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Add patch to use new webrtc version

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Resolves: RHEL-28928
This commit is contained in:
Wim Taymans 2024-07-15 16:53:21 +02:00
parent e0ce7a4225
commit 851eb29c89
4 changed files with 1077 additions and 1 deletions
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881
0001-webrtcdsp-Update-code-for-webrtc-audio-processing-1.patch Normal file
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@ -0,0 +1,881 @@
From 0757dbc0b406ebd6c38c8d86bf4d4738a232db40 Mon Sep 17 00:00:00 2001
From: Arun Raghavan <arun@asymptotic.io>
Date: Wed, 2 Dec 2020 18:31:44 -0500
Subject: [PATCH 1/3] 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 7ee09488fb..c9a7cdae2f 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 acdb3d8a7d..8e8ca064c4 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 36fd34f179..488c0e958f 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 5aeae69a44..09565e27c7 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
--
2.45.2

36
0002-webrtcdsp-Map-probe-buffers-with-probe-info-not-dsp-.patch Normal file
View File

@ -0,0 +1,36 @@
From 187eb3d4874ff6ceccf85bed3e8161f68f8e09da Mon Sep 17 00:00:00 2001
From: Nirbheek Chauhan <nirbheek@centricular.com>
Date: Tue, 6 Jun 2023 16:34:19 +0530
Subject: [PATCH 2/3] webrtcdsp: Map probe buffers with probe info, not dsp
info
The probe's info may not precisely match the dsp's info. For instance,
the number of channels or their layout might be different.
```
GStreamer-Audio-CRITICAL **: 16:21:32.899: the GstAudioInfo argument is not equal to the GstAudioMeta's attached info
```
This broke in d5755744c3e2b70e9f04704ae9d18b928d9fa456.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4849>
---
subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcdsp.cpp | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcdsp.cpp b/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcdsp.cpp
index c9a7cdae2f..87e5cbc4a2 100644
--- a/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcdsp.cpp
+++ b/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcdsp.cpp
@@ -404,7 +404,7 @@ gst_webrtc_dsp_analyze_reverse_stream (GstWebrtcDsp * self,
goto done;
}
- gst_audio_buffer_map (&abuf, &self->info, buf, GST_MAP_READWRITE);
+ gst_audio_buffer_map (&abuf, &probe->info, buf, GST_MAP_READWRITE);
if (probe->interleaved) {
int16_t * const data = (int16_t * const) abuf.planes[0];
--
2.45.2

146
0003-webrtcdsp-Deal-with-echo-probe-info-not-being-availa.patch Normal file
View File

@ -0,0 +1,146 @@
From 7c5003367fe7528c5b5a5bac0243910e46019d0f Mon Sep 17 00:00:00 2001
From: Arun Raghavan <arun@asymptotic.io>
Date: Tue, 13 Jun 2023 11:41:43 -0400
Subject: [PATCH 3/3] webrtcdsp: Deal with echo probe info not being available
Even if we don't yet know what the echo probe format is, we want to be able to
provide silence for the reverse path, so that when the probe becomes available,
there is no ambiguity around what time period the new set of samples are for.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4849>
---
.../ext/webrtcdsp/gstwebrtcdsp.cpp | 20 ++++++++------
.../ext/webrtcdsp/gstwebrtcechoprobe.cpp | 26 +++++++++++++------
.../ext/webrtcdsp/gstwebrtcechoprobe.h | 3 ++-
3 files changed, 32 insertions(+), 17 deletions(-)
diff --git a/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcdsp.cpp b/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcdsp.cpp
index 87e5cbc4a2..dc4190cff0 100644
--- a/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcdsp.cpp
+++ b/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcdsp.cpp
@@ -370,6 +370,8 @@ gst_webrtc_dsp_analyze_reverse_stream (GstWebrtcDsp * self,
GstWebrtcEchoProbe *probe = NULL;
webrtc::AudioProcessing *apm;
GstBuffer *buf = NULL;
+ GstAudioInfo info;
+ gboolean interleaved = self->interleaved;
GstAudioBuffer abuf;
GstFlowReturn ret = GST_FLOW_OK;
gint err, delay;
@@ -377,36 +379,38 @@ gst_webrtc_dsp_analyze_reverse_stream (GstWebrtcDsp * self,
GST_OBJECT_LOCK (self);
if (self->echo_cancel)
probe = GST_WEBRTC_ECHO_PROBE (g_object_ref (self->probe));
+ info = self->info;
GST_OBJECT_UNLOCK (self);
/* If echo cancellation is disabled */
if (!probe)
return GST_FLOW_OK;
- webrtc::StreamConfig config (probe->info.rate, probe->info.channels,
- false);
- apm = self->apm;
+ delay =
+ gst_webrtc_echo_probe_read (probe, rec_time, &buf, &info, &interleaved);
- delay = gst_webrtc_echo_probe_read (probe, rec_time, &buf);
+ apm = self->apm;
apm->set_stream_delay_ms (delay);
+ webrtc::StreamConfig config (info.rate, info.channels, false);
+
g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR);
if (delay < 0)
goto done;
- if (probe->info.rate != self->info.rate) {
+ if (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.",
- probe->info.rate, self->info.rate), (NULL));
+ info.rate, self->info.rate), (NULL));
ret = GST_FLOW_ERROR;
goto done;
}
- gst_audio_buffer_map (&abuf, &probe->info, buf, GST_MAP_READWRITE);
+ gst_audio_buffer_map (&abuf, &info, buf, GST_MAP_READWRITE);
- if (probe->interleaved) {
+ if (interleaved) {
int16_t * const data = (int16_t * const) abuf.planes[0];
if ((err = apm->ProcessReverseStream (data, config, config, data)) < 0)
diff --git a/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.cpp b/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.cpp
index 8e8ca064c4..13d0fc6cb0 100644
--- a/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.cpp
+++ b/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.cpp
@@ -304,7 +304,7 @@ gst_webrtc_release_echo_probe (GstWebrtcEchoProbe * probe)
gint
gst_webrtc_echo_probe_read (GstWebrtcEchoProbe * self, GstClockTime rec_time,
- GstBuffer ** buf)
+ GstBuffer ** buf, GstAudioInfo * info, gboolean * interleaved)
{
GstClockTimeDiff diff;
gsize avail, skip, offset, size = 0;
@@ -315,10 +315,17 @@ gst_webrtc_echo_probe_read (GstWebrtcEchoProbe * self, GstClockTime rec_time,
/* 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))
+ if (!GST_CLOCK_TIME_IS_VALID (self->latency))
goto copy;
+ /* If we have a format, use that, else generate silence in input format */
+ if (!GST_AUDIO_INFO_IS_VALID (&self->info)) {
+ goto copy;
+ } else {
+ *info = self->info;
+ *interleaved = self->interleaved;
+ }
+
if (self->interleaved)
avail = gst_adapter_available (self->adapter) / self->info.bpf;
else
@@ -375,11 +382,14 @@ gst_webrtc_echo_probe_read (GstWebrtcEchoProbe * self, GstClockTime rec_time,
copy:
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);
+ /* No data, provide a period's worth of silence, using our format if we have
+ * it, or the provided format if we don't */
+ guint period_samples = info->rate / 100;
+ guint period_size = period_samples * info->bpf;
+
+ *buf = gst_buffer_new_allocate (NULL, period_size, NULL);
+ gst_buffer_memset (*buf, 0, 0, period_size);
+ gst_buffer_add_audio_meta (*buf, info, 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 */
diff --git a/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.h b/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.h
index 488c0e958f..30ab50b355 100644
--- a/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.h
+++ b/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.h
@@ -99,7 +99,8 @@ 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, GstBuffer ** buf);
+ GstClockTime rec_time, GstBuffer ** buf, GstAudioInfo * info,
+ gboolean * interleaved);
G_END_DECLS
#endif /* __GST_WEBRTC_ECHO_PROBE_H__ */
--
2.45.2

15
gstreamer1-plugins-bad-free.spec
View File

@ -12,7 +12,7 @@
Name: gstreamer1-plugins-bad-free
Version: 1.22.12
Release: 2%{?dist}
Release: 3%{?dist}
Summary: GStreamer streaming media framework "bad" plugins
License: LGPLv2+ and LGPLv2
@ -34,6 +34,11 @@ Patch: openh264-add-license-file.patch
# https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5780
Patch: openh264-drop-runtime-version-checks.patch
Patch: 0001-webrtcdsp-Update-code-for-webrtc-audio-processing-1.patch
Patch: 0002-webrtcdsp-Map-probe-buffers-with-probe-info-not-dsp-.patch
Patch: 0003-webrtcdsp-Deal-with-echo-probe-info-not-being-availa.patch
BuildRequires: meson >= 0.48.0
BuildRequires: gcc-c++
BuildRequires: gstreamer1-devel >= %{version}
@ -78,7 +83,11 @@ BuildRequires: libwebp-devel
BuildRequires: mesa-libEGL-devel
BuildRequires: vulkan-devel
#BuildRequires: mesa-vulkan-devel
%if 0%{?fedora} >= 40 || 0%{?rhel} >= 10
BuildRequires: pkgconfig(webrtc-audio-processing-1)
%else
BuildRequires: pkgconfig(webrtc-audio-processing) >= 0.3
%endif
%if 0
BuildRequires: wpewebkit-devel
BuildRequires: wpebackend-fdo-devel
@ -777,6 +786,10 @@ rm $RPM_BUILD_ROOT%{_bindir}/playout
%changelog
* Mon Jul 15 2024 Wim Taymans <wtaymans@redhat.com> - 1.22.12-3
- Add patch to use new webrtc version
- Resolves: RHEL-28928
* Mon Jun 24 2024 Troy Dawson <tdawson@redhat.com> - 1.22.12-2
- Bump release for June 2024 mass rebuild