Sync from SUSE:ALP:Source:Standard:1.0 gstreamer-plugins-bad revision 8fdd5b90bef1bc3e8abd5dfeb16d8223

2024-02-10 20:12:34 +01:00 · 2024-02-10 20:12:34 +01:00 · 0185800098
commit 0185800098
10 changed files with 6391 additions and 0 deletions
--- a/.gitattributes
+++ b/.gitattributes
@ -0,0 +1,23 @@
+## Default LFS
+*.7z filter=lfs diff=lfs merge=lfs -text
+*.bsp filter=lfs diff=lfs merge=lfs -text
+*.bz2 filter=lfs diff=lfs merge=lfs -text
+*.gem filter=lfs diff=lfs merge=lfs -text
+*.gz filter=lfs diff=lfs merge=lfs -text
+*.jar filter=lfs diff=lfs merge=lfs -text
+*.lz filter=lfs diff=lfs merge=lfs -text
+*.lzma filter=lfs diff=lfs merge=lfs -text
+*.obscpio filter=lfs diff=lfs merge=lfs -text
+*.oxt filter=lfs diff=lfs merge=lfs -text
+*.pdf filter=lfs diff=lfs merge=lfs -text
+*.png filter=lfs diff=lfs merge=lfs -text
+*.rpm filter=lfs diff=lfs merge=lfs -text
+*.tbz filter=lfs diff=lfs merge=lfs -text
+*.tbz2 filter=lfs diff=lfs merge=lfs -text
+*.tgz filter=lfs diff=lfs merge=lfs -text
+*.ttf filter=lfs diff=lfs merge=lfs -text
+*.txz filter=lfs diff=lfs merge=lfs -text
+*.whl filter=lfs diff=lfs merge=lfs -text
+*.xz filter=lfs diff=lfs merge=lfs -text
+*.zip filter=lfs diff=lfs merge=lfs -text
+*.zst filter=lfs diff=lfs merge=lfs -text
--- a/0001-Update-code-for-webrtc-audio-processing-1.patch
+++ b/0001-Update-code-for-webrtc-audio-processing-1.patch
@ -0,0 +1,912 @@
+From d5755744c3e2b70e9f04704ae9d18b928d9fa456 Mon Sep 17 00:00:00 2001
+From: Arun Raghavan <arun@asymptotic.io>
+Date: Wed, 2 Dec 2020 18:31:44 -0500
+Subject: [PATCH 1/2] webrtcdsp: Update code for webrtc-audio-processing-1
+
+Updated API usage appropriately, and now we have a versioned package to
+track breaking vs. non-breaking updates.
+
+Deprecates a number of properties (and we have to plug in our own values
+for related enums which are now gone):
+
+  * echo-suprression-level
+  * experimental-agc
+  * extended-filter
+  * delay-agnostic
+  * voice-detection-frame-size-ms
+  * voice-detection-likelihood
+
+Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2943>
+---
+ .../ext/webrtcdsp/gstwebrtcdsp.cpp            | 271 +++++++-----------
+ .../ext/webrtcdsp/gstwebrtcechoprobe.cpp      |  87 +++---
+ .../ext/webrtcdsp/gstwebrtcechoprobe.h        |   9 +-
+ .../gst-plugins-bad/ext/webrtcdsp/meson.build |   4 +-
+ 4 files changed, 164 insertions(+), 207 deletions(-)
+
+diff --git a/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcdsp.cpp b/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcdsp.cpp
+index 7ee09488fb7..c9a7cdae2f4 100644
+--- a/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcdsp.cpp
+++ b/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcdsp.cpp
+@@ -71,9 +71,7 @@
+ #include "gstwebrtcdsp.h"
+ #include "gstwebrtcechoprobe.h"
+ 
+-#include <webrtc/modules/audio_processing/include/audio_processing.h>
+-#include <webrtc/modules/interface/module_common_types.h>
+-#include <webrtc/system_wrappers/include/trace.h>
+#include <modules/audio_processing/include/audio_processing.h>
+ 
+ GST_DEBUG_CATEGORY (webrtc_dsp_debug);
+ #define GST_CAT_DEFAULT (webrtc_dsp_debug)
+@@ -82,10 +80,9 @@ GST_DEBUG_CATEGORY (webrtc_dsp_debug);
+ #define DEFAULT_COMPRESSION_GAIN_DB 9
+ #define DEFAULT_STARTUP_MIN_VOLUME 12
+ #define DEFAULT_LIMITER TRUE
+-#define DEFAULT_GAIN_CONTROL_MODE webrtc::GainControl::kAdaptiveDigital
+#define DEFAULT_GAIN_CONTROL_MODE webrtc::AudioProcessing::Config::GainController1::Mode::kAdaptiveDigital
+ #define DEFAULT_VOICE_DETECTION FALSE
+ #define DEFAULT_VOICE_DETECTION_FRAME_SIZE_MS 10
+-#define DEFAULT_VOICE_DETECTION_LIKELIHOOD webrtc::VoiceDetection::kLowLikelihood
+ 
+ static GstStaticPadTemplate gst_webrtc_dsp_sink_template =
+ GST_STATIC_PAD_TEMPLATE ("sink",
+@@ -119,7 +116,7 @@ GST_STATIC_PAD_TEMPLATE ("src",
+         "channels = (int) [1, MAX]")
+     );
+ 
+-typedef webrtc::EchoCancellation::SuppressionLevel GstWebrtcEchoSuppressionLevel;
+typedef int GstWebrtcEchoSuppressionLevel;
+ #define GST_TYPE_WEBRTC_ECHO_SUPPRESSION_LEVEL \
+     (gst_webrtc_echo_suppression_level_get_type ())
+ static GType
+@@ -127,10 +124,9 @@ gst_webrtc_echo_suppression_level_get_type (void)
+ {
+   static GType suppression_level_type = 0;
+   static const GEnumValue level_types[] = {
+-    {webrtc::EchoCancellation::kLowSuppression, "Low Suppression", "low"},
+-    {webrtc::EchoCancellation::kModerateSuppression,
+-      "Moderate Suppression", "moderate"},
+-    {webrtc::EchoCancellation::kHighSuppression, "high Suppression", "high"},
+    {1, "Low Suppression", "low"},
+    {2, "Moderate Suppression", "moderate"},
+    {3, "high Suppression", "high"},
+     {0, NULL, NULL}
+   };
+ 
+@@ -141,7 +137,7 @@ gst_webrtc_echo_suppression_level_get_type (void)
+   return suppression_level_type;
+ }
+ 
+-typedef webrtc::NoiseSuppression::Level GstWebrtcNoiseSuppressionLevel;
+typedef webrtc::AudioProcessing::Config::NoiseSuppression::Level GstWebrtcNoiseSuppressionLevel;
+ #define GST_TYPE_WEBRTC_NOISE_SUPPRESSION_LEVEL \
+     (gst_webrtc_noise_suppression_level_get_type ())
+ static GType
+@@ -149,10 +145,10 @@ gst_webrtc_noise_suppression_level_get_type (void)
+ {
+   static GType suppression_level_type = 0;
+   static const GEnumValue level_types[] = {
+-    {webrtc::NoiseSuppression::kLow, "Low Suppression", "low"},
+-    {webrtc::NoiseSuppression::kModerate, "Moderate Suppression", "moderate"},
+-    {webrtc::NoiseSuppression::kHigh, "High Suppression", "high"},
+-    {webrtc::NoiseSuppression::kVeryHigh, "Very High Suppression",
+    {webrtc::AudioProcessing::Config::NoiseSuppression::Level::kLow, "Low Suppression", "low"},
+    {webrtc::AudioProcessing::Config::NoiseSuppression::Level::kModerate, "Moderate Suppression", "moderate"},
+    {webrtc::AudioProcessing::Config::NoiseSuppression::Level::kHigh, "High Suppression", "high"},
+    {webrtc::AudioProcessing::Config::NoiseSuppression::Level::kVeryHigh, "Very High Suppression",
+       "very-high"},
+     {0, NULL, NULL}
+   };
+@@ -164,7 +160,7 @@ gst_webrtc_noise_suppression_level_get_type (void)
+   return suppression_level_type;
+ }
+ 
+-typedef webrtc::GainControl::Mode GstWebrtcGainControlMode;
+typedef webrtc::AudioProcessing::Config::GainController1::Mode GstWebrtcGainControlMode;
+ #define GST_TYPE_WEBRTC_GAIN_CONTROL_MODE \
+     (gst_webrtc_gain_control_mode_get_type ())
+ static GType
+@@ -172,8 +168,9 @@ gst_webrtc_gain_control_mode_get_type (void)
+ {
+   static GType gain_control_mode_type = 0;
+   static const GEnumValue mode_types[] = {
+-    {webrtc::GainControl::kAdaptiveDigital, "Adaptive Digital", "adaptive-digital"},
+-    {webrtc::GainControl::kFixedDigital, "Fixed Digital", "fixed-digital"},
+    {webrtc::AudioProcessing::Config::GainController1::kAdaptiveDigital, "Adaptive Digital", "adaptive-digital"},
+    {webrtc::AudioProcessing::Config::GainController1::kFixedDigital, "Fixed Digital", "fixed-digital"},
+    {webrtc::AudioProcessing::Config::GainController1::kAdaptiveAnalog, "Adaptive Analog", "adaptive-analog"},
+     {0, NULL, NULL}
+   };
+ 
+@@ -184,7 +181,7 @@ gst_webrtc_gain_control_mode_get_type (void)
+   return gain_control_mode_type;
+ }
+ 
+-typedef webrtc::VoiceDetection::Likelihood GstWebrtcVoiceDetectionLikelihood;
+typedef int GstWebrtcVoiceDetectionLikelihood;
+ #define GST_TYPE_WEBRTC_VOICE_DETECTION_LIKELIHOOD \
+     (gst_webrtc_voice_detection_likelihood_get_type ())
+ static GType
+@@ -192,10 +189,10 @@ gst_webrtc_voice_detection_likelihood_get_type (void)
+ {
+   static GType likelihood_type = 0;
+   static const GEnumValue likelihood_types[] = {
+-    {webrtc::VoiceDetection::kVeryLowLikelihood, "Very Low Likelihood", "very-low"},
+-    {webrtc::VoiceDetection::kLowLikelihood, "Low Likelihood", "low"},
+-    {webrtc::VoiceDetection::kModerateLikelihood, "Moderate Likelihood", "moderate"},
+-    {webrtc::VoiceDetection::kHighLikelihood, "High Likelihood", "high"},
+    {1, "Very Low Likelihood", "very-low"},
+    {2, "Low Likelihood", "low"},
+    {3, "Moderate Likelihood", "moderate"},
+    {4, "High Likelihood", "high"},
+     {0, NULL, NULL}
+   };
+ 
+@@ -227,6 +224,7 @@ enum
+   PROP_VOICE_DETECTION,
+   PROP_VOICE_DETECTION_FRAME_SIZE_MS,
+   PROP_VOICE_DETECTION_LIKELIHOOD,
+  PROP_EXTRA_DELAY_MS,
+ };
+ 
+ /**
+@@ -248,7 +246,7 @@ struct _GstWebrtcDsp
+   /* Protected by the stream lock */
+   GstAdapter *adapter;
+   GstPlanarAudioAdapter *padapter;
+-  webrtc::AudioProcessing * apm;
+  webrtc::AudioProcessing *apm;
+ 
+   /* Protected by the object lock */
+   gchar *probe_name;
+@@ -257,21 +255,15 @@ struct _GstWebrtcDsp
+   /* Properties */
+   gboolean high_pass_filter;
+   gboolean echo_cancel;
+-  webrtc::EchoCancellation::SuppressionLevel echo_suppression_level;
+   gboolean noise_suppression;
+-  webrtc::NoiseSuppression::Level noise_suppression_level;
+  webrtc::AudioProcessing::Config::NoiseSuppression::Level noise_suppression_level;
+   gboolean gain_control;
+-  gboolean experimental_agc;
+-  gboolean extended_filter;
+-  gboolean delay_agnostic;
+   gint target_level_dbfs;
+   gint compression_gain_db;
+   gint startup_min_volume;
+   gboolean limiter;
+-  webrtc::GainControl::Mode gain_control_mode;
+  webrtc::AudioProcessing::Config::GainController1::Mode gain_control_mode;
+   gboolean voice_detection;
+-  gint voice_detection_frame_size_ms;
+-  webrtc::VoiceDetection::Likelihood voice_detection_likelihood;
+ };
+ 
+ G_DEFINE_TYPE_WITH_CODE (GstWebrtcDsp, gst_webrtc_dsp, GST_TYPE_AUDIO_FILTER,
+@@ -376,9 +368,9 @@ gst_webrtc_dsp_analyze_reverse_stream (GstWebrtcDsp * self,
+     GstClockTime rec_time)
+ {
+   GstWebrtcEchoProbe *probe = NULL;
+-  webrtc::AudioProcessing * apm;
+-  webrtc::AudioFrame frame;
+  webrtc::AudioProcessing *apm;
+   GstBuffer *buf = NULL;
+  GstAudioBuffer abuf;
+   GstFlowReturn ret = GST_FLOW_OK;
+   gint err, delay;
+ 
+@@ -391,48 +383,44 @@ gst_webrtc_dsp_analyze_reverse_stream (GstWebrtcDsp * self,
+   if (!probe)
+     return GST_FLOW_OK;
+ 
+  webrtc::StreamConfig config (probe->info.rate, probe->info.channels,
+      false);
+   apm = self->apm;
+ 
+-  if (self->delay_agnostic)
+-    rec_time = GST_CLOCK_TIME_NONE;
+-
+-again:
+-  delay = gst_webrtc_echo_probe_read (probe, rec_time, (gpointer) &frame, &buf);
+  delay = gst_webrtc_echo_probe_read (probe, rec_time, &buf);
+   apm->set_stream_delay_ms (delay);
+ 
+  g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR);
+
+   if (delay < 0)
+     goto done;
+ 
+-  if (frame.sample_rate_hz_ != self->info.rate) {
+  if (probe->info.rate != self->info.rate) {
+     GST_ELEMENT_ERROR (self, STREAM, FORMAT,
+         ("Echo Probe has rate %i , while the DSP is running at rate %i,"
+          " use a caps filter to ensure those are the same.",
+-         frame.sample_rate_hz_, self->info.rate), (NULL));
+         probe->info.rate, self->info.rate), (NULL));
+     ret = GST_FLOW_ERROR;
+     goto done;
+   }
+ 
+-  if (buf) {
+-    webrtc::StreamConfig config (frame.sample_rate_hz_, frame.num_channels_,
+-        false);
+-    GstAudioBuffer abuf;
+-    float * const * data;
+  gst_audio_buffer_map (&abuf, &self->info, buf, GST_MAP_READWRITE);
+
+  if (probe->interleaved) {
+    int16_t * const data = (int16_t * const) abuf.planes[0];
+ 
+-    gst_audio_buffer_map (&abuf, &self->info, buf, GST_MAP_READWRITE);
+-    data = (float * const *) abuf.planes;
+     if ((err = apm->ProcessReverseStream (data, config, config, data)) < 0)
+       GST_WARNING_OBJECT (self, "Reverse stream analyses failed: %s.",
+           webrtc_error_to_string (err));
+-    gst_audio_buffer_unmap (&abuf);
+-    gst_buffer_replace (&buf, NULL);
+   } else {
+-    if ((err = apm->AnalyzeReverseStream (&frame)) < 0)
+    float * const * data = (float * const *) abuf.planes;
+
+    if ((err = apm->ProcessReverseStream (data, config, config, data)) < 0)
+       GST_WARNING_OBJECT (self, "Reverse stream analyses failed: %s.",
+           webrtc_error_to_string (err));
+   }
+ 
+-  if (self->delay_agnostic)
+-      goto again;
+  gst_audio_buffer_unmap (&abuf);
+ 
+ done:
+   gst_object_unref (probe);
+@@ -443,16 +431,14 @@ done:
+ 
+ static void
+ gst_webrtc_vad_post_activity (GstWebrtcDsp *self, GstBuffer *buffer,
+-    gboolean stream_has_voice)
+    gboolean stream_has_voice, guint8 level)
+ {
+   GstClockTime timestamp = GST_BUFFER_PTS (buffer);
+   GstBaseTransform *trans = GST_BASE_TRANSFORM_CAST (self);
+   GstStructure *s;
+   GstClockTime stream_time;
+   GstAudioLevelMeta *meta;
+-  guint8 level;
+ 
+-  level = self->apm->level_estimator ()->RMS ();
+   meta = gst_buffer_get_audio_level_meta (buffer);
+   if (meta) {
+     meta->voice_activity = stream_has_voice;
+@@ -481,6 +467,7 @@ gst_webrtc_dsp_process_stream (GstWebrtcDsp * self,
+ {
+   GstAudioBuffer abuf;
+   webrtc::AudioProcessing * apm = self->apm;
+  webrtc::StreamConfig config (self->info.rate, self->info.channels, false);
+   gint err;
+ 
+   if (!gst_audio_buffer_map (&abuf, &self->info, buffer,
+@@ -490,19 +477,10 @@ gst_webrtc_dsp_process_stream (GstWebrtcDsp * self,
+   }
+ 
+   if (self->interleaved) {
+-    webrtc::AudioFrame frame;
+-    frame.num_channels_ = self->info.channels;
+-    frame.sample_rate_hz_ = self->info.rate;
+-    frame.samples_per_channel_ = self->period_samples;
+-
+-    memcpy (frame.data_, abuf.planes[0], self->period_size);
+-    err = apm->ProcessStream (&frame);
+-    if (err >= 0)
+-      memcpy (abuf.planes[0], frame.data_, self->period_size);
+    int16_t * const data = (int16_t * const) abuf.planes[0];
+    err = apm->ProcessStream (data, config, config, data);
+   } else {
+     float * const * data = (float * const *) abuf.planes;
+-    webrtc::StreamConfig config (self->info.rate, self->info.channels, false);
+-
+     err = apm->ProcessStream (data, config, config, data);
+   }
+ 
+@@ -511,10 +489,13 @@ gst_webrtc_dsp_process_stream (GstWebrtcDsp * self,
+         webrtc_error_to_string (err));
+   } else {
+     if (self->voice_detection) {
+-      gboolean stream_has_voice = apm->voice_detection ()->stream_has_voice ();
+      webrtc::AudioProcessingStats stats = apm->GetStatistics ();
+      gboolean stream_has_voice = stats.voice_detected && *stats.voice_detected;
+      // The meta takes the value as -dbov, so we negate
+      guint8 level = stats.output_rms_dbfs ? (guint8) -(*stats.output_rms_dbfs) : 127;
+ 
+       if (stream_has_voice != self->stream_has_voice)
+-        gst_webrtc_vad_post_activity (self, buffer, stream_has_voice);
+        gst_webrtc_vad_post_activity (self, buffer, stream_has_voice, level);
+ 
+       self->stream_has_voice = stream_has_voice;
+     }
+@@ -583,21 +564,9 @@ static gboolean
+ gst_webrtc_dsp_start (GstBaseTransform * btrans)
+ {
+   GstWebrtcDsp *self = GST_WEBRTC_DSP (btrans);
+-  webrtc::Config config;
+ 
+   GST_OBJECT_LOCK (self);
+ 
+-  config.Set < webrtc::ExtendedFilter >
+-      (new webrtc::ExtendedFilter (self->extended_filter));
+-  config.Set < webrtc::ExperimentalAgc >
+-      (new webrtc::ExperimentalAgc (self->experimental_agc, self->startup_min_volume));
+-  config.Set < webrtc::DelayAgnostic >
+-      (new webrtc::DelayAgnostic (self->delay_agnostic));
+-
+-  /* TODO Intelligibility enhancer, Beamforming, etc. */
+-
+-  self->apm = webrtc::AudioProcessing::Create (config);
+-
+   if (self->echo_cancel) {
+     self->probe = gst_webrtc_acquire_echo_probe (self->probe_name);
+ 
+@@ -618,10 +587,8 @@ static gboolean
+ gst_webrtc_dsp_setup (GstAudioFilter * filter, const GstAudioInfo * info)
+ {
+   GstWebrtcDsp *self = GST_WEBRTC_DSP (filter);
+-  webrtc::AudioProcessing * apm;
+-  webrtc::ProcessingConfig pconfig;
+  webrtc::AudioProcessing::Config config;
+   GstAudioInfo probe_info = *info;
+-  gint err = 0;
+ 
+   GST_LOG_OBJECT (self, "setting format to %s with %i Hz and %i channels",
+       info->finfo->description, info->rate, info->channels);
+@@ -633,7 +600,7 @@ gst_webrtc_dsp_setup (GstAudioFilter * filter, const GstAudioInfo * info)
+ 
+   self->info = *info;
+   self->interleaved = (info->layout == GST_AUDIO_LAYOUT_INTERLEAVED);
+-  apm = self->apm;
+  self->apm = webrtc::AudioProcessingBuilder().Create();
+ 
+   if (!self->interleaved)
+     gst_planar_audio_adapter_configure (self->padapter, info);
+@@ -642,8 +609,7 @@ gst_webrtc_dsp_setup (GstAudioFilter * filter, const GstAudioInfo * info)
+   self->period_samples = info->rate / 100;
+   self->period_size = self->period_samples * info->bpf;
+ 
+-  if (self->interleaved &&
+-      (webrtc::AudioFrame::kMaxDataSizeSamples * 2) < self->period_size)
+  if (self->interleaved && (self->period_size > MAX_DATA_SIZE_SAMPLES * 2))
+     goto period_too_big;
+ 
+   if (self->probe) {
+@@ -658,40 +624,31 @@ gst_webrtc_dsp_setup (GstAudioFilter * filter, const GstAudioInfo * info)
+     GST_WEBRTC_ECHO_PROBE_UNLOCK (self->probe);
+   }
+ 
+-  /* input stream */
+-  pconfig.streams[webrtc::ProcessingConfig::kInputStream] =
+-      webrtc::StreamConfig (info->rate, info->channels, false);
+-  /* output stream */
+-  pconfig.streams[webrtc::ProcessingConfig::kOutputStream] =
+-      webrtc::StreamConfig (info->rate, info->channels, false);
+-  /* reverse input stream */
+-  pconfig.streams[webrtc::ProcessingConfig::kReverseInputStream] =
+-      webrtc::StreamConfig (probe_info.rate, probe_info.channels, false);
+-  /* reverse output stream */
+-  pconfig.streams[webrtc::ProcessingConfig::kReverseOutputStream] =
+-      webrtc::StreamConfig (probe_info.rate, probe_info.channels, false);
+-
+-  if ((err = apm->Initialize (pconfig)) < 0)
+-    goto initialize_failed;
+-
+   /* Setup Filters */
+  // TODO: expose pre_amplifier
+
+   if (self->high_pass_filter) {
+     GST_DEBUG_OBJECT (self, "Enabling High Pass filter");
+-    apm->high_pass_filter ()->Enable (true);
+    config.high_pass_filter.enabled = true;
+   }
+ 
+   if (self->echo_cancel) {
+     GST_DEBUG_OBJECT (self, "Enabling Echo Cancellation");
+-    apm->echo_cancellation ()->enable_drift_compensation (false);
+-    apm->echo_cancellation ()
+-        ->set_suppression_level (self->echo_suppression_level);
+-    apm->echo_cancellation ()->Enable (true);
+    config.echo_canceller.enabled = true;
+   }
+ 
+   if (self->noise_suppression) {
+     GST_DEBUG_OBJECT (self, "Enabling Noise Suppression");
+-    apm->noise_suppression ()->set_level (self->noise_suppression_level);
+-    apm->noise_suppression ()->Enable (true);
+    config.noise_suppression.enabled = true;
+    config.noise_suppression.level = self->noise_suppression_level;
+  }
+
+  // TODO: expose transient suppression
+
+  if (self->voice_detection) {
+    GST_DEBUG_OBJECT (self, "Enabling Voice Activity Detection");
+    config.voice_detection.enabled = true;
+    self->stream_has_voice = FALSE;
+   }
+ 
+   if (self->gain_control) {
+@@ -706,30 +663,17 @@ gst_webrtc_dsp_setup (GstAudioFilter * filter, const GstAudioInfo * info)
+ 
+     g_type_class_unref (mode_class);
+ 
+-    apm->gain_control ()->set_mode (self->gain_control_mode);
+-    apm->gain_control ()->set_target_level_dbfs (self->target_level_dbfs);
+-    apm->gain_control ()->set_compression_gain_db (self->compression_gain_db);
+-    apm->gain_control ()->enable_limiter (self->limiter);
+-    apm->gain_control ()->Enable (true);
+    config.gain_controller1.enabled = true;
+    config.gain_controller1.target_level_dbfs = self->target_level_dbfs;
+    config.gain_controller1.compression_gain_db = self->compression_gain_db;
+    config.gain_controller1.enable_limiter = self->limiter;
+    config.level_estimation.enabled = true;
+   }
+ 
+-  if (self->voice_detection) {
+-    GEnumClass *likelihood_class = (GEnumClass *)
+-        g_type_class_ref (GST_TYPE_WEBRTC_VOICE_DETECTION_LIKELIHOOD);
+-    GST_DEBUG_OBJECT (self, "Enabling Voice Activity Detection, frame size "
+-      "%d milliseconds, likelihood: %s", self->voice_detection_frame_size_ms,
+-      g_enum_get_value (likelihood_class,
+-          self->voice_detection_likelihood)->value_name);
+-    g_type_class_unref (likelihood_class);
+  // TODO: expose gain controller 2
+  // TODO: expose residual echo detector
+ 
+-    self->stream_has_voice = FALSE;
+-
+-    apm->voice_detection ()->Enable (true);
+-    apm->voice_detection ()->set_likelihood (self->voice_detection_likelihood);
+-    apm->voice_detection ()->set_frame_size_ms (
+-        self->voice_detection_frame_size_ms);
+-    apm->level_estimator ()->Enable (true);
+-  }
+  self->apm->ApplyConfig (config);
+ 
+   GST_OBJECT_UNLOCK (self);
+ 
+@@ -738,9 +682,9 @@ gst_webrtc_dsp_setup (GstAudioFilter * filter, const GstAudioInfo * info)
+ period_too_big:
+   GST_OBJECT_UNLOCK (self);
+   GST_WARNING_OBJECT (self, "webrtcdsp format produce too big period "
+-      "(maximum is %" G_GSIZE_FORMAT " samples and we have %u samples), "
+      "(maximum is %d samples and we have %u samples), "
+       "reduce the number of channels or the rate.",
+-      webrtc::AudioFrame::kMaxDataSizeSamples, self->period_size / 2);
+      MAX_DATA_SIZE_SAMPLES, self->period_size / 2);
+   return FALSE;
+ 
+ probe_has_wrong_rate:
+@@ -751,14 +695,6 @@ probe_has_wrong_rate:
+           " use a caps filter to ensure those are the same.",
+           probe_info.rate, info->rate), (NULL));
+   return FALSE;
+-
+-initialize_failed:
+-  GST_OBJECT_UNLOCK (self);
+-  GST_ELEMENT_ERROR (self, LIBRARY, INIT,
+-      ("Failed to initialize WebRTC Audio Processing library"),
+-      ("webrtc::AudioProcessing::Initialize() failed: %s",
+-          webrtc_error_to_string (err)));
+-  return FALSE;
+ }
+ 
+ static gboolean
+@@ -803,8 +739,6 @@ gst_webrtc_dsp_set_property (GObject * object,
+       self->echo_cancel = g_value_get_boolean (value);
+       break;
+     case PROP_ECHO_SUPPRESSION_LEVEL:
+-      self->echo_suppression_level =
+-          (GstWebrtcEchoSuppressionLevel) g_value_get_enum (value);
+       break;
+     case PROP_NOISE_SUPPRESSION:
+       self->noise_suppression = g_value_get_boolean (value);
+@@ -817,13 +751,10 @@ gst_webrtc_dsp_set_property (GObject * object,
+       self->gain_control = g_value_get_boolean (value);
+       break;
+     case PROP_EXPERIMENTAL_AGC:
+-      self->experimental_agc = g_value_get_boolean (value);
+       break;
+     case PROP_EXTENDED_FILTER:
+-      self->extended_filter = g_value_get_boolean (value);
+       break;
+     case PROP_DELAY_AGNOSTIC:
+-      self->delay_agnostic = g_value_get_boolean (value);
+       break;
+     case PROP_TARGET_LEVEL_DBFS:
+       self->target_level_dbfs = g_value_get_int (value);
+@@ -845,11 +776,8 @@ gst_webrtc_dsp_set_property (GObject * object,
+       self->voice_detection = g_value_get_boolean (value);
+       break;
+     case PROP_VOICE_DETECTION_FRAME_SIZE_MS:
+-      self->voice_detection_frame_size_ms = g_value_get_int (value);
+       break;
+     case PROP_VOICE_DETECTION_LIKELIHOOD:
+-      self->voice_detection_likelihood =
+-          (GstWebrtcVoiceDetectionLikelihood) g_value_get_enum (value);
+       break;
+     default:
+       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+@@ -876,7 +804,7 @@ gst_webrtc_dsp_get_property (GObject * object,
+       g_value_set_boolean (value, self->echo_cancel);
+       break;
+     case PROP_ECHO_SUPPRESSION_LEVEL:
+-      g_value_set_enum (value, self->echo_suppression_level);
+      g_value_set_enum (value, (GstWebrtcEchoSuppressionLevel) 2);
+       break;
+     case PROP_NOISE_SUPPRESSION:
+       g_value_set_boolean (value, self->noise_suppression);
+@@ -888,13 +816,13 @@ gst_webrtc_dsp_get_property (GObject * object,
+       g_value_set_boolean (value, self->gain_control);
+       break;
+     case PROP_EXPERIMENTAL_AGC:
+-      g_value_set_boolean (value, self->experimental_agc);
+      g_value_set_boolean (value, false);
+       break;
+     case PROP_EXTENDED_FILTER:
+-      g_value_set_boolean (value, self->extended_filter);
+      g_value_set_boolean (value, false);
+       break;
+     case PROP_DELAY_AGNOSTIC:
+-      g_value_set_boolean (value, self->delay_agnostic);
+      g_value_set_boolean (value, false);
+       break;
+     case PROP_TARGET_LEVEL_DBFS:
+       g_value_set_int (value, self->target_level_dbfs);
+@@ -915,10 +843,10 @@ gst_webrtc_dsp_get_property (GObject * object,
+       g_value_set_boolean (value, self->voice_detection);
+       break;
+     case PROP_VOICE_DETECTION_FRAME_SIZE_MS:
+-      g_value_set_int (value, self->voice_detection_frame_size_ms);
+      g_value_set_int (value, 0);
+       break;
+     case PROP_VOICE_DETECTION_LIKELIHOOD:
+-      g_value_set_enum (value, self->voice_detection_likelihood);
+      g_value_set_enum (value, 2);
+       break;
+     default:
+       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+@@ -1005,13 +933,13 @@ gst_webrtc_dsp_class_init (GstWebrtcDspClass * klass)
+ 
+   g_object_class_install_property (gobject_class,
+       PROP_ECHO_SUPPRESSION_LEVEL,
+-      g_param_spec_enum ("echo-suppression-level", "Echo Suppression Level",
+      g_param_spec_enum ("echo-suppression-level",
+          "Echo Suppression Level (does nothing)",
+           "Controls the aggressiveness of the suppressor. A higher level "
+           "trades off double-talk performance for increased echo suppression.",
+-          GST_TYPE_WEBRTC_ECHO_SUPPRESSION_LEVEL,
+-          webrtc::EchoCancellation::kModerateSuppression,
+          GST_TYPE_WEBRTC_ECHO_SUPPRESSION_LEVEL, 2,
+           (GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
+-              G_PARAM_CONSTRUCT)));
+              G_PARAM_CONSTRUCT | G_PARAM_DEPRECATED)));
+ 
+   g_object_class_install_property (gobject_class,
+       PROP_NOISE_SUPPRESSION,
+@@ -1026,7 +954,7 @@ gst_webrtc_dsp_class_init (GstWebrtcDspClass * klass)
+           "Controls the aggressiveness of the suppression. Increasing the "
+           "level will reduce the noise level at the expense of a higher "
+           "speech distortion.", GST_TYPE_WEBRTC_NOISE_SUPPRESSION_LEVEL,
+-          webrtc::EchoCancellation::kModerateSuppression,
+          webrtc::AudioProcessing::Config::NoiseSuppression::Level::kModerate,
+           (GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
+               G_PARAM_CONSTRUCT)));
+ 
+@@ -1039,24 +967,26 @@ gst_webrtc_dsp_class_init (GstWebrtcDspClass * klass)
+ 
+   g_object_class_install_property (gobject_class,
+       PROP_EXPERIMENTAL_AGC,
+-      g_param_spec_boolean ("experimental-agc", "Experimental AGC",
+      g_param_spec_boolean ("experimental-agc",
+          "Experimental AGC (does nothing)",
+           "Enable or disable experimental automatic gain control.",
+           FALSE, (GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
+-              G_PARAM_CONSTRUCT)));
+              G_PARAM_CONSTRUCT | G_PARAM_DEPRECATED)));
+ 
+   g_object_class_install_property (gobject_class,
+       PROP_EXTENDED_FILTER,
+       g_param_spec_boolean ("extended-filter", "Extended Filter",
+           "Enable or disable the extended filter.",
+           TRUE, (GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
+-              G_PARAM_CONSTRUCT)));
+              G_PARAM_CONSTRUCT | G_PARAM_DEPRECATED)));
+ 
+   g_object_class_install_property (gobject_class,
+       PROP_DELAY_AGNOSTIC,
+-      g_param_spec_boolean ("delay-agnostic", "Delay Agnostic",
+      g_param_spec_boolean ("delay-agnostic",
+          "Delay agnostic mode (does nothing)",
+           "Enable or disable the delay agnostic mode.",
+           FALSE, (GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
+-              G_PARAM_CONSTRUCT)));
+              G_PARAM_CONSTRUCT | G_PARAM_DEPRECATED)));
+ 
+   g_object_class_install_property (gobject_class,
+       PROP_TARGET_LEVEL_DBFS,
+@@ -1111,24 +1041,23 @@ gst_webrtc_dsp_class_init (GstWebrtcDspClass * klass)
+   g_object_class_install_property (gobject_class,
+       PROP_VOICE_DETECTION_FRAME_SIZE_MS,
+       g_param_spec_int ("voice-detection-frame-size-ms",
+-          "Voice Detection Frame Size Milliseconds",
+          "Voice detection frame size in milliseconds (does nothing)",
+           "Sets the |size| of the frames in ms on which the VAD will operate. "
+           "Larger frames will improve detection accuracy, but reduce the "
+           "frequency of updates",
+           10, 30, DEFAULT_VOICE_DETECTION_FRAME_SIZE_MS,
+           (GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
+-              G_PARAM_CONSTRUCT)));
+              G_PARAM_CONSTRUCT | G_PARAM_DEPRECATED)));
+ 
+   g_object_class_install_property (gobject_class,
+       PROP_VOICE_DETECTION_LIKELIHOOD,
+       g_param_spec_enum ("voice-detection-likelihood",
+-          "Voice Detection Likelihood",
+          "Voice detection likelihood (does nothing)",
+           "Specifies the likelihood that a frame will be declared to contain "
+           "voice.",
+-          GST_TYPE_WEBRTC_VOICE_DETECTION_LIKELIHOOD,
+-          DEFAULT_VOICE_DETECTION_LIKELIHOOD,
+          GST_TYPE_WEBRTC_VOICE_DETECTION_LIKELIHOOD, 2,
+           (GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
+-              G_PARAM_CONSTRUCT)));
+              G_PARAM_CONSTRUCT | G_PARAM_DEPRECATED)));
+ 
+   gst_type_mark_as_plugin_api (GST_TYPE_WEBRTC_GAIN_CONTROL_MODE, (GstPluginAPIFlags) 0);
+   gst_type_mark_as_plugin_api (GST_TYPE_WEBRTC_NOISE_SUPPRESSION_LEVEL, (GstPluginAPIFlags) 0);
+diff --git a/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.cpp b/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.cpp
+index acdb3d8a7d5..8e8ca064c48 100644
+--- a/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.cpp
+++ b/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.cpp
+@@ -33,7 +33,8 @@
+ 
+ #include "gstwebrtcechoprobe.h"
+ 
+-#include <webrtc/modules/interface/module_common_types.h>
+#include <modules/audio_processing/include/audio_processing.h>
+
+ #include <gst/audio/audio.h>
+ 
+ GST_DEBUG_CATEGORY_EXTERN (webrtc_dsp_debug);
+@@ -102,7 +103,7 @@ gst_webrtc_echo_probe_setup (GstAudioFilter * filter, const GstAudioInfo * info)
+   self->period_size = self->period_samples * info->bpf;
+ 
+   if (self->interleaved &&
+-      (webrtc::AudioFrame::kMaxDataSizeSamples * 2) < self->period_size)
+      (MAX_DATA_SIZE_SAMPLES * 2) < self->period_size)
+     goto period_too_big;
+ 
+   GST_WEBRTC_ECHO_PROBE_UNLOCK (self);
+@@ -112,9 +113,9 @@ gst_webrtc_echo_probe_setup (GstAudioFilter * filter, const GstAudioInfo * info)
+ period_too_big:
+   GST_WEBRTC_ECHO_PROBE_UNLOCK (self);
+   GST_WARNING_OBJECT (self, "webrtcdsp format produce too big period "
+-      "(maximum is %" G_GSIZE_FORMAT " samples and we have %u samples), "
+      "(maximum is %d samples and we have %u samples), "
+       "reduce the number of channels or the rate.",
+-      webrtc::AudioFrame::kMaxDataSizeSamples, self->period_size / 2);
+      MAX_DATA_SIZE_SAMPLES, self->period_size / 2);
+   return FALSE;
+ }
+ 
+@@ -303,18 +304,20 @@ gst_webrtc_release_echo_probe (GstWebrtcEchoProbe * probe)
+ 
+ gint
+ gst_webrtc_echo_probe_read (GstWebrtcEchoProbe * self, GstClockTime rec_time,
+-    gpointer _frame, GstBuffer ** buf)
+    GstBuffer ** buf)
+ {
+-  webrtc::AudioFrame * frame = (webrtc::AudioFrame *) _frame;
+   GstClockTimeDiff diff;
+-  gsize avail, skip, offset, size;
+  gsize avail, skip, offset, size = 0;
+   gint delay = -1;
+ 
+   GST_WEBRTC_ECHO_PROBE_LOCK (self);
+ 
+  /* We always return a buffer -- if don't have data (size == 0), we generate a
+   * silence buffer */
+
+   if (!GST_CLOCK_TIME_IS_VALID (self->latency) ||
+       !GST_AUDIO_INFO_IS_VALID (&self->info))
+-    goto done;
+    goto copy;
+ 
+   if (self->interleaved)
+     avail = gst_adapter_available (self->adapter) / self->info.bpf;
+@@ -324,7 +327,7 @@ gst_webrtc_echo_probe_read (GstWebrtcEchoProbe * self, GstClockTime rec_time,
+   /* In delay agnostic mode, just return 10ms of data */
+   if (!GST_CLOCK_TIME_IS_VALID (rec_time)) {
+     if (avail < self->period_samples)
+-      goto done;
+      goto copy;
+ 
+     size = self->period_samples;
+     skip = 0;
+@@ -371,23 +374,51 @@ gst_webrtc_echo_probe_read (GstWebrtcEchoProbe * self, GstClockTime rec_time,
+   size = MIN (avail - offset, self->period_samples - skip);
+ 
+ copy:
+-  if (self->interleaved) {
+-    skip *= self->info.bpf;
+-    offset *= self->info.bpf;
+-    size *= self->info.bpf;
+-
+-    if (size < self->period_size)
+-      memset (frame->data_, 0, self->period_size);
+-
+-    if (size) {
+-      gst_adapter_copy (self->adapter, (guint8 *) frame->data_ + skip,
+-          offset, size);
+-      gst_adapter_flush (self->adapter, offset + size);
+-    }
+  if (!size) {
+    /* No data, provide a period's worth of silence */
+    *buf = gst_buffer_new_allocate (NULL, self->period_size, NULL);
+    gst_buffer_memset (*buf, 0, 0, self->period_size);
+    gst_buffer_add_audio_meta (*buf, &self->info, self->period_samples,
+        NULL);
+   } else {
+    /* We have some actual data, pop period_samples' worth if have it, else pad
+     * with silence and provide what we do have */
+     GstBuffer *ret, *taken, *tmp;
+ 
+-    if (size) {
+    if (self->interleaved) {
+      skip *= self->info.bpf;
+      offset *= self->info.bpf;
+      size *= self->info.bpf;
+
+      gst_adapter_flush (self->adapter, offset);
+
+      /* we need to fill silence at the beginning and/or the end of the
+       * buffer in order to have period_samples in the buffer */
+      if (size < self->period_size) {
+        gsize padding = self->period_size - (skip + size);
+
+        taken = gst_adapter_take_buffer (self->adapter, size);
+        ret = gst_buffer_new ();
+
+        /* need some silence at the beginning */
+        if (skip) {
+          tmp = gst_buffer_new_allocate (NULL, skip, NULL);
+          gst_buffer_memset (tmp, 0, 0, skip);
+          ret = gst_buffer_append (ret, tmp);
+        }
+
+        ret = gst_buffer_append (ret, taken);
+
+        /* need some silence at the end */
+        if (padding) {
+          tmp = gst_buffer_new_allocate (NULL, padding, NULL);
+          gst_buffer_memset (tmp, 0, 0, padding);
+          ret = gst_buffer_append (ret, tmp);
+        }
+      } else {
+        ret = gst_adapter_take_buffer (self->adapter, size);
+      }
+    } else {
+       gst_planar_audio_adapter_flush (self->padapter, offset);
+ 
+       /* we need to fill silence at the beginning and/or the end of each
+@@ -430,23 +461,13 @@ copy:
+         ret = gst_planar_audio_adapter_take_buffer (self->padapter, size,
+           GST_MAP_READWRITE);
+       }
+-    } else {
+-      ret = gst_buffer_new_allocate (NULL, self->period_size, NULL);
+-      gst_buffer_memset (ret, 0, 0, self->period_size);
+-      gst_buffer_add_audio_meta (ret, &self->info, self->period_samples,
+-          NULL);
+     }
+ 
+     *buf = ret;
+   }
+ 
+-  frame->num_channels_ = self->info.channels;
+-  frame->sample_rate_hz_ = self->info.rate;
+-  frame->samples_per_channel_ = self->period_samples;
+-
+   delay = self->delay;
+ 
+-done:
+   GST_WEBRTC_ECHO_PROBE_UNLOCK (self);
+ 
+   return delay;
+diff --git a/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.h b/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.h
+index 36fd34f1794..488c0e958f3 100644
+--- a/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.h
+++ b/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.h
+@@ -45,6 +45,12 @@ G_BEGIN_DECLS
+ #define GST_WEBRTC_ECHO_PROBE_LOCK(obj) g_mutex_lock (&GST_WEBRTC_ECHO_PROBE (obj)->lock)
+ #define GST_WEBRTC_ECHO_PROBE_UNLOCK(obj) g_mutex_unlock (&GST_WEBRTC_ECHO_PROBE (obj)->lock)
+ 
+/* From the webrtc audio_frame.h definition of kMaxDataSizeSamples:
+ * Stereo, 32 kHz, 120 ms (2 * 32 * 120)
+ * Stereo, 192 kHz, 20 ms (2 * 192 * 20)
+ */
+#define MAX_DATA_SIZE_SAMPLES 7680
+
+ typedef struct _GstWebrtcEchoProbe GstWebrtcEchoProbe;
+ typedef struct _GstWebrtcEchoProbeClass GstWebrtcEchoProbeClass;
+ 
+@@ -71,6 +77,7 @@ struct _GstWebrtcEchoProbe
+   GstClockTime latency;
+   gint delay;
+   gboolean interleaved;
+  gint extra_delay;
+ 
+   GstSegment segment;
+   GstAdapter *adapter;
+@@ -92,7 +99,7 @@ GST_ELEMENT_REGISTER_DECLARE (webrtcechoprobe);
+ GstWebrtcEchoProbe *gst_webrtc_acquire_echo_probe (const gchar * name);
+ void gst_webrtc_release_echo_probe (GstWebrtcEchoProbe * probe);
+ gint gst_webrtc_echo_probe_read (GstWebrtcEchoProbe * self,
+-    GstClockTime rec_time, gpointer frame, GstBuffer ** buf);
+    GstClockTime rec_time, GstBuffer ** buf);
+ 
+ G_END_DECLS
+ #endif /* __GST_WEBRTC_ECHO_PROBE_H__ */
+diff --git a/subprojects/gst-plugins-bad/ext/webrtcdsp/meson.build b/subprojects/gst-plugins-bad/ext/webrtcdsp/meson.build
+index 5aeae69a44d..09565e27c73 100644
+--- a/subprojects/gst-plugins-bad/ext/webrtcdsp/meson.build
+++ b/subprojects/gst-plugins-bad/ext/webrtcdsp/meson.build
+@@ -4,7 +4,7 @@ webrtc_sources = [
+   'gstwebrtcdspplugin.cpp'
+ ]
+ 
+-webrtc_dep = dependency('webrtc-audio-processing', version : ['>= 0.2', '< 0.4'],
+webrtc_dep = dependency('webrtc-audio-processing-1', version : ['>= 1.0'],
+                         required : get_option('webrtcdsp'))
+ 
+ if not gnustl_dep.found() and get_option('webrtcdsp').enabled()
+@@ -20,7 +20,7 @@ if webrtc_dep.found() and gnustl_dep.found()
+     dependencies : [gstbase_dep, gstaudio_dep, gstbadaudio_dep, webrtc_dep, gnustl_dep],
+     install : true,
+     install_dir : plugins_install_dir,
+-    override_options : ['cpp_std=c++11'],
+    override_options : ['cpp_std=c++17'],
+   )
+   plugins += [gstwebrtcdsp]
+ endif
+-- 
+GitLab
+
+
+#From 37aab17be305b8033e682276ad9d4ea2d0ab9ee2 Mon Sep 17 00:00:00 2001
+#From: Nirbheek Chauhan <nirbheek@centricular.com>
+#Date: Wed, 31 May 2023 17:51:38 +0530
+#Subject: [PATCH 2/2] meson: Update webrtc-audio-processing wrap to 1.1
+#
+#Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2943>
+#---
+# subprojects/webrtc-audio-processing.wrap | 8 ++++----
+# 1 file changed, 4 insertions(+), 4 deletions(-)
+#
+#diff --git a/subprojects/webrtc-audio-processing.wrap b/subprojects/webrtc-audio-processing.wrap
+#index 11e9390bc53..bba7dd0b516 100644
+#--- a/subprojects/webrtc-audio-processing.wrap
+#+++ b/subprojects/webrtc-audio-processing.wrap
+#@@ -1,8 +1,8 @@
+# [wrap-git]
+#-directory=webrtc-audio-processing
+#-url=https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing.git
+#-push-url=git@gitlab.freedesktop.org:pulseaudio/webrtc-audio-processing.git
+#-revision=v1.0
+#+directory = webrtc-audio-processing
+#+url = https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing.git
+#+push-url = git@gitlab.freedesktop.org:pulseaudio/webrtc-audio-processing.git
+#+revision = v1.1
+# 
+# [provide]
+# dependency_names = webrtc-audio-coding-1, webrtc-audio-processing-1
+#-- 
+#GitLab
+#
--- a/baselibs.conf
+++ b/baselibs.conf
@ -0,0 +1,24 @@
+gstreamer-plugins-bad
+gstreamer-plugins-bad-chromaprint
+gstreamer-plugins-bad-fluidsynth
+gstreamer-plugins-bad-orig-addon
+libgstadaptivedemux-1_0-0
+libgstbadaudio-1_0-0
+libgstbasecamerabinsrc-1_0-0
+libgstcodecparsers-1_0-0
+libgstcodecs-1_0-0
+libgstcuda-1_0-0
+libgstinsertbin-1_0-0
+libgstisoff-1_0-0
+libgstmpegts-1_0-0
+libgstphotography-1_0-0
+libgstplay-1_0-0
+libgstplayer-1_0-0
+libgstsctp-1_0-0
+libgsttranscoder-1_0-0
+libgsturidownloader-1_0-0
+libgstva-1_0-0
+libgstvulkan-1_0-0
+libgstwayland-1_0-0
+libgstwebrtc-1_0-0
+libgstwebrtcnice-1_0-0
--- a/fix-build-with-srt-1.3.4.patch
+++ b/fix-build-with-srt-1.3.4.patch
@ -0,0 +1,27 @@
+From: Antonio Larrosa <alarrosa@suse.com>
+Subject: Fix build with srt 1.3.4 since gstreamer expects 1.4.x
+
+SRTO_STRICTENC was just renamed to SRTO_ENFORCEDENCRYPTION in 1.4, so revert that.
+SRTO_PACKETFILTER was introduced in 1.4.0 so we can't support urls
+specifying a value for it in the url.
+
+Index: gst-plugins-bad-1.22.1/ext/srt/gstsrtobject.c
+===================================================================
+--- gst-plugins-bad-1.22.1.orig/ext/srt/gstsrtobject.c
+++ gst-plugins-bad-1.22.1/ext/srt/gstsrtobject.c
+@@ -177,13 +177,13 @@ SrtOption srt_options[] = {
+   {"transtype", SRTO_TRANSTYPE, G_TYPE_INT},
+   {"kmrefreshrate", SRTO_KMREFRESHRATE, G_TYPE_INT},
+   {"kmpreannounce", SRTO_KMPREANNOUNCE, G_TYPE_INT},
+-  {"enforcedencryption", SRTO_ENFORCEDENCRYPTION, G_TYPE_BOOLEAN},
+  {"enforcedencryption", SRTO_STRICTENC, G_TYPE_BOOLEAN},
+   {"ipv6only", SRTO_IPV6ONLY, G_TYPE_INT},
+   {"peeridletimeo", SRTO_PEERIDLETIMEO, G_TYPE_INT},
+ #if SRT_VERSION_VALUE >= 0x10402
+   {"bindtodevice", SRTO_BINDTODEVICE, G_TYPE_STRING},
+ #endif
+-  {"packetfilter", SRTO_PACKETFILTER, G_TYPE_STRING},
+  //{"packetfilter", SRTO_PACKETFILTER, G_TYPE_STRING},
+   {"retransmitalgo", SRTO_RETRANSMITALGO, G_TYPE_INT},
+   {NULL}
+ };
--- a/gst-plugins-bad-1.22.9.tar.xz
+++ b/gst-plugins-bad-1.22.9.tar.xz
--- a/gstreamer-plugins-bad.appdata.xml
+++ b/gstreamer-plugins-bad.appdata.xml
@ -0,0 +1,39 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!-- Copyright 2013 Richard Hughes <richard@hughsie.com> -->
+<component type="codec">
+  <id>gstreamer-plugins-bad</id>
+  <metadata_license>CC0-1.0</metadata_license>
+  <name>GStreamer Multimedia Codecs - Extra</name>
+  <summary>Multimedia playback for AIFF, DVB, GSM, MIDI, MXF and Opus</summary>
+  <description>
+    <p>
+      This addon includes several additional codecs that are missing
+      something - perhaps a good code review, some documentation, a set of
+      tests, a real live maintainer, or some actual wide use.
+      However, they might be good enough to play your media files.
+    </p>
+    <p>
+      These codecs can be used to encode and decode media files where the
+      format is not patent encumbered.
+    </p>
+    <p>
+      A codec decodes audio and video for for playback or editing and is also
+      used for transmission or storage.
+      Different codecs are used in video-conferencing, streaming media and
+      video editing applications.
+    </p>
+  </description>
+  <keywords>
+    <keyword>AIFF</keyword>
+    <keyword>DVB</keyword>
+    <keyword>GSM</keyword>
+    <keyword>MIDI</keyword>
+    <keyword>MXF</keyword>
+    <keyword>Opus</keyword>
+  </keywords>
+  <url type="homepage">http://gstreamer.freedesktop.org/</url>
+  <url type="bugtracker">https://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer</url>
+  <url type="help">http://gstreamer.freedesktop.org/documentation/</url>
+  <url type="donation">http://www.gnome.org/friends/</url>
+  <update_contact><!-- upstream-contact_at_email.com --></update_contact>
+</component>
--- a/gstreamer-plugins-bad.changes
+++ b/gstreamer-plugins-bad.changes
--- a/gstreamer-plugins-bad.spec
+++ b/gstreamer-plugins-bad.spec
--- a/reduce-required-meson.patch
+++ b/reduce-required-meson.patch
@ -0,0 +1,12 @@
+Index: gst-plugins-bad-1.22.9/meson.build
+===================================================================
+--- gst-plugins-bad-1.22.9.orig/meson.build
+++ gst-plugins-bad-1.22.9/meson.build
+@@ -1,6 +1,6 @@
+ project('gst-plugins-bad', 'c', 'cpp',
+   version : '1.22.9',
+-  meson_version : '>= 0.62',
+  meson_version : '>= 0.61',
+   default_options : [ 'warning_level=1',
+                       'buildtype=debugoptimized' ])
+ 
--- a/spandsp3.patch
+++ b/spandsp3.patch
@ -0,0 +1,21 @@
+From: Jan Engelhardt <jengelh@inai.de>
+Date: 2022-04-10 13:08:44.803090340 +0200
+
+Make the build with spandsp-3.x succeed again.
+
+---
+ ext/spandsp/gstspanplc.c |    1 +
+ 1 file changed, 1 insertion(+)
+
+Index: gst-plugins-bad-1.20.1/ext/spandsp/gstspanplc.c
+===================================================================
+--- gst-plugins-bad-1.20.1.orig/ext/spandsp/gstspanplc.c
+++ gst-plugins-bad-1.20.1/ext/spandsp/gstspanplc.c
+@@ -36,6 +36,7 @@
+ #include "gstspanplc.h"
+ 
+ #include <gst/audio/audio.h>
+#include <spandsp/private/plc.h>
+ 
+ G_DEFINE_TYPE (GstSpanPlc, gst_span_plc, GST_TYPE_ELEMENT);
+ GST_ELEMENT_REGISTER_DEFINE (spanplc, "spanplc", GST_RANK_PRIMARY,