodeniuk, Edward Hervey, Elliot Chen, F. Duncanh, François Laignel, Haihua Hu, He Junyan, Hou Qi, Jan Schmidt,
+Jimmy Ohn, Maksym Khomenko, Matthew Waters, Nicolas Dufresne, Nirbheek Chauhan, Philippe Normand, Philipp Zabel, Qian Hu (胡骞),
+Sanchayan Maity, Sebastian Dröge, Seungha Yang, Simonas Kazlauskas, Taruntej Kanakamalla, Tim Blechmann, Tim-Philipp Müller, U.
+Artie Eoff, Víctor Manuel Jáquez Leal, William Wedler, Xavier Claessens,
+
+… and many others who have contributed bug reports, translations, sent suggestions or helped testing. Thank you all!
+
+List of merge requests and issues fixed in 1.24.3
+
+-   List of Merge Requests applied in 1.24.3
+-   List of Issues fixed in 1.24.3
+
+1.24.4
+
+The fourth 1.24 bug-fix release (1.24.4) was released on 29 May 2024.
 
--   Pull ninja from system if possible, avoid spurious bootstrap of
-    cmake
--   ffmpeg: update to 5.0.3
--   libsoup: update to 2.74.3
--   orc: update to 0.4.34
-
-Contributors to 1.22.5
-
-Andoni Morales Alastruey, Bastien Nocera, Carlos Rafael Giani, David
-Craven, Doug Nazar, Edward Hervey, François Laignel, Guillaume
-Desmottes, He Junyan, Hou Qi, Jan Alexander Steffens (heftig), Jan
-Schmidt, Maksym Khomenko, Mathieu Duponchelle, Matthew Waters, Michael
-Olbrich, Michael Tretter, Nicolas Dufresne, Nirbheek Chauhan, Philippe
-Normand, Ruslan Khamidullin, Sebastian Dröge, Seungha Yang, Théo
-Maillart, Thibault Saunier, Tim-Philipp Müller, Víctor Manuel Jáquez
-Leal, Vivia Nikolaidou, Yatin Maan,
-
-… and many others who have contributed bug reports, translations, sent
-suggestions or helped testing. Thank you all!
-
-List of merge requests and issues fixed in 1.22.5
-
--   List of Merge Requests applied in 1.22.5
--   List of Issues fixed in 1.22.5
-
-1.22.6
-
-The sixth 1.22 bug-fix release (1.22.6) was released on 20 September
-2023.
-
-This release only contains bugfixes and security fixes and it should be
-safe to update from 1.22.x.
-
-Highlighted bugfixes in 1.22.6
-
--   Security fixes for the MXF demuxer and H.265 video parser
--   Fix latency regression in H.264 hardware decoder base class
--   androidmedia: fix HEVC codec profile registration and fix coded_data
-    handling
--   decodebin3: fix switching from a raw stream to an encoded stream
--   gst-inspect: prettier and more correct signal and action signals
-    printing
--   rtmp2: Allow NULL flash version, omitting the field, for better RTMP
-    server compatibility
--   rtspsrc: better compatibility with buggy RTSP servers that don’t set
-    a clock-rate
--   rtpjitterbuffer: fix integer overflow that led to more packets being
-    declared lost than have been lost
--   v4l2: fix video encoding regression on RPi and fix support for left
-    and top padding
--   waylandsink: Crop surfaces to their display width height
--   cerbero: recognise Manjaro; add Rust support for MSVC ARM64; cmake
-    detection fixes
--   various bug fixes, build fixes, memory leak fixes, and other
-    stability and reliability improvements
+This release only contains bugfixes and it should be safe to update from 1.24.x.
+
+Highlighted bugfixes in 1.24.4
+
+-   audioconvert: support more than 64 audio channels
+-   avvidec: fix dropped frames when doing multi-threaded decoding of I-frame codecs such as DV Video
+-   mpegtsmux: Correctly time out in live pipelines, esp. for sparse streams like KLV and DVB subtitles
+-   vtdec deadlock fixes on shutdown and format/resolution changes (as might happen with e.g. HLS/DASH)
+-   fmp4mux, isomp4mux: Add support for adding AV1 header OBUs into the MP4 headers, and add language from tags
+-   gtk4paintablesink improvements: fullscreen mode and gst-play-1.0 support
+-   webrtcsink: add support for insecure TLS and improve error handling and VP9 handling
+-   v4l2codecs: decoder: Reorder caps to prefer DMA_DRM ones, fixes issues with playbin3
+-   vah264enc, vah265enc: timestamp handling fixes; generate IDR frames on force-keyunit-requests, not I frames
+-   Visualizer plugins fixes
+-   Avoid using private APIs on iOS
+-   Various bug fixes, memory leak fixes, and other stability and reliability improvements
 
 gstreamer
 
--   gst-inspect: prettier and more correct signal printing, and print
-    action signals in g_signal_emit_by_name() format
--   gst-launch: Disable fault signal handlers on macOS
+-   clock: Fix 32 bit assertions in GST_TIME_TO_TIMEVAL and GST_TIME_TO_TIMESPEC
+-   systemclock: fix usage of __STDC_NO_ATOMICS__
+-   filesrc: Don’t abort on _get_osfhandle() on Windows
+-   tests: Avoid using “bool” for the variable name
+-   Various Solaris / Illumos fixes
+-   parse: Don’t assume that child proxy child objects are GstObjects
 
 gst-plugins-base
 
--   audio: Make sure to stop ringbuffer on error
--   decodebin3: avoid identity, sinkpad, parsebin leakage when reset
-    input
--   decodebin3: Ensure the slot is unlinked before linking to decoder
--   sdp: fix wrong debug log error message for missing clock-rate in
-    caps
--   sdp: Parse zero clock-rate as default
+-   audioconvert: Support converting >64 channels
+-   decodebin3: Fix caps and stream leaks
+-   playbin(3), streamsynchronizer: Fix deadlock when streams have been flushed before others start
+-   videotestsrc: fix race condition when clearing cached buffer
+-   Fix visualization plugins
+-   GstDiscoverer hangs when processing media file containing mebx on MacOS
+-   glmixer: Add GL_SYNC_META option to buffer pool
+-   typefinding: Fix ID_ODD_SIZE handling regression in WavPack typefinder
 
 gst-plugins-good
 
--   adaptivedemux2: fix memory leak
--   pulsedeviceprovider: fix incorrect usage of GST_ELEMENT_ERROR
--   qt: Unbreak build with qt-egl enabled but viv_fb missing
--   qt: Fix searching of qt5/qt6 tools with qmake in Meson
--   qtdemux: Fix premature EOS when some files are played in push mode
--   qtdemux: attach cbcs crypt info at the right moment
--   rtpjitterbuffer: Avoid integer overflow in max saveable packets
-    calculation with negative offset
--   videoflip: fix concurrent access when modifying the tag list
--   v4l2: allocator: Don’t close foreign dmabuf
--   v4l2: bufferpool: Fix large encoded stream regression
--   v4l2: bufferpool: Problems when checking for truncated buffer
--   v4l2: Fix support for left and top padding
--   v4l2object: clear format lists if source change event is received
+-   osxaudio: Avoid using private APIs on iOS
+-   qtdemux: unit test fixes for 32-bit platforms
 
 gst-plugins-bad
 
--   androidmedia/enc: handle codec-data before popping
-    GstVideoCodecFrames
--   androidmedia: fix hevc codec profile registration
--   androidmedia: Small fixes
--   androidmedia: Add more null checks (of env) to JNI utilities
--   applemedia: Fix pixel format for I420 and NV12
--   audiolatency: Forward latency query and event upstream
--   av1parser: Fix segmentation params update
--   codecparsers: Fix MPEG-1 aspect ratio table
--   d3d11convert: Passthrough allocation query on same caps
--   h264decoder: Update latency dynamically
--   h265parser: Allow partially broken hvcC data
--   h265parser: Fix possible overflow using max_sub_layers_minus1
--   hlssink2: Always use forward slash separator
--   mdns: Fix a crash on context error
--   mxfdemux: Fix integer overflow causing out of bounds writes when
-    handling invalid uncompressed video and check channels for AES3
--   nvencoder: Fix negotiation error when interlace-mode is unspecified
--   rtmp2: Allow NULL flash version, omitting the field
--   rtmp2sink: fix crash if message conversion failed
--   transcodebin: Fixes for upstream selectable support
--   va: Fix in error logs functions mismatches
--   waylandsink: Crop surfaces to their display width height
--   waylandsink: Fix cropping for video with non-square aspect ratio
--   webrtc: Fix docs for create-data-channel action signal
--   win32ipc: Fix pipe handle leak
+-   cudamemory: Fix offset of subsampled planar formats
+-   d3d11: Revert “d3d11device: protect device_lock vs device_new
+-   d3dshader: Fix gamma and primaries conversion pixel shader
+-   dtlsconnection: Fix overflow in timeout calculation on systems with 32 bit time_t
+-   GstPlay: Initialize debug category and error quark in class_init
+-   kmssink: Do not close the DRM prime handle twice
+-   mpegtsmux: Correctly time out and mux anyway in live pipelines
+-   nvcodec: Accept progressive-high profiles for h264
+-   nvencoder: Fix maximum QP value setting
+-   qsvh264dec, qsvh265dec: Fix nalu leaks
+-   v4l2codecs: decoder: Reorder caps to prefer DMA_DRM ones
+-   vah264enc, vah265enc: Let FORCE_KEYFRAME be IDR frame rather than just I frame
+-   vah264enc, vah265enc: Set DTS offset before PTS
+-   vkh26xdec: Fix stop memory leak
+-   vtdec: Fix deadlock when negotiating format change
+-   vtdec: Fix PAUSED->READY deadlock when output loop is running
+-   wayland: Use wl_display_create_queue_with_name
+-   webrtc: request-aux-sender, only sink floating refs
+-   webrtcbin: Allow session level setup attribute in SDP
 
 gst-plugins-ugly
 
 -   No changes
 
-gst-plugins-rs
+GStreamer Rust plugins
 
--   fallbackswitch: locking/deadlock fixes
--   onvifmetadataparse: Skip metadata frames with unrepresentable UTC
-    time
--   transcriberbin: Configure audioresample in front of transcriber
--   webrtcsink: Propagate GstContext messages
--   webrtcsink: Add support for d3d11 memory and qsvh264enc
--   webrtcsink: fix TWCC extension adding
--   webrtcsink: don’t forget to setup encoders for discoveries
--   webrtcsink: NVIDIA V4L2 encoders always require NVMM memory
--   meson: Fix handling of optional deps, and don’t require Python 3.8
+-   fmp4mux, isomp4mux: Add support for adding AV1 header OBUs into the MP4 headers
+-   fmp4mux, isomp4mux: Add language from tags
+-   gtk4paintablesink: Also create own window for gst-play-1.0
+-   gtk4paintablesink: Add GST_GTK4_WINDOW_FULLSCREEN environment variable to fullscreen window for testing with
+    e.g. gst-launch-1.0
+-   gtk4: Fix plugin description and update python examples
+-   rtpgccbwe: Log effective bitrate in more places, and log delay and loss target bitrates separately
+-   webrtcsink: Improve error when no encoder was found or RTP plugins were missing
+-   webrtcsink: Add VP9 parser after the encoder for VP9 too
+-   webrtc: Add support for insecure TLS connections
 
 gst-libav
 
--   No changes
+-   avvidec: Fix dropping wrong “ghost” frames - fixing multi-threaded decoding of I-frame codecs such as DV Video
 
 gst-rtsp-server
 
@@ -2325,11 +1952,11 @@
 
 gst-editing-services
 
--   No changes
+-   ges-pipeline: Configure encodebin before linking
 
-gst-validate + gst-integration-testsuites
+gst-devtools, gst-validate + gst-integration-testsuites
 
--   gst-validate: Disable fault signal handlers on macOS
+-   No changes
 
 gst-examples
 
@@ -2337,179 +1964,113 @@
 
 Development build environment
 
--   macos-bison: Update to 3.8.2 and add an ARM64 build
--   wrap: update libpsl to 0.21.2
+-   No changes
+
+Cerbero build tool and packaging changes in 1.24.4
+
+-   Add Fedora 40 support
+-   srt: bump version to 1.5.3
+-   pango: Fix leaks on Windows (textoverlay plugins)
+-   windows runtime installer has empty qt6 msm
+-   WiX: assorted fixes
+
+Contributors to 1.24.4
+
+Alexander Slobodeniuk, Bill Nottingham, Brad Reitmeyer, Chris Del Guercio, Daniel Stone, Edward Hervey, Emil Pettersson, He
+Junyan, Jordan Petridis, Joshua Breeden, L. E. Segovia, Martin Nordholts, Mathieu Duponchelle, Matthew Waters, Nirbheek Chauhan,
+Philippe Normand, Piotr Brzeziński, Rafael Carício, Robert Ayrapetyan, Robert Mader, Sebastian Dröge, Seungha Yang, Shengqi Yu,
+Stéphane Cerveau, Tim-Philipp Müller,
 
-Cerbero build tool and packaging changes in 1.22.6
+… and many others who have contributed bug reports, translations, sent suggestions or helped testing. Thank you all!
 
--   Add Rust support for MSVC ARM64
--   Recognise PERL5LIB as a joinable Unix variable
--   Recognise Manjaro as an Arch derivative
--   Fix picking up cmake from build-tools
-
-Contributors to 1.22.6
-
-Akihiro Sagawa, Alicia Boya García, Guillaume Desmottes, Haihua Hu,
-Hugues Fruchet, Ivan Molodetskikh, Jan Alexander Steffens (heftig), Jan
-Schmidt, L. E. Segovia, Mathieu Duponchelle, Matthew Waters, Ming Qian,
-Nicolas Dufresne, Nirbheek Chauhan, Olivier Blin, Olivier Crête,
-Philippe Normand, Piotr Brzeziński, Robert Ayrapetyan, Ryan Pavlik,
-Sebastian Dröge, Seungha Yang, Stéphane Cerveau, Stephan Seitz, Thomas
-Schneider, Tim-Philipp Müller, Víctor Manuel Jáquez Leal, Wang Chuan,
-Xabier Rodriguez Calvar,
-
-… and many others who have contributed bug reports, translations, sent
-suggestions or helped testing. Thank you all!
-
-List of merge requests and issues fixed in 1.22.6
-
--   List of Merge Requests applied in 1.22.6
--   List of Issues fixed in 1.22.6
-
-1.22.7
-
-The seventh 1.22 bug-fix release (1.22.7) was released on 13 November
-2023.
-
-This release only contains bugfixes and security fixes and it should be
-safe to update from 1.22.x.
-
-Highlighted bugfixes in 1.22.7
-
--   Security fixes for the MXF demuxer and AV1 codec parser
--   glfilter: Memory leak fix for OpenGL filter elements
--   d3d11videosink: Fix toggling between fullscreen and maximized, and
-    window switching in fullscreen mode
--   DASH / HLS adaptive streaming fixes
--   Decklink card device provider device name string handling fixes
--   interaudiosrc: handle non-interleaved audio properly
--   openh264: Fail gracefully if openh264 encoder/decoder creation fails
--   rtspsrc: improved whitespace handling in response headers by certain
-    cameras
--   v4l2codecs: avoid wrap-around after 1000000 frames; tiled formats
-    handling fixes
--   video-scaler, audio-resampler: downgraded “Can’t find exact taps”
-    debug log messages
--   wasapi2: Don’t use global volume control object
--   Rust plugins: various improvements in aws, fmp4mux, hlssink3,
-    livesync, ndisrc, rtpav1depay, rsfilesink, s3sink, sccparse
--   WebRTC: various webrtchttp, webrtcsrc, and webrtcsink improvements
-    and fixes
--   Cerbero build tools: recognise Windows 11; restrict parallelism of
-    gst-plugins-rs build on small systems
--   Packages: ca-certificates update; fix gio module loading and TLS
-    support on macOS
+List of merge requests and issues fixed in 1.24.4
+
+-   List of Merge Requests applied in 1.24.4
+-   List of Issues fixed in 1.24.4
+
+1.24.5
+
+The fifth 1.24 bug-fix release (1.24.5) was released on 20 June 2024.
+
+This release only contains bugfixes and it should be safe to update from 1.24.x.
+
+Highlighted bugfixes in 1.24.5
+
+-   webrtcsink: Support for AV1 via nvav1enc, av1enc or rav1enc encoders
+-   AV1 RTP payloader/depayloader fixes to work correctly with Chrome and Pion WebRTC
+-   av1parse, av1dec error handling/robustness improvements
+-   av1enc: Handle force-keyunit events properly for WebRTC
+-   decodebin3: selection and collection handling improvements
+-   hlsdemux2: Various fixes for discontinuities, variant switching, playlist updates
+-   qml6glsink: fix RGB format support
+-   rtspsrc: more control URL handling fixes
+-   v4l2src: Interpret V4L2 report of sync loss as video signal loss
+-   d3d12 encoder, memory and videosink fixes
+-   vtdec: more robust error handling, fix regression
+-   ndi: support for NDI SDK v6
+-   Various bug fixes, memory leak fixes, and other stability and reliability improvements
 
 gstreamer
 
--   debugutils: provide gst_debug_bin_to_dot_data() implementation even
-    if debug system is disabled
+-   promise: Don’t use g_return_* for internal checks
+-   debug: Add missing gst_debug_log_id_literal() dummy with gst_debug=false
+-   ptp-helper: Add GNU/Hurd support
 
 gst-plugins-base
 
--   audioaggregator, audiomixer: Make access to the pad list thread-safe
-    while mixing
--   basetextoverlay: Fix overlay never rendering again if width reaches
-    1px
--   glfiter: Protect GstGLContext access
--   glfilter: Only add parent meta if inbuf != outbuf
--   macOS: fix huge memory leak with glfilter-based elements
--   rtspconnection: Ignore trailing whitespace in rtsp headers
--   video-scaler, audio-resampler: downgrade ‘can’t find exact taps’ to
-    debug
+-   uridecodebin3: Don’t hold PLAY_ITEMS lock when activating them
+-   decodebin3: Always ensure we end up with parsebin or identity
+-   decodebin3: Properly support changing input collections
+-   decodebin3: Avoid usage of parsebin even more
+-   decodebin3: Fix dealing with input without caps
+-   video-info: Don’t crash in gst_video_info_is_equal() if one videoinfo is zero-initialized
 
 gst-plugins-good
 
--   adaptivedemux2: Do not submit_transfer when cancelled
--   adaptivedemux2: Call GTasks’s return functions for blocking tasks
--   flacenc: Correctly handle up to 255 cue entries
--   flvmux: set the src segment position as running time
--   imagesequencesrc: fix deadlock when feeding the same image in a loop
--   pngenc: output one frame only in snapshot mode and mark output
-    frames as I-frames
--   qmlglsrc: sync on the streaming thread
--   souphttpsrc: Chain up to finalize to fix memory leak
--   wavparse: fix buffer leak with adtl tag
--   v4l2codecs: Avoid QBUF/DQBUF struct timeval .tv_usec wrap-around at
-    frame 1000000
--   v4l2codecs: Fix tiled formats stride conversion
+-   flacparse: fix buffer overflow
+-   hlsdemux2: Various fixes for discontinuities, variant switching, playlist updates
+-   qml6glsink: fix RGB format support
+-   rtpdtmfdepay: fix caps negotiation with audioconvert
+-   rtpdtmfsrc: fix leak when shutting down mid-event
+-   rtspsrc: fix invalid seqnum assertions
+-   rtspsrc: Various control URL handling fixes
+-   v4l2src: Interpret V4L2 report of sync loss as video signal loss
 
 gst-plugins-bad
 
--   audiobuffersplit: disable max-silence-time if set to 0
--   libde265: Do not decode the non 4:2:0 8 bits format
--   codecparsers: av1: Clip max tile rows and cols values
--   codecs: h265: Do not free slice header before using it
--   d3d11converter: Fix 10/12bits planar output
--   d3d11decoder: Fix crash on negotiate() when decoder is not
-    configured
--   d3d11videosink: Fix toggling between fullscreen and maximized
--   d3d11videosink: Fix window switching in case of fullscreen mode
--   d3d11screencapturesrc: Fix mouse cursor blending
--   decklink: Fix broken COM string conversion
--   decklink: Decklink Device Provider wrongly parses SDK strings
--   gstwayland: Don’t depend on wayland-protocols
--   interaudiosrc: Add audio meta to buffers containing non-interleaved
-    samples
--   kmssink: Add TIDSS auto-detection
--   mfvideoencoder: Fix typo in template caps
--   mxfdemux: Store GstMXFDemuxEssenceTrack in their own fixed
-    allocation
--   nvcodec: fix bounds for auto-select GPU enumeration
--   openh264: Fail gracefully if openh264 encoder/decoder creation fails
--   tsmux: More cleanups
--   tsmux: Fill padding packets with stuffing bytes
--   v4l2codecs: Fix tiled formats stride conversion
--   v4l2videodec: Correctly free caps to avoid memory leak
--   wasapi2: Don’t use global volume control object
--   wasapi2device: Ignore activation failed device
+-   av1parse: Do not return error when expectedFrameId mismatch
+-   av1dec: Don’t treat decoding errors as fatal and print more error details
+-   av1enc: Handle force-keyunit events properly by requesting keyframes
+-   codectimestamper: never set DTS to NONE
+-   d3d12encoder: Do not print error log for not-supported feature
+-   d3d12memory: Fix staging buffer alignment
+-   d3d12videosink: Disconnect window signal handler on dispose as intended
+-   dtlssrtpenc: Don’t crash if no pad name is provided when requesting a new pad
+-   glcolorconvert: update existing sync meta if outbuf has one
+-   pcapparse: Parsing code assumes unaligned memory accesses are OK
+-   pcapparse: Avoid unaligned memory access
+-   tsdemux: Fix maximum PCR/DTS values
+-   vtdec: Improve error handling in edge cases
+-   vtdec, qtdemux: regression in 1.24.3 - internal data stream error
+-   uvcgadget: Use g_path_get_basename instead of libc basename
 
 gst-plugins-ugly
 
 -   No changes
 
-gst-plugins-rs
+GStreamer Rust plugins
 
--   aws: s3sink: Fix handling of special characters in key
--   aws: s3: Properly percent-decode GstS3Url
--   fmp4mux: Don’t overflow negative composition offset calculation
--   fmp4mux: specify the fragment duration unit
--   hlssink3: Use Path API for getting file name
--   hlssink3: Use sprintf for segment name formatting
--   hlssink3: Remove unused deps
--   hlssink3: Don’t remove old files if max-files is zero
--   hlssink3: Don’t remove uri from playlist if playlist-length is zero
--   hlssink3: Various cleanup
--   livesync: log new pending segments
--   livesync: display jitter when waiting on clock
--   livesync: Rename activatemode methods to *_activatemode
--   livesync: Simplify start_src_task and src_loop
--   livesync: Improve audio duration fixups
--   livesync: Log a category error when we are missing the segment
--   livesync: Clean up state handling
--   livesync: Replace an if-let with match
--   livesync: Move a notify closer to the interesting state change
--   livesync: Move num_in counting to the src task
--   livesync: Simplify num_duplicate counting
--   livesync: Handle flags and late buffer patching after queueing
--   livesync: Separate out_buffer duplicate status from GAP flag
--   livesync: Use fallback_duration for audio repeat buffers as well
--   livesync: example: Add identities single-segment=1
--   livesync: Split fallback_duration into in_ and out_duration
--   livesync: Keep existing buffer duration in some cases
--   livesync: Remove the stop from outgoing segments
--   ndisrc: Assume input with more than 8 raw audio channels is
-    unpositioned
--   rtpav1depay: Skip unexpected leading fragments
--   rtpav1depay: Don’t push stale temporal delimiters downstream
--   rsfilesink: set sync=false
--   s3sink: set sync=false
--   sccparse: Fix leading spaces between the tab and caption data
--   webrtchttp: Respect HTTP redirects
--   webrtcsrc: use @watch instead of @to-owned
--   webrtcsrc: add turn-servers property
--   webrtc: Fix paths in README
--   webrtcsink: don’t miss ice candidates
+-   gtk4: update flatpak integration code
+-   ndi: Add support for loading NDI SDK v6
+-   reqwesthttpsrc: Fix race condition when unlocking
+-   rtp: Don’t restrict payload types for payloaders
+-   rtp: av1pay: Correctly use N flag for marking keyframes
+-   rtp: av1depay: Parse internal size fields of OBUs and handle them
+-   webrtcsink: Refactor value retrieval to avoid lock poisoning
+-   webrtcsink: Add support for AV1
+-   webrtc: Update to async-tungstenite 0.26
+-   Fix various new clippy 1.79 warnings
+-   meson: Fix various issues in dependency management, feature detection, some regressions, and add logging
 
 gst-libav
 
@@ -2517,7 +2078,7 @@
 
 gst-rtsp-server
 
--   rtspclientsink: Don’t leak previous server_ip
+-   No changes
 
 gstreamer-vaapi
 
@@ -2539,9 +2100,9 @@
 
 -   No changes
 
-gst-validate + gst-integration-testsuites
+gst-devtools, gst-validate + gst-integration-testsuites
 
--   gst-validate: Fix compatibility with Python 3.12
+-   No changes
 
 gst-examples
 
@@ -2551,54 +2112,36 @@
 
 -   No changes
 
-Cerbero build tool and packaging changes in 1.22.7
+Cerbero build tool and packaging changes in 1.24.5
+
+-   No changes
+
+Contributors to 1.24.5
+
+Angelo Verlain, Chris Del Guercio, Corentin Damman, Edward Hervey, Francisco Javier Velázquez-García, He Junyan, Jakub Adam,
+Jakub Vaněk, Khem Raj, Martin Nordholts, Mathieu Duponchelle, Nirbheek Chauhan, Piotr Brzeziński, Samuel Thibault, Sebastian
+Dröge, Sergey Krivohatskiy, Seungha Yang, Tim-Philipp Müller, Zach van Rijn,
+
+… and many others who have contributed bug reports, translations, sent suggestions or helped testing. Thank you all!
+
+List of merge requests and issues fixed in 1.24.5
+
+-   List of Merge Requests applied in 1.24.5
+-   List of Issues fixed in 1.24.5
+
+Schedule for 1.26
+
+Our next major feature release will be 1.26, and 1.25 will be the unstable development version leading up to the stable 1.26
+release. The development of 1.25/1.25 will happen in the git main branch of the GStreamer mono repository.
+
+The schedule for 1.26 is yet to be decided.
+
+1.26 will be backwards-compatible to the stable 1.24, 1.22, 1.20, 1.18, 1.16, 1.14, 1.12, 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0
+release series.
+
+--------------------------------------------------------------------------------------------------------------------------------
 
--   Add Windows 11 to the supported versions list
--   ca-certificates: Update to version from 2023-08-22
--   cargo: Restrict parallelism if a small system is detected (for
-    gst-plugins-rs build)
--   Fix venv setup on Python 3.11+
--   Fix unlinking of Android NDK directories if install fails midway
--   glib: Work around AppleClang + -Werror test build failure
--   glib: Re-add gio module loading patch for macOS, remove unused patch
-    files
-
-Contributors to 1.22.7
-
-Albert Sjölund, Arun Raghavan, Balló György, Benjamin Gaignard, Detlev
-Casanova, Doug Nazar, Eric, Florian Zwoch, François Laignel, Guillaume
-Desmottes, He Junyan, Hou Qi, James Oliver, Jan Alexander Steffens
-(heftig), Jan Schmidt, Johan Adam Nilsson, Jordan Yelloz, Kalev Lember,
-L. E. Segovia, Lieven Paulissen, Maksym Khomenko, Marek Vasut, Matthias
-Fuchs, Michiel Westerbeek, Nicolas Dufresne, Philippe Normand, Piotr
-Brzeziński, Rahul T R, Sean DuBois, Sebastian Dröge, Seungha Yang,
-Stéphane Cerveau, Thibault Saunier, Tim-Philipp Müller,
-
-… and many others who have contributed bug reports, translations, sent
-suggestions or helped testing. Thank you all!
-
-List of merge requests and issues fixed in 1.22.7
-
--   List of Merge Requests applied in 1.22.7
--   List of Issues fixed in 1.22.7
-
-Schedule for 1.24
-
-Our next major feature release will be 1.24, and 1.23 will be the
-unstable development version leading up to the stable 1.24 release. The
-development of 1.23/1.24 will happen in the git main branch of the
-GStreamer mono repository.
-
-The plan for the 1.24 development cycle is yet to be confirmed.
-
-1.24 will be backwards-compatible to the stable 1.22, 1.20, 1.18, 1.16,
-1.14, 1.12, 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
-
-------------------------------------------------------------------------
-
-These release notes have been prepared by Tim-Philipp Müller with
-contributions from Edward Hervey, Matthew Waters, Nicolas Dufresne,
-Nirbheek Chauhan, Olivier Crête, Sebastian Dröge, Seungha Yang, and
-Thibault Saunier.
+These release notes have been prepared by Tim-Philipp Müller with contributions from Edward Hervey, Matthew Waters, Nicolas
+Dufresne, Nirbheek Chauhan, Olivier Crête, Sebastian Dröge, Seungha Yang, Thibault Saunier, and Víctor Manuel Jáquez Leal.
 
 License: CC BY-SA 4.0

gst-plugins-ugly-1.22.7.tar.xz/README.md -> gst-plugins-ugly-1.24.5.tar.xz/README.md Changed

gst-plugins-ugly-1.22.7.tar.xz/RELEASE -> gst-plugins-ugly-1.24.5.tar.xz/RELEASE Changed

@@ -1,17 +1,17 @@
-This is GStreamer gst-plugins-ugly 1.22.7.
+This is GStreamer gst-plugins-ugly 1.24.5.
 
 The GStreamer team is thrilled to announce a new major feature release
 of your favourite cross-platform multimedia framework!
 
 As always, this release is again packed with new features, bug fixes and
 other improvements.
-
-The 1.22 release series adds new features on top of the 1.20 series and is
+ 
+The 1.24 release series adds new features on top of the 1.22 series and is
 part of the API and ABI-stable 1.x release series.
 
 Full release notes can be found at:
 
-  https://gstreamer.freedesktop.org/releases/1.22/
+  https://gstreamer.freedesktop.org/releases/1.24/
 
 Binaries for Android, iOS, Mac OS X and Windows will usually be provided
 shortly after the release.
@@ -44,10 +44,7 @@
 
  - gstreamer-vaapi: hardware-accelerated video decoding and encoding using
                     VA-API on Linux. Primarily for Intel graphics hardware.
-
- - gst-omx: hardware-accelerated video decoding and encoding, primarily for
-                    embedded Linux systems that provide an OpenMax
-                    implementation layer such as the Raspberry Pi.
+                    (Deprecated, use the new "va" plugin instead)
 
  - gst-rtsp-server: library to serve files or streaming pipelines via RTSP
 
@@ -80,14 +77,18 @@
 
 for more details.
 
-For help and support, please subscribe to and send questions to the
-gstreamer-devel mailing list (see below for details).
+For help and support, please head over to our Discourse forum at
 
-There is also a #gstreamer IRC channel on the OFTC IRC network, which is
-also bridged into the Matrix network.
+  https://discourse.gstreamer.org/
 
-Please do not submit support requests in GitLab, we only use it
-for bug tracking and merge requests review.
+or pop into one of our Matrix chat rooms, see
+
+  https://discourse.gstreamer.org/t/new-gstreamer-matrix-chat-space/675
+
+for more details.
+
+Please do not submit support requests in GitLab, we only use it for
+bug tracking and merge requests review. Use the Discourse forum instead.
 
 ==== Developers ====
 
@@ -99,6 +100,9 @@
 Merge Requests or file issues for bugs or feature requests.
 
 Interested developers of the core library, plugins, and applications should
-subscribe to the gstreamer-devel list:
+join us on Matrix for chat and the Discourse forum for announcements, help
+and discussions.
+
+There is also a gstreamer-devel mailing list, but Discourse is preferred:
 
   https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

gst-plugins-ugly-1.22.7.tar.xz/REQUIREMENTS -> gst-plugins-ugly-1.24.5.tar.xz/REQUIREMENTS Changed

gst-plugins-ugly-1.22.7.tar.xz/docs/gst_plugins_cache.json -> gst-plugins-ugly-1.24.5.tar.xz/docs/gst_plugins_cache.json Changed

@@ -122,209 +122,6 @@
         "tracers": {},
         "url": "Unknown package origin"
     },
-    "amrnb": {
-        "description": "Adaptive Multi-Rate Narrow-Band",
-        "elements": {
-            "amrnbdec": {
-                "author": "GStreamer maintainers <gstreamer-devel@lists.freedesktop.org>",
-                "description": "Adaptive Multi-Rate Narrow-Band audio decoder",
-                "hierarchy": 
-                    "GstAmrnbDec",
-                    "GstAudioDecoder",
-                    "GstElement",
-                    "GstObject",
-                    "GInitiallyUnowned",
-                    "GObject"
-                ,
-                "klass": "Codec/Decoder/Audio",
-                "long-name": "AMR-NB audio decoder",
-                "pad-templates": {
-                    "sink": {
-                        "caps": "audio/AMR:\n           rate: 8000\n       channels: 1\n",
-                        "direction": "sink",
-                        "presence": "always"
-                    },
-                    "src": {
-                        "caps": "audio/x-raw:\n         format: S16LE\n         layout: interleaved\n           rate: 8000\n       channels: 1\n",
-                        "direction": "src",
-                        "presence": "always"
-                    }
-                },
-                "properties": {
-                    "variant": {
-                        "blurb": "The decoder variant",
-                        "conditionally-available": false,
-                        "construct": true,
-                        "construct-only": false,
-                        "controllable": false,
-                        "default": "IF1 (0)",
-                        "mutable": "null",
-                        "readable": true,
-                        "type": "GstAmrnbVariant",
-                        "writable": true
-                    }
-                },
-                "rank": "primary"
-            },
-            "amrnbenc": {
-                "author": "Wim Taymans <wim.taymans@gmail.com>",
-                "description": "Adaptive Multi-Rate Narrow-Band audio encoder",
-                "hierarchy": 
-                    "GstAmrnbEnc",
-                    "GstAudioEncoder",
-                    "GstElement",
-                    "GstObject",
-                    "GInitiallyUnowned",
-                    "GObject"
-                ,
-                "interfaces": 
-                    "GstPreset"
-                ,
-                "klass": "Codec/Encoder/Audio",
-                "long-name": "AMR-NB audio encoder",
-                "pad-templates": {
-                    "sink": {
-                        "caps": "audio/x-raw:\n         format: S16LE\n         layout: interleaved\n           rate: 8000\n       channels: 1\n",
-                        "direction": "sink",
-                        "presence": "always"
-                    },
-                    "src": {
-                        "caps": "audio/AMR:\n           rate: 8000\n       channels: 1\n",
-                        "direction": "src",
-                        "presence": "always"
-                    }
-                },
-                "properties": {
-                    "band-mode": {
-                        "blurb": "Encoding Band Mode (Kbps)",
-                        "conditionally-available": false,
-                        "construct": true,
-                        "construct-only": false,
-                        "controllable": false,
-                        "default": "MR122 (7)",
-                        "mutable": "null",
-                        "readable": true,
-                        "type": "GstAmrnbEncBandMode",
-                        "writable": true
-                    }
-                },
-                "rank": "secondary"
-            }
-        },
-        "filename": "gstamrnb",
-        "license": "unknown",
-        "other-types": {
-            "GstAmrnbEncBandMode": {
-                "kind": "enum",
-                "values": 
-                    {
-                        "desc": "MR475",
-                        "name": "MR475",
-                        "value": "0"
-                    },
-                    {
-                        "desc": "MR515",
-                        "name": "MR515",
-                        "value": "1"
-                    },
-                    {
-                        "desc": "MR59",
-                        "name": "MR59",
-                        "value": "2"
-                    },
-                    {
-                        "desc": "MR67",
-                        "name": "MR67",
-                        "value": "3"
-                    },
-                    {
-                        "desc": "MR74",
-                        "name": "MR74",
-                        "value": "4"
-                    },
-                    {
-                        "desc": "MR795",
-                        "name": "MR795",
-                        "value": "5"
-                    },
-                    {
-                        "desc": "MR102",
-                        "name": "MR102",
-                        "value": "6"
-                    },
-                    {
-                        "desc": "MR122",
-                        "name": "MR122",
-                        "value": "7"
-                    },
-                    {
-                        "desc": "MRDTX",
-                        "name": "MRDTX",
-                        "value": "8"
-                    }
-                
-            },
-            "GstAmrnbVariant": {
-                "kind": "enum",
-                "values": 
-                    {
-                        "desc": "IF1",
-                        "name": "IF1",
-                        "value": "0"
-                    },
-                    {
-                        "desc": "IF2",
-                        "name": "IF2",
-                        "value": "1"
-                    }
-                
-            }
-        },
-        "package": "GStreamer Ugly Plug-ins",
-        "source": "gst-plugins-ugly",
-        "tracers": {},
-        "url": "Unknown package origin"
-    },
-    "amrwbdec": {
-        "description": "Adaptive Multi-Rate Wide-Band Decoder",
-        "elements": {
-            "amrwbdec": {
-                "author": "Renato Araujo <renato.filho@indt.org.br>",
-                "description": "Adaptive Multi-Rate Wideband audio decoder",
-                "hierarchy": 
-                    "GstAmrwbDec",
-                    "GstAudioDecoder",
-                    "GstElement",
-                    "GstObject",
-                    "GInitiallyUnowned",
-                    "GObject"
-                ,
-                "klass": "Codec/Decoder/Audio",
-                "long-name": "AMR-WB audio decoder",
-                "pad-templates": {
-                    "sink": {
-                        "caps": "audio/AMR-WB:\n           rate: 16000\n       channels: 1\n",
-                        "direction": "sink",
-                        "presence": "always"
-                    },
-                    "src": {
-                        "caps": "audio/x-raw:\n         format: S16LE\n         layout: interleaved\n           rate: 16000\n       channels: 1\n",
-                        "direction": "src",
-                        "presence": "always"
-                    }
-                },
-                "properties": {},
-                "rank": "primary"
-            }
-        },
-        "filename": "gstamrwbdec",
-        "license": "unknown",
-        "other-types": {},
-        "package": "GStreamer Ugly Plug-ins",
-        "source": "gst-plugins-ugly",
-        "tracers": {},
-        "url": "Unknown package origin"
-    },
     "asf": {
         "description": "Demuxes and muxes audio and video in Microsofts ASF format",
         "elements": {
@@ -710,46 +507,6 @@
     "realmedia": {
         "description": "RealMedia support plugins",
         "elements": {
-            "pnmsrc": {
-                "author": "Wim Taymans <wim.taymans@gmail.com>",
-                "description": "Receive data over the network via PNM",
-                "hierarchy": 
-                    "GstPNMSrc",
-                    "GstPushSrc",
-                    "GstBaseSrc",
-                    "GstElement",
-                    "GstObject",
-                    "GInitiallyUnowned",
-                    "GObject"
-                ,
-                "interfaces": 
-                    "GstURIHandler"
-                ,
-                "klass": "Source/Network",
-                "long-name": "PNM packet receiver",
-                "pad-templates": {
-                    "src": {
-                        "caps": "application/vnd.rn-realmedia:\n",
-                        "direction": "src",
-                        "presence": "always"
-                    }
-                },
-                "properties": {
-                    "location": {
-                        "blurb": "Location of the PNM url to read",
-                        "conditionally-available": false,
-                        "construct": false,
-                        "construct-only": false,
-                        "controllable": false,
-                        "default": "NULL",
-                        "mutable": "null",
-                        "readable": true,
-                        "type": "gchararray",
-                        "writable": true
-                    }
-                },
-                "rank": "marginal"
-            },
             "rademux": {
                 "author": "Tim-Philipp Müller <tim centricular net>",
                 "description": "Demultiplex a RealAudio file",
@@ -761,7 +518,6 @@
                     "GObject"
                 ,
                 "klass": "Codec/Demuxer",
-                "long-name": "RealAudio Demuxer",
                 "pad-templates": {
                     "sink": {
                         "caps": "application/x-pn-realaudio:\n",
@@ -774,163 +530,7 @@
                         "presence": "sometimes"
                     }
                 },
-                "rank": "secondary",
-                "signals": {}
-            },
-            "rdtdepay": {
-                "author": "Lutz Mueller <lutz at topfrose dot de>, Wim Taymans <wim@fluendo.com>",
-                "description": "Extracts RealMedia from RDT packets",
-                "hierarchy": 
-                    "GstRDTDepay",
-                    "GstElement",
-                    "GstObject",
-                    "GInitiallyUnowned",
-                    "GObject"
-                ,
-                "klass": "Codec/Depayloader/Network",
-                "long-name": "RDT packet parser",
-                "pad-templates": {
-                    "sink": {
-                        "caps": "application/x-rdt:\n          media: application\n     clock-rate:  1, 2147483647 \n  encoding-name: X-REAL-RDT\n",
-                        "direction": "sink",
-                        "presence": "always"
-                    },
-                    "src": {
-                        "caps": "application/vnd.rn-realmedia:\n",
-                        "direction": "src",
-                        "presence": "always"
-                    }
-                },
-                "rank": "marginal"
-            },
-            "rdtmanager": {
-                "author": "Wim Taymans <wim.taymans@gmail.com>",
-                "description": "Accepts raw RTP and RTCP packets and sends them forward",
-                "hierarchy": 
-                    "GstRDTManager",
-                    "GstElement",
-                    "GstObject",
-                    "GInitiallyUnowned",
-                    "GObject"
-                ,
-                "klass": "Codec/Parser/Network",
-                "long-name": "RTP Decoder",
-                "pad-templates": {
-                    "recv_rtcp_sink_%%u": {
-                        "caps": "application/x-rtcp:\n",
-                        "direction": "sink",
-                        "presence": "request"
-                    },
-                    "recv_rtp_sink_%%u": {
-                        "caps": "application/x-rdt:\n",
-                        "direction": "sink",
-                        "presence": "request"
-                    },
-                    "recv_rtp_src_%%u_%%u_%%u": {
-                        "caps": "application/x-rdt:\n",
-                        "direction": "src",
-                        "presence": "sometimes"
-                    },
-                    "rtcp_src_%%u": {
-                        "caps": "application/x-rtcp:\n",
-                        "direction": "src",
-                        "presence": "request"
-                    }
-                },
-                "properties": {
-                    "latency": {
-                        "blurb": "Amount of ms to buffer",
-                        "conditionally-available": false,
-                        "construct": false,
-                        "construct-only": false,
-                        "controllable": false,
-                        "default": "200",
-                        "max": "-1",
-                        "min": "0",
-                        "mutable": "null",
-                        "readable": true,
-                        "type": "guint",
-                        "writable": true
-                    }
-                },
-                "rank": "none",
-                "signals": {
-                    "clear-pt-map": {
-                        "args": ,
-                        "return-type": "void",
-                        "when": "last"
-                    },
-                    "on-bye-ssrc": {
-                        "args": 
-                            {
-                                "name": "arg0",
-                                "type": "guint"
-                            },
-                            {
-                                "name": "arg1",
-                                "type": "guint"
-                            }
-                        ,
-                        "return-type": "void",
-                        "when": "last"
-                    },
-                    "on-bye-timeout": {
-                        "args": 
-                            {
-                                "name": "arg0",
-                                "type": "guint"
-                            },
-                            {
-                                "name": "arg1",
-                                "type": "guint"
-                            }
-                        ,
-                        "return-type": "void",
-                        "when": "last"
-                    },
-                    "on-npt-stop": {
-                        "args": 
-                            {
-                                "name": "arg0",
-                                "type": "guint"
-                            },
-                            {
-                                "name": "arg1",
-                                "type": "guint"
-                            }
-                        ,
-                        "return-type": "void",
-                        "when": "last"
-                    },
-                    "on-timeout": {
-                        "args": 
-                            {
-                                "name": "arg0",
-                                "type": "guint"
-                            },
-                            {
-                                "name": "arg1",
-                                "type": "guint"
-                            }
-                        ,
-                        "return-type": "void",
-                        "when": "last"
-                    },
-                    "request-pt-map": {
-                        "args": 
-                            {
-                                "name": "arg0",
-                                "type": "guint"
-                            },
-                            {
-                                "name": "arg1",
-                                "type": "guint"
-                            }
-                        ,
-                        "return-type": "GstCaps",
-                        "when": "last"
-                    }
-                }
+                "rank": "secondary"
             },
             "rmdemux": {
                 "author": "David Schleef <ds@schleef.org>",
@@ -943,7 +543,6 @@
                     "GObject"
                 ,
                 "klass": "Codec/Demuxer",
-                "long-name": "RealMedia Demuxer",
                 "pad-templates": {
                     "audio_%%u": {
                         "caps": "ANY",
@@ -961,25 +560,7 @@
                         "presence": "sometimes"
                     }
                 },
-                "rank": "primary",
-                "signals": {}
-            },
-            "rtspreal": {
-                "author": "Wim Taymans <wim.taymans@gmail.com>",
-                "description": "Extends RTSP so that it can handle RealMedia setup",
-                "hierarchy": 
-                    "GstRTSPReal",
-                    "GstElement",
-                    "GstObject",
-                    "GInitiallyUnowned",
-                    "GObject"
-                ,
-                "interfaces": 
-                    "GstRTSPExtension"
-                ,
-                "klass": "Network/Extension/Protocol",
-                "long-name": "RealMedia RTSP Extension",
-                "rank": "marginal"
+                "rank": "primary"
             }
         },
         "filename": "gstrealmedia",
@@ -1198,7 +779,7 @@
                 "long-name": "x264 H.264 Encoder",
                 "pad-templates": {
                     "sink": {
-                        "caps": "video/x-raw:\n      framerate:  0/1, 2147483647/1 \n          width:  1, 2147483647 \n         height:  1, 2147483647 \n         format: { Y444, Y42B, I420, YV12, NV12, Y444_10LE, I422_10LE, I420_10LE }\n",
+                        "caps": "video/x-raw:\n      framerate:  0/1, 2147483647/1 \n          width:  1, 2147483647 \n         height:  1, 2147483647 \n         format: { Y444, Y42B, I420, YV12, NV12, GRAY8, Y444_10LE, I422_10LE, I420_10LE }\n",
                         "direction": "sink",
                         "presence": "always"
                     },

gst-plugins-ugly-1.22.7.tar.xz/docs/meson.build -> gst-plugins-ugly-1.24.5.tar.xz/docs/meson.build Changed

@@ -95,23 +95,25 @@
         print(':'.join(json.load(f).keys()), end='')
 '''.format(plugins_cache),
   check: true)
-foreach plugin_name: list_plugin_res.stdout().split(':')
-    plugins_doc += hotdoc.generate_doc(plugin_name,
-        project_version: api_version,
-        sitemap: 'sitemap.txt',
-        index: 'index.md',
-        gst_index: 'index.md',
-        gst_smart_index: true,
-        gst_c_sources: 
-            join_paths(root_rel, 'sys/*/*.ch'),
-            join_paths(root_rel, 'ext/*/*.ch'),
-            join_paths(root_rel, 'gst/*/*.ch'),
-        ,
-        gst_c_source_filters: excludes,
-        dependencies: gst_dep,
-        gst_order_generated_subpages: true,
-        disable_incremental_build: true,
-        gst_cache_file: plugins_cache,
-        gst_plugin_name: plugin_name,
-    )
-endforeach
+cdir = meson.current_source_dir()
+if host_machine.system() == 'windows'
+  pathsep = ';'
+else
+  pathsep = ':'
+endif
+gst_plugins_doc = run_command(
+    plugins_cache_generator,
+    'hotdoc-config',
+    '--builddir', meson.current_build_dir(),
+    '--project_version', api_version,
+    '--sitemap', cdir / 'sitemap.txt',
+    '--index', cdir / 'index.md',
+    '--gst_index', cdir / 'index.md',
+    '--gst_c_sources',
+        root_rel / 'sys/*/*.ch',
+        root_rel / 'ext/*/*.ch',
+        root_rel / 'gst/*/*.ch',
+    '--gst_c_source_filters', excludes,
+    '--gst_cache_file', plugins_cache,
+    check: true,
+).stdout().split(pathsep)

gst-plugins-ugly-1.22.7.tar.xz/ext/meson.build -> gst-plugins-ugly-1.24.5.tar.xz/ext/meson.build Changed

gst-plugins-ugly-1.22.7.tar.xz/ext/mpeg2dec/gstmpeg2dec.c -> gst-plugins-ugly-1.24.5.tar.xz/ext/mpeg2dec/gstmpeg2dec.c Changed

@@ -82,7 +82,7 @@
     GstQuery * query);
 
 static void gst_mpeg2dec_clear_buffers (GstMpeg2dec * mpeg2dec);
-static gboolean gst_mpeg2dec_crop_buffer (GstMpeg2dec * dec,
+static GstFlowReturn gst_mpeg2dec_crop_buffer (GstMpeg2dec * dec,
     GstVideoCodecFrame * in_frame, GstVideoFrame * in_vframe);
 
 static void
@@ -548,7 +548,7 @@
   while ((l = g_list_first (mpeg2dec->buffers))) {
     GstMpeg2DecBuffer *mbuf = l->data;
     gst_video_frame_unmap (&mbuf->frame);
-    g_slice_free (GstMpeg2DecBuffer, mbuf);
+    g_free (mbuf);
     mpeg2dec->buffers = g_list_delete_link (mpeg2dec->buffers, l);
   }
 }
@@ -561,7 +561,7 @@
 
   GST_LOG_OBJECT (mpeg2dec, "Saving local info for frame %d", id);
 
-  mbuf = g_slice_new0 (GstMpeg2DecBuffer);
+  mbuf = g_new0 (GstMpeg2DecBuffer, 1);
   mbuf->id = id;
   mbuf->frame = *frame;
 
@@ -585,7 +585,7 @@
   if (l) {
     GstMpeg2DecBuffer *mbuf = l->data;
     gst_video_frame_unmap (&mbuf->frame);
-    g_slice_free (GstMpeg2DecBuffer, mbuf);
+    g_free (mbuf);
     mpeg2dec->buffers = g_list_delete_link (mpeg2dec->buffers, l);
     GST_LOG_OBJECT (mpeg2dec, "Discarded local info for frame %d", id);
   } else {

gst-plugins-ugly-1.22.7.tar.xz/ext/x264/gstx264enc.c -> gst-plugins-ugly-1.24.5.tar.xz/ext/x264/gstx264enc.c Changed

@@ -113,6 +113,16 @@
 GST_DEBUG_CATEGORY_STATIC (x264_enc_debug);
 #define GST_CAT_DEFAULT x264_enc_debug
 
+enum AllowedSubsamplingFlags
+{
+  ALLOW_400_8 = 1 << 0,
+  ALLOW_420_8 = 1 << 1,
+  ALLOW_420_10 = 1 << 2,
+  ALLOW_422 = 1 << 4,
+  ALLOW_444 = 1 << 5,
+  ALLOW_ANY = 0xffff
+};
+
 struct _GstX264EncVTable
 {
   GModule *module;
@@ -210,8 +220,7 @@
 
 static gboolean
 gst_x264_enc_add_x264_chroma_format (GstStructure * s,
-    gboolean allow_420_8, gboolean allow_420_10, gboolean allow_422,
-    gboolean allow_444)
+    enum AllowedSubsamplingFlags flags)
 {
   GValue fmts = G_VALUE_INIT;
   GValue fmt = G_VALUE_INIT;
@@ -223,17 +232,20 @@
   if (vtable_8bit) {
     gint chroma_format = *vtable_8bit->x264_chroma_format;
 
-    if ((chroma_format == 0 || chroma_format == X264_CSP_I444) && allow_444) {
+    if ((chroma_format == 0 || chroma_format == X264_CSP_I444) &&
+        flags & ALLOW_444) {
       g_value_set_string (&fmt, "Y444");
       gst_value_list_append_value (&fmts, &fmt);
     }
 
-    if ((chroma_format == 0 || chroma_format == X264_CSP_I422) && allow_422) {
+    if ((chroma_format == 0 || chroma_format == X264_CSP_I422) &&
+        flags & ALLOW_422) {
       g_value_set_string (&fmt, "Y42B");
       gst_value_list_append_value (&fmts, &fmt);
     }
 
-    if ((chroma_format == 0 || chroma_format == X264_CSP_I420) && allow_420_8) {
+    if ((chroma_format == 0 || chroma_format == X264_CSP_I420) &&
+        flags & ALLOW_420_8) {
       g_value_set_string (&fmt, "I420");
       gst_value_list_append_value (&fmts, &fmt);
       g_value_set_string (&fmt, "YV12");
@@ -241,12 +253,19 @@
       g_value_set_string (&fmt, "NV12");
       gst_value_list_append_value (&fmts, &fmt);
     }
+
+    if ((chroma_format == 0 || chroma_format == X264_CSP_I400) &&
+        flags & ALLOW_400_8) {
+      g_value_set_string (&fmt, "GRAY8");
+      gst_value_list_append_value (&fmts, &fmt);
+    }
   }
 
   if (vtable_10bit) {
     gint chroma_format = *vtable_10bit->x264_chroma_format;
 
-    if ((chroma_format == 0 || chroma_format == X264_CSP_I444) && allow_444) {
+    if ((chroma_format == 0 || chroma_format == X264_CSP_I444) &&
+        flags & ALLOW_444) {
       if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
         g_value_set_string (&fmt, "Y444_10LE");
       else
@@ -255,7 +274,8 @@
       gst_value_list_append_value (&fmts, &fmt);
     }
 
-    if ((chroma_format == 0 || chroma_format == X264_CSP_I422) && allow_422) {
+    if ((chroma_format == 0 || chroma_format == X264_CSP_I422) &&
+        flags & ALLOW_422) {
       if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
         g_value_set_string (&fmt, "I422_10LE");
       else
@@ -264,7 +284,8 @@
       gst_value_list_append_value (&fmts, &fmt);
     }
 
-    if ((chroma_format == 0 || chroma_format == X264_CSP_I420) && allow_420_10) {
+    if ((chroma_format == 0 || chroma_format == X264_CSP_I420) &&
+        flags & ALLOW_420_10) {
       if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
         g_value_set_string (&fmt, "I420_10LE");
       else
@@ -785,20 +806,20 @@
 }
 
 static void
-check_formats (const gchar * str, gboolean * has_420_8, gboolean * has_420_10,
-    gboolean * has_422, gboolean * has_444)
+check_formats (const gchar * str, enum AllowedSubsamplingFlags *flags)
 {
   if (g_str_has_prefix (str, "high-4:4:4"))
-    *has_444 = TRUE;
+    *flags |= ALLOW_444;
   else if (g_str_has_prefix (str, "high-4:2:2"))
-    *has_422 = TRUE;
+    *flags |= ALLOW_422;
   else if (g_str_has_prefix (str, "high-10"))
-    *has_420_10 = TRUE;
+    *flags |= ALLOW_420_10;
+  else if (g_str_has_prefix (str, "high"))
+    *flags |= ALLOW_420_8 | ALLOW_400_8;
   else
-    *has_420_8 = TRUE;
+    *flags |= ALLOW_420_8;
 }
 
-
 /* allowed input caps depending on whether libx264 was built for 8 or 10 bits */
 static GstCaps *
 gst_x264_enc_sink_getcaps (GstVideoEncoder * enc, GstCaps * filter)
@@ -853,26 +874,20 @@
         gst_structure_set_value (s, "chroma-site", val);
 
       if ((val = gst_structure_get_value (allowed_s, "profile"))) {
-        gboolean has_420_8 = FALSE;
-        gboolean has_420_10 = FALSE;
-        gboolean has_422 = FALSE;
-        gboolean has_444 = FALSE;
+        enum AllowedSubsamplingFlags flags = 0;
 
         if (G_VALUE_HOLDS_STRING (val)) {
-          check_formats (g_value_get_string (val), &has_420_8, &has_420_10,
-              &has_422, &has_444);
+          check_formats (g_value_get_string (val), &flags);
         } else if (GST_VALUE_HOLDS_LIST (val)) {
           for (k = 0; k < gst_value_list_get_size (val); k++) {
             const GValue *vlist = gst_value_list_get_value (val, k);
 
             if (G_VALUE_HOLDS_STRING (vlist))
-              check_formats (g_value_get_string (vlist), &has_420_8,
-                  &has_420_10, &has_422, &has_444);
+              check_formats (g_value_get_string (vlist), &flags);
           }
         }
 
-        gst_x264_enc_add_x264_chroma_format (s, has_420_8, has_420_10, has_422,
-            has_444);
+        gst_x264_enc_add_x264_chroma_format (s, flags);
       }
 
       filter_caps = gst_caps_merge_structure (filter_caps, s);
@@ -1211,7 +1226,7 @@
       "height", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
 
   gst_x264_enc_add_x264_chroma_format (gst_caps_get_structure
-      (supported_sinkcaps, 0), TRUE, TRUE, TRUE, TRUE);
+      (supported_sinkcaps, 0), ALLOW_ANY);
 
   sink_templ = gst_pad_template_new ("sink",
       GST_PAD_SINK, GST_PAD_ALWAYS, supported_sinkcaps);
@@ -1346,7 +1361,7 @@
   if (!gst_video_frame_map (&vframe, info, frame->input_buffer, GST_MAP_READ))
     return NULL;
 
-  fdata = g_slice_new (FrameData);
+  fdata = g_new (FrameData, 1);
   fdata->frame = gst_video_codec_frame_ref (frame);
   fdata->vframe = vframe;
 
@@ -1368,7 +1383,7 @@
 
     gst_video_frame_unmap (&fdata->vframe);
     gst_video_codec_frame_unref (fdata->frame);
-    g_slice_free (FrameData, fdata);
+    g_free (fdata);
 
     enc->pending_frames = g_list_delete_link (enc->pending_frames, l);
     return;
@@ -1385,7 +1400,7 @@
 
     gst_video_frame_unmap (&fdata->vframe);
     gst_video_codec_frame_unref (fdata->frame);
-    g_slice_free (FrameData, fdata);
+    g_free (fdata);
   }
   g_list_free (enc->pending_frames);
   enc->pending_frames = NULL;
@@ -1517,6 +1532,10 @@
 gst_x264_enc_gst_to_x264_video_format (GstVideoFormat format, gint * nplanes)
 {
   switch (format) {
+    case GST_VIDEO_FORMAT_GRAY8:
+      if (nplanes)
+        *nplanes = 1;
+      return X264_CSP_I400;
     case GST_VIDEO_FORMAT_I420:
     case GST_VIDEO_FORMAT_YV12:
       if (nplanes)
@@ -2332,7 +2351,13 @@
     if (info->finfo->format == old->finfo->format
         && info->width == old->width && info->height == old->height
         && info->fps_n == old->fps_n && info->fps_d == old->fps_d
-        && info->par_n == old->par_n && info->par_d == old->par_d) {
+        && info->par_n == old->par_n && info->par_d == old->par_d
+        && info->interlace_mode == old->interlace_mode
+        && gst_video_colorimetry_is_equal (&info->colorimetry,
+            &old->colorimetry)
+        && GST_VIDEO_INFO_CHROMA_SITE (info) == GST_VIDEO_INFO_CHROMA_SITE (old)
+        && GST_VIDEO_INFO_MULTIVIEW_MODE (info) ==
+        GST_VIDEO_INFO_MULTIVIEW_MODE (old)) {
       gst_video_codec_state_unref (encoder->input_state);
       encoder->input_state = gst_video_codec_state_ref (state);
       return TRUE;

gst-plugins-ugly-1.22.7.tar.xz/ext/x264/meson.build -> gst-plugins-ugly-1.24.5.tar.xz/ext/x264/meson.build Changed

gst-plugins-ugly-1.22.7.tar.xz/gst-plugins-ugly.doap -> gst-plugins-ugly-1.24.5.tar.xz/gst-plugins-ugly.doap Changed

@@ -35,71 +35,91 @@
 
  <release>
   <Version>
-   <revision>1.22.7</revision>
-   <branch>1.22</branch>
+   <revision>1.24.5</revision>
+   <branch>1.24</branch>
    <name></name>
-   <created>2023-11-13</created>
-   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-ugly/gst-plugins-ugly-1.22.7.tar.xz" />
+   <created>2024-06-20</created>
+   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-ugly/gst-plugins-ugly-1.24.5.tar.xz" />
   </Version>
  </release>
 
  <release>
   <Version>
-   <revision>1.22.6</revision>
-   <branch>1.22</branch>
+   <revision>1.24.4</revision>
+   <branch>1.24</branch>
    <name></name>
-   <created>2023-09-20</created>
-   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-ugly/gst-plugins-ugly-1.22.6.tar.xz" />
+   <created>2024-05-29</created>
+   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-ugly/gst-plugins-ugly-1.24.4.tar.xz" />
   </Version>
  </release>
 
  <release>
   <Version>
-   <revision>1.22.5</revision>
-   <branch>1.22</branch>
+   <revision>1.24.3</revision>
+   <branch>1.24</branch>
    <name></name>
-   <created>2023-07-20</created>
-   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-ugly/gst-plugins-ugly-1.22.5.tar.xz" />
+   <created>2024-04-30</created>
+   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-ugly/gst-plugins-ugly-1.24.3.tar.xz" />
   </Version>
  </release>
 
  <release>
   <Version>
-   <revision>1.22.4</revision>
-   <branch>1.22</branch>
+   <revision>1.24.2</revision>
+   <branch>1.24</branch>
    <name></name>
-   <created>2023-06-20</created>
-   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-ugly/gst-plugins-ugly-1.22.4.tar.xz" />
+   <created>2024-04-09</created>
+   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-ugly/gst-plugins-ugly-1.24.2.tar.xz" />
   </Version>
  </release>
 
  <release>
   <Version>
-   <revision>1.22.3</revision>
-   <branch>1.22</branch>
+   <revision>1.24.1</revision>
+   <branch>1.24</branch>
    <name></name>
-   <created>2023-05-19</created>
-   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-ugly/gst-plugins-ugly-1.22.3.tar.xz" />
+   <created>2024-03-21</created>
+   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-ugly/gst-plugins-ugly-1.24.1.tar.xz" />
   </Version>
  </release>
 
  <release>
   <Version>
-   <revision>1.22.2</revision>
-   <branch>1.22</branch>
+   <revision>1.24.0</revision>
+   <branch>main</branch>
+   <name></name>
+   <created>2024-03-04</created>
+   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-ugly/gst-plugins-ugly-1.24.0.tar.xz" />
+  </Version>
+ </release>
+
+ <release>
+  <Version>
+   <revision>1.23.90</revision>
+   <branch>main</branch>
    <name></name>
-   <created>2023-04-11</created>
-   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-ugly/gst-plugins-ugly-1.22.2.tar.xz" />
+   <created>2024-02-23</created>
+   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-ugly/gst-plugins-ugly-1.23.90.tar.xz" />
   </Version>
  </release>
 
  <release>
   <Version>
-   <revision>1.22.1</revision>
-   <branch>1.22</branch>
+   <revision>1.23.2</revision>
+   <branch>main</branch>
+   <name></name>
+   <created>2024-02-15</created>
+   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-ugly/gst-plugins-ugly-1.23.2.tar.xz" />
+  </Version>
+ </release>
+
+ <release>
+  <Version>
+   <revision>1.23.1</revision>
+   <branch>main</branch>
    <name></name>
-   <created>2023-03-04</created>
-   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-ugly/gst-plugins-ugly-1.22.1.tar.xz" />
+   <created>2024-02-06</created>
+   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-ugly/gst-plugins-ugly-1.23.1.tar.xz" />
   </Version>
  </release>

gst-plugins-ugly-1.22.7.tar.xz/gst/asfdemux/gstasfdemux.c -> gst-plugins-ugly-1.24.5.tar.xz/gst/asfdemux/gstasfdemux.c Changed

gst-plugins-ugly-1.22.7.tar.xz/gst/asfdemux/gstrtpasfdepay.c -> gst-plugins-ugly-1.24.5.tar.xz/gst/asfdemux/gstrtpasfdepay.c Changed

@@ -104,6 +104,9 @@
 static void
 gst_rtp_asf_depay_init (GstRtpAsfDepay * depay)
 {
+  gst_rtp_base_depayload_set_aggregate_hdrext_enabled (GST_RTP_BASE_DEPAYLOAD
+      (depay), TRUE);
+
   depay->adapter = gst_adapter_new ();
 }
 
@@ -335,6 +338,7 @@
   GstRtpAsfDepay *depay;
   const guint8 *payload;
   GstBuffer *outbuf;
+  GstBufferList *outbufs = NULL;
   gboolean S, L, R, D, I;
   guint payload_len, hdr_len, offset;
   guint len_offs;
@@ -359,6 +363,7 @@
 
   GST_LOG_OBJECT (depay, "got payload len of %u", payload_len);
 
+  outbufs = gst_buffer_list_new ();
   do {
     guint packet_len;
 
@@ -473,9 +478,12 @@
           gst_rtp_buffer_get_payload_subbuffer (&rtpbuf, offset, packet_len);
     }
 
-    /* If we haven't completed a full ASF packet, return */
-    if (!outbuf)
+    /* If we haven't completed a full ASF packet, return but first
+       push what we have so far */
+    if (!outbuf) {
+      gst_rtp_base_depayload_push_list (depayload, outbufs);
       return NULL;
+    }
 
     outbuf = gst_rtp_asf_depay_update_padding (depay, outbuf);
 
@@ -490,7 +498,7 @@
 
     GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
 
-    gst_rtp_base_depayload_push (depayload, outbuf);
+    gst_buffer_list_add (outbufs, outbuf);
 
     /* only apply the timestamp to the first buffer of this packet */
     timestamp = -1;
@@ -501,6 +509,8 @@
     payload_len -= packet_len;
   } while (payload_len > 0);
 
+  gst_rtp_base_depayload_push_list (depayload, outbufs);
+
   gst_rtp_buffer_unmap (&rtpbuf);
 
   return NULL;
@@ -511,6 +521,12 @@
     gst_rtp_buffer_unmap (&rtpbuf);
     GST_WARNING_OBJECT (depayload, "Payload too small, expected at least 4 "
         "bytes for header, but got only %d bytes", payload_len);
+    if (gst_buffer_list_length (outbufs) == 0) {
+      gst_rtp_base_depayload_dropped (depayload);
+      gst_buffer_list_unref (outbufs);
+    } else {
+      gst_rtp_base_depayload_push_list (depayload, outbufs);
+    }
     return NULL;
   }
 }

gst-plugins-ugly-1.22.7.tar.xz/gst/realmedia/meson.build -> gst-plugins-ugly-1.24.5.tar.xz/gst/realmedia/meson.build Changed

gst-plugins-ugly-1.22.7.tar.xz/gst/realmedia/realmedia.c -> gst-plugins-ugly-1.24.5.tar.xz/gst/realmedia/realmedia.c Changed

gst-plugins-ugly-1.22.7.tar.xz/meson.build -> gst-plugins-ugly-1.24.5.tar.xz/meson.build Changed

@@ -1,6 +1,6 @@
 project('gst-plugins-ugly', 'c',
-  version : '1.22.7',
-  meson_version : '>= 0.62',
+  version : '1.24.5',
+  meson_version : '>= 1.1',
   default_options :  'warning_level=1',
                       'buildtype=debugoptimized' )
 
@@ -19,7 +19,7 @@
 
 have_cxx = add_languages('cpp', native: false, required: false)
 
-glib_req = '>= 2.62.0'
+glib_req = '>= 2.64.0'
 
 if gst_version_is_stable
   gst_req = '>= @0@.@1@.0'.format(gst_version_major, gst_version_minor)
@@ -66,8 +66,6 @@
       '/we4053', # one void operand for '?:'
       '/we4062', # enumerator 'identifier' in switch of enum 'enumeration' is not handled
       '/we4098', # 'function' : void function returning a value
-      '/we4101', # 'identifier' : unreferenced local variable
-      '/we4189', # 'identifier' : local variable is initialized but not referenced
     )
   endif
   if have_cxx
@@ -220,13 +218,28 @@
 
 if gst_debug_disabled
   message('GStreamer debug system is disabled')
-  if cc.has_argument('-Wno-unused')
-    add_project_arguments('-Wno-unused', language: 'c')
-  endif
-  if have_cxx and cxx.has_argument ('-Wno-unused')
-    add_project_arguments('-Wno-unused', language: 'cpp')
+  if cc.get_argument_syntax() == 'msvc'
+    msvc_args = cc.get_supported_arguments(
+      '/wd4101', # 'identifier' : unreferenced local variable
+      '/wd4189', # 'identifier' : local variable is initialized but not referenced
+    )
+    add_project_arguments(msvc_args, language: 'c', 'cpp')
+  else
+    if cc.has_argument('-Wno-unused')
+      add_project_arguments('-Wno-unused', language: 'c')
+    endif
+    if have_cxx and cxx.has_argument ('-Wno-unused')
+      add_project_arguments('-Wno-unused', language: 'cpp')
+    endif
   endif
 else
+  if cc.get_argument_syntax() == 'msvc' and gst_version_is_dev
+    msvc_args = cc.get_supported_arguments(
+      '/we4101', # 'identifier' : unreferenced local variable
+      '/we4189', # 'identifier' : local variable is initialized but not referenced
+    )
+    add_project_arguments(msvc_args, language: 'c', 'cpp')
+  endif
   message('GStreamer debug system is enabled')
 endif
 
@@ -329,7 +342,7 @@
 
 configure_file(output : 'config.h', configuration : cdata)
 
-meson.add_dist_script('scripts/gen-changelog.py', meson.project_name(), '1.20.0', meson.project_version())
+meson.add_dist_script('scripts/gen-changelog.py', meson.project_name(), '1.22.0', meson.project_version())
 
 plugin_names = 
 gst_plugins =

gst-plugins-ugly-1.22.7.tar.xz/meson_options.txt -> gst-plugins-ugly-1.24.5.tar.xz/meson_options.txt Changed

gst-plugins-ugly-1.22.7.tar.xz/po/gst-plugins-ugly-1.0.pot -> gst-plugins-ugly-1.24.5.tar.xz/po/gst-plugins-ugly-1.0.pot Changed

gst-plugins-ugly-1.22.7.tar.xz/po/gst-plugins-ugly.pot -> gst-plugins-ugly-1.24.5.tar.xz/po/gst-plugins-ugly.pot Changed

gst-plugins-ugly-1.22.7.tar.xz/po/lv.po -> gst-plugins-ugly-1.24.5.tar.xz/po/lv.po Changed

@@ -3,50 +3,50 @@
 # Arvis Lācis <arvis.lacis@inbox.lv>, 2009.
 # Rihards Prieditis <rprieditis@gmail.com>, 2011.
 # Rūdolfs Mazurs <rudolfs.mazurs@gmail.com>, 2014.
+# Rihards Prieditis <rprieditis@gmail.com>, 2023.
 msgid ""
 msgstr ""
-"Project-Id-Version: gst-plugins-ugly 1.2.1\n"
+"Project-Id-Version: gst-plugins-ugly 1.21.90\n"
 "Report-Msgid-Bugs-To: \n"
-"POT-Creation-Date: 2021-10-25 01:02+0100\n"
-"PO-Revision-Date: 2014-04-20 16:24+0300\n"
-"Last-Translator: Rihards Prieditis <rprieditis@gmail.com>\n"
+"POT-Creation-Date: 2023-01-13 19:11+0000\n"
+"PO-Revision-Date: 2023-12-24 15:38+0000\n"
+"Last-Translator: Rihards Priedītis <rprieditis@gmail.com>\n"
 "Language-Team: Latvian <translation-team-lv@lists.sourceforge.net>\n"
 "Language: lv\n"
 "MIME-Version: 1.0\n"
 "Content-Type: text/plain; charset=UTF-8\n"
 "Content-Transfer-Encoding: 8bit\n"
+"Plural-Forms: nplurals=3; plural=(n%10==0 || (n%100>=11 && n%100<=19) ? 0 : "
+"n%10==1 && n%100!=11 ? 1 : 2);\n"
 "X-Bugs: Report translation errors to the Language-Team address.\n"
-"Plural-Forms: nplurals=3; plural=(n%10==1 && n%100!=11 ? 0 : n != 0 ? 1 : "
-"2);\n"
-"X-Generator: Lokalize 1.5\n"
+"X-Generator: Poedit 3.4.2\n"
 
 msgid "Could not read from CD."
-msgstr "Nevarēja nolasīt no CD."
+msgstr "Nevar nolasīt no CD."
 
 msgid "Could not open CD device for reading."
-msgstr "Nevarēja atvērt CD ierīci lasīšanai."
+msgstr "Nevar atvērt CD ierīci lasīšanai."
 
 msgid "Disc is not an Audio CD."
 msgstr "Disks nav Audio CD."
 
 msgid "Could not open DVD"
-msgstr "Nevarēja atvērt DVD"
+msgstr "Nevar atvērt DVD"
 
 #, c-format
 msgid "Could not open DVD title %d"
-msgstr "Nevarēja atvērt DVD nosaukumu %d"
+msgstr "Nevar atvērt DVD sadaļu %d"
 
 #, c-format
 msgid "Failed to go to chapter %d of DVD title %d"
-msgstr "Neizdevās pāriet uz %d nodaļu DVD nosaukumā %d"
+msgstr "Neizdevās pāriet uz %d nodaļu DVD sadaļā %d"
 
 #, c-format
 msgid ""
 "Could not open DVD title %d. Interactive titles are not supported by this "
 "element"
 msgstr ""
-"Nevarēja atvērt DVD nosaukumu %d. Šis elements nenodrošina interaktīvos "
-"nosaukumus."
+"Nevar atvērt DVD sadaļu %d. Šis elements neatbalsta interaktīvus nosaukumus"
 
 msgid ""
 "Could not read DVD. This may be because the DVD is encrypted and a DVD "
@@ -56,10 +56,10 @@
 "bibliotēka nav instalēta."
 
 msgid "Could not read DVD."
-msgstr "Nevarēja nolasīt DVD."
+msgstr "Nevar nolasīt DVD."
 
 msgid "This stream contains no data."
-msgstr "Šī straume nesatur datus."
+msgstr "Šajā plūsmā nav datu."
 
 #~ msgid ""
 #~ "Failed to configure LAME mp3 audio encoder. Check your encoding "
@@ -68,7 +68,6 @@
 #~ "Neizdevās konfigurēt LAME mp3 audio kodētāju. Pārbaudiet savus kodēšanas "
 #~ "parametrus."
 
-#, c-format
 #~ msgid ""
 #~ "The requested bitrate %d kbit/s for property '%s' is not allowed. The "
 #~ "bitrate was changed to %d kbit/s."

gst-plugins-ugly-1.22.7.tar.xz/scripts/gen-changelog.py -> gst-plugins-ugly-1.24.5.tar.xz/scripts/gen-changelog.py Changed

gst-plugins-ugly-1.22.7.tar.xz/tests/check/elements/x264enc.c -> gst-plugins-ugly-1.24.5.tar.xz/tests/check/elements/x264enc.c Changed

gst-plugins-ugly-1.22.7.tar.xz/tests/check/gst-plugins-ugly.supp -> gst-plugins-ugly-1.24.5.tar.xz/tests/check/gst-plugins-ugly.supp Changed

@@ -1,190 +0,0 @@
-### amrnb suppressions
-### these should/could be submitted to amrnb
-
-{
-   <amrnb>
-   Memcheck:Cond
-   fun:Encoder_Interface_Encode
-   fun:gst_amrnbenc_chain
-   fun:gst_pad_chain
-   fun:gst_pad_push
-}
-
-{
-   <amrnb>
-   Memcheck:Cond
-   fun:Speech_Encode_Frame
-   fun:Encoder_Interface_Encode
-   fun:gst_amrnbenc_chain
-   fun:gst_pad_chain
-   fun:gst_pad_push
-}
-
-{
-   <amrnb>
-   Memcheck:Cond
-   obj:/usr/lib/libamrnb.so*
-   fun:Speech_Encode_Frame
-   fun:Encoder_Interface_Encode
-   fun:gst_amrnbenc_chain
-   fun:gst_pad_chain
-   fun:gst_pad_push
-}
-
-{
-   <amrnb>
-   Memcheck:Cond
-   obj:/usr/lib/libamrnb.so*
-   obj:/usr/lib/libamrnb.so*
-   fun:Speech_Encode_Frame
-   fun:Encoder_Interface_Encode
-   fun:gst_amrnbenc_chain
-   fun:gst_pad_chain
-   fun:gst_pad_push
-}
-
-{
-   <amrnb>
-   Memcheck:Cond
-   obj:/usr/lib/libamrnb.so*
-   obj:/usr/lib/libamrnb.so*
-   obj:/usr/lib/libamrnb.so*
-   fun:Speech_Encode_Frame
-   fun:Encoder_Interface_Encode
-   fun:gst_amrnbenc_chain
-   fun:gst_pad_chain
-   fun:gst_pad_push
-}
-
-{
-   <amrnb acos>
-   Memcheck:Cond
-   fun:acos
-   obj:/usr/lib/libamrnb.so*
-   obj:/usr/lib/libamrnb.so*
-   fun:Speech_Encode_Frame
-   fun:Encoder_Interface_Encode
-   fun:gst_amrnbenc_chain
-   fun:gst_pad_chain
-   fun:gst_pad_push
-}
-
-{
-   <AMR encoder>
-   Memcheck:Cond
-   fun:Levinson
-   fun:Speech_Encode_Frame
-   fun:Encoder_Interface_Encode
-}
-
-{
-   <AMR encoder>
-   Memcheck:Cond
-   fun:Az_lsp
-   fun:Speech_Encode_Frame
-   fun:Encoder_Interface_Encode
-}
-
-{
-   <AMR encoder>
-   Memcheck:Cond
-   fun:acos
-   fun:Lsp_lsf
-   fun:Speech_Encode_Frame
-   fun:Encoder_Interface_Encode
-}
-
-{
-   <AMR encoder>
-   Memcheck:Cond
-   fun:Lsf_wt
-   fun:Speech_Encode_Frame
-   fun:Encoder_Interface_Encode
-}
-{
-   <AMR encoder>
-   Memcheck:Cond
-   fun:Vq_subvec
-   fun:Speech_Encode_Frame
-   fun:Encoder_Interface_Encode
-}
-
-{
-   <AMR encoder>
-   Memcheck:Cond
-   fun:Lag_max
-   fun:Pitch_ol
-   fun:ol_ltp
-   fun:Speech_Encode_Frame
-   fun:Encoder_Interface_Encode
-}
-
-{
-   <AMR encoder>
-   Memcheck:Cond
-   fun:Pitch_ol
-   fun:ol_ltp
-   fun:Speech_Encode_Frame
-   fun:Encoder_Interface_Encode
-}
-
-{
-   <AMR encoder>
-   Memcheck:Cond
-   fun:cl_ltp
-   fun:Speech_Encode_Frame
-   fun:Encoder_Interface_Encode
-}
-
-{
-   <AMR encoder>
-   Memcheck:Cond
-   fun:searchFrac
-   fun:cl_ltp
-   fun:Speech_Encode_Frame
-   fun:Encoder_Interface_Encode
-}
-
-{
-   <AMR encoder>
-   Memcheck:Cond
-   fun:q_gain_pitch
-   fun:cl_ltp
-   fun:Speech_Encode_Frame
-   fun:Encoder_Interface_Encode
-}
-
-{
-   <AMR encoder>
-   Memcheck:Cond
-   fun:set_sign12k2
-   fun:code_10i40_35bits
-   fun:cbsearch
-   fun:Speech_Encode_Frame
-   fun:Encoder_Interface_Encode
-}
-
-{
-   <AMR encoder>
-   Memcheck:Cond
-   fun:code_10i40_35bits
-   fun:cbsearch
-   fun:Speech_Encode_Frame
-   fun:Encoder_Interface_Encode
-}
-
-{
-   <AMR encoder>
-   Memcheck:Cond
-   fun:gainQuant
-   fun:Speech_Encode_Frame
-   fun:Encoder_Interface_Encode
-}
-
-{
-   <AMR encoder on FC4>
-   Memcheck:Value2
-   obj:/usr/lib/libamrnb.so.0.0.0
-   fun:Speech_Encode_Frame
-   fun:Encoder_Interface_Encode
-}

gst-plugins-ugly-1.22.7.tar.xz/tests/check/meson.build -> gst-plugins-ugly-1.24.5.tar.xz/tests/check/meson.build Changed

gst-plugins-ugly-1.22.7.tar.xz/tests/meson.build -> gst-plugins-ugly-1.24.5.tar.xz/tests/meson.build Changed

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zaitor created request 4 months ago

New stable branch

zaitor accepted request 4 months ago

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