File 0001-Update-code-for-webrtc-audio-processing-1.... of Package gstreamer-plugins-bad

Overview Repositories Revisions Requests Users Attributes Meta

File 0001-Update-code-for-webrtc-audio-processing-1.patch of Package gstreamer-plugins-bad (Revision d769631f7435d72969c5c20278ca1a05)

Currently displaying revision d769631f7435d72969c5c20278ca1a05 , Show latest

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
#

Places

File 0001-Update-code-for-webrtc-audio-processing-1.patch of Package gstreamer-plugins-bad (Revision d769631f7435d72969c5c20278ca1a05)

Places