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webrtcdsp: Update code for webrtc-audio-processing-1

Browse source 
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>
This commit is contained in:
Arun Raghavan 2020-12-02 18:31:44 -05:00 committed by GStreamer Marge Bot
parent b3d27da397
commit d5755744c3
4 changed files with 164 additions and 207 deletions
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271
subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcdsp.cpp
View file

@ -71,9 +71,7 @@
#include "gstwebrtcdsp.h" #include "gstwebrtcdsp.h"
#include "gstwebrtcechoprobe.h" #include "gstwebrtcechoprobe.h"
#include <webrtc/modules/audio_processing/include/audio_processing.h> #include <modules/audio_processing/include/audio_processing.h>
#include <webrtc/modules/interface/module_common_types.h>
#include <webrtc/system_wrappers/include/trace.h>
GST_DEBUG_CATEGORY (webrtc_dsp_debug); GST_DEBUG_CATEGORY (webrtc_dsp_debug);
#define GST_CAT_DEFAULT (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_COMPRESSION_GAIN_DB 9
#define DEFAULT_STARTUP_MIN_VOLUME 12 #define DEFAULT_STARTUP_MIN_VOLUME 12
#define DEFAULT_LIMITER TRUE #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 FALSE
#define DEFAULT_VOICE_DETECTION_FRAME_SIZE_MS 10 #define DEFAULT_VOICE_DETECTION_FRAME_SIZE_MS 10
#define DEFAULT_VOICE_DETECTION_LIKELIHOOD webrtc::VoiceDetection::kLowLikelihood
static GstStaticPadTemplate gst_webrtc_dsp_sink_template = static GstStaticPadTemplate gst_webrtc_dsp_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink", GST_STATIC_PAD_TEMPLATE ("sink",
@ -119,7 +116,7 @@ GST_STATIC_PAD_TEMPLATE ("src",
"channels = (int) [1, MAX]") "channels = (int) [1, MAX]")
); );
typedef webrtc::EchoCancellation::SuppressionLevel GstWebrtcEchoSuppressionLevel; typedef int GstWebrtcEchoSuppressionLevel;
#define GST_TYPE_WEBRTC_ECHO_SUPPRESSION_LEVEL \ #define GST_TYPE_WEBRTC_ECHO_SUPPRESSION_LEVEL \
(gst_webrtc_echo_suppression_level_get_type ()) (gst_webrtc_echo_suppression_level_get_type ())
static GType static GType
@ -127,10 +124,9 @@ gst_webrtc_echo_suppression_level_get_type (void)
{ {
static GType suppression_level_type = 0; static GType suppression_level_type = 0;
static const GEnumValue level_types[] = { static const GEnumValue level_types[] = {
{webrtc::EchoCancellation::kLowSuppression, "Low Suppression", "low"}, {1, "Low Suppression", "low"},
{webrtc::EchoCancellation::kModerateSuppression, {2, "Moderate Suppression", "moderate"},
"Moderate Suppression", "moderate"}, {3, "high Suppression", "high"},
{webrtc::EchoCancellation::kHighSuppression, "high Suppression", "high"},
{0, NULL, NULL} {0, NULL, NULL}
}; };
@ -141,7 +137,7 @@ gst_webrtc_echo_suppression_level_get_type (void)
return suppression_level_type; return suppression_level_type;
} }
typedef webrtc::NoiseSuppression::Level GstWebrtcNoiseSuppressionLevel; typedef webrtc::AudioProcessing::Config::NoiseSuppression::Level GstWebrtcNoiseSuppressionLevel;
#define GST_TYPE_WEBRTC_NOISE_SUPPRESSION_LEVEL \ #define GST_TYPE_WEBRTC_NOISE_SUPPRESSION_LEVEL \
(gst_webrtc_noise_suppression_level_get_type ()) (gst_webrtc_noise_suppression_level_get_type ())
static GType static GType
@ -149,10 +145,10 @@ gst_webrtc_noise_suppression_level_get_type (void)
{ {
static GType suppression_level_type = 0; static GType suppression_level_type = 0;
static const GEnumValue level_types[] = { static const GEnumValue level_types[] = {
{webrtc::NoiseSuppression::kLow, "Low Suppression", "low"}, {webrtc::AudioProcessing::Config::NoiseSuppression::Level::kLow, "Low Suppression", "low"},
{webrtc::NoiseSuppression::kModerate, "Moderate Suppression", "moderate"}, {webrtc::AudioProcessing::Config::NoiseSuppression::Level::kModerate, "Moderate Suppression", "moderate"},
{webrtc::NoiseSuppression::kHigh, "High Suppression", "high"}, {webrtc::AudioProcessing::Config::NoiseSuppression::Level::kHigh, "High Suppression", "high"},
{webrtc::NoiseSuppression::kVeryHigh, "Very High Suppression", {webrtc::AudioProcessing::Config::NoiseSuppression::Level::kVeryHigh, "Very High Suppression",
"very-high"}, "very-high"},
{0, NULL, NULL} {0, NULL, NULL}
}; };
@ -164,7 +160,7 @@ gst_webrtc_noise_suppression_level_get_type (void)
return suppression_level_type; return suppression_level_type;
} }
typedef webrtc::GainControl::Mode GstWebrtcGainControlMode; typedef webrtc::AudioProcessing::Config::GainController1::Mode GstWebrtcGainControlMode;
#define GST_TYPE_WEBRTC_GAIN_CONTROL_MODE \ #define GST_TYPE_WEBRTC_GAIN_CONTROL_MODE \
(gst_webrtc_gain_control_mode_get_type ()) (gst_webrtc_gain_control_mode_get_type ())
static GType static GType
@ -172,8 +168,9 @@ gst_webrtc_gain_control_mode_get_type (void)
{ {
static GType gain_control_mode_type = 0; static GType gain_control_mode_type = 0;
static const GEnumValue mode_types[] = { static const GEnumValue mode_types[] = {
{webrtc::GainControl::kAdaptiveDigital, "Adaptive Digital", "adaptive-digital"}, {webrtc::AudioProcessing::Config::GainController1::kAdaptiveDigital, "Adaptive Digital", "adaptive-digital"},
{webrtc::GainControl::kFixedDigital, "Fixed Digital", "fixed-digital"}, {webrtc::AudioProcessing::Config::GainController1::kFixedDigital, "Fixed Digital", "fixed-digital"},
{webrtc::AudioProcessing::Config::GainController1::kAdaptiveAnalog, "Adaptive Analog", "adaptive-analog"},
{0, NULL, NULL} {0, NULL, NULL}
}; };
@ -184,7 +181,7 @@ gst_webrtc_gain_control_mode_get_type (void)
return gain_control_mode_type; return gain_control_mode_type;
} }
typedef webrtc::VoiceDetection::Likelihood GstWebrtcVoiceDetectionLikelihood; typedef int GstWebrtcVoiceDetectionLikelihood;
#define GST_TYPE_WEBRTC_VOICE_DETECTION_LIKELIHOOD \ #define GST_TYPE_WEBRTC_VOICE_DETECTION_LIKELIHOOD \
(gst_webrtc_voice_detection_likelihood_get_type ()) (gst_webrtc_voice_detection_likelihood_get_type ())
static GType static GType
@ -192,10 +189,10 @@ gst_webrtc_voice_detection_likelihood_get_type (void)
{ {
static GType likelihood_type = 0; static GType likelihood_type = 0;
static const GEnumValue likelihood_types[] = { static const GEnumValue likelihood_types[] = {
{webrtc::VoiceDetection::kVeryLowLikelihood, "Very Low Likelihood", "very-low"}, {1, "Very Low Likelihood", "very-low"},
{webrtc::VoiceDetection::kLowLikelihood, "Low Likelihood", "low"}, {2, "Low Likelihood", "low"},
{webrtc::VoiceDetection::kModerateLikelihood, "Moderate Likelihood", "moderate"}, {3, "Moderate Likelihood", "moderate"},
{webrtc::VoiceDetection::kHighLikelihood, "High Likelihood", "high"}, {4, "High Likelihood", "high"},
{0, NULL, NULL} {0, NULL, NULL}
}; };
@ -227,6 +224,7 @@ enum
PROP_VOICE_DETECTION, PROP_VOICE_DETECTION,
PROP_VOICE_DETECTION_FRAME_SIZE_MS, PROP_VOICE_DETECTION_FRAME_SIZE_MS,
PROP_VOICE_DETECTION_LIKELIHOOD, PROP_VOICE_DETECTION_LIKELIHOOD,
PROP_EXTRA_DELAY_MS,
}; };
/** /**
@ -248,7 +246,7 @@ struct _GstWebrtcDsp
/* Protected by the stream lock */ /* Protected by the stream lock */
GstAdapter *adapter; GstAdapter *adapter;
GstPlanarAudioAdapter *padapter; GstPlanarAudioAdapter *padapter;
webrtc::AudioProcessing * apm; webrtc::AudioProcessing *apm;
/* Protected by the object lock */ /* Protected by the object lock */
gchar *probe_name; gchar *probe_name;
@ -257,21 +255,15 @@ struct _GstWebrtcDsp
/* Properties */ /* Properties */
gboolean high_pass_filter; gboolean high_pass_filter;
gboolean echo_cancel; gboolean echo_cancel;
webrtc::EchoCancellation::SuppressionLevel echo_suppression_level;
gboolean noise_suppression; gboolean noise_suppression;
webrtc::NoiseSuppression::Level noise_suppression_level; webrtc::AudioProcessing::Config::NoiseSuppression::Level noise_suppression_level;
gboolean gain_control; gboolean gain_control;
gboolean experimental_agc;
gboolean extended_filter;
gboolean delay_agnostic;
gint target_level_dbfs; gint target_level_dbfs;
gint compression_gain_db; gint compression_gain_db;
gint startup_min_volume; gint startup_min_volume;
gboolean limiter; gboolean limiter;
webrtc::GainControl::Mode gain_control_mode; webrtc::AudioProcessing::Config::GainController1::Mode gain_control_mode;
gboolean voice_detection; 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, 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) GstClockTime rec_time)
{ {
GstWebrtcEchoProbe *probe = NULL; GstWebrtcEchoProbe *probe = NULL;
webrtc::AudioProcessing * apm; webrtc::AudioProcessing *apm;
webrtc::AudioFrame frame;
GstBuffer *buf = NULL; GstBuffer *buf = NULL;
GstAudioBuffer abuf;
GstFlowReturn ret = GST_FLOW_OK; GstFlowReturn ret = GST_FLOW_OK;
gint err, delay; gint err, delay;
@ -391,48 +383,44 @@ gst_webrtc_dsp_analyze_reverse_stream (GstWebrtcDsp * self,
if (!probe) if (!probe)
return GST_FLOW_OK; return GST_FLOW_OK;
webrtc::StreamConfig config (probe->info.rate, probe->info.channels,
false);
apm = self->apm; apm = self->apm;
if (self->delay_agnostic) delay = gst_webrtc_echo_probe_read (probe, rec_time, &buf);
rec_time = GST_CLOCK_TIME_NONE;
again:
delay = gst_webrtc_echo_probe_read (probe, rec_time, (gpointer) &frame, &buf);
apm->set_stream_delay_ms (delay); apm->set_stream_delay_ms (delay);
g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR);
if (delay < 0) if (delay < 0)
goto done; goto done;
if (frame.sample_rate_hz_ != self->info.rate) { if (probe->info.rate != self->info.rate) {
GST_ELEMENT_ERROR (self, STREAM, FORMAT, GST_ELEMENT_ERROR (self, STREAM, FORMAT,
("Echo Probe has rate %i , while the DSP is running at rate %i," ("Echo Probe has rate %i , while the DSP is running at rate %i,"
" use a caps filter to ensure those are the same.", " 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; ret = GST_FLOW_ERROR;
goto done; goto done;
} }
if (buf) { gst_audio_buffer_map (&abuf, &self->info, buf, GST_MAP_READWRITE);
webrtc::StreamConfig config (frame.sample_rate_hz_, frame.num_channels_,
false); if (probe->interleaved) {
GstAudioBuffer abuf; int16_t * const data = (int16_t * const) abuf.planes[0];
float * const * data;
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) if ((err = apm->ProcessReverseStream (data, config, config, data)) < 0)
GST_WARNING_OBJECT (self, "Reverse stream analyses failed: %s.", GST_WARNING_OBJECT (self, "Reverse stream analyses failed: %s.",
webrtc_error_to_string (err)); webrtc_error_to_string (err));
gst_audio_buffer_unmap (&abuf);
gst_buffer_replace (&buf, NULL);
} else { } 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.", GST_WARNING_OBJECT (self, "Reverse stream analyses failed: %s.",
webrtc_error_to_string (err)); webrtc_error_to_string (err));
} }
if (self->delay_agnostic) gst_audio_buffer_unmap (&abuf);
goto again;
done: done:
gst_object_unref (probe); gst_object_unref (probe);
@ -443,16 +431,14 @@ done:
static void static void
gst_webrtc_vad_post_activity (GstWebrtcDsp *self, GstBuffer *buffer, 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); GstClockTime timestamp = GST_BUFFER_PTS (buffer);
GstBaseTransform *trans = GST_BASE_TRANSFORM_CAST (self); GstBaseTransform *trans = GST_BASE_TRANSFORM_CAST (self);
GstStructure *s; GstStructure *s;
GstClockTime stream_time; GstClockTime stream_time;
GstAudioLevelMeta *meta; GstAudioLevelMeta *meta;
guint8 level;
level = self->apm->level_estimator ()->RMS ();
meta = gst_buffer_get_audio_level_meta (buffer); meta = gst_buffer_get_audio_level_meta (buffer);
if (meta) { if (meta) {
meta->voice_activity = stream_has_voice; meta->voice_activity = stream_has_voice;
@ -481,6 +467,7 @@ gst_webrtc_dsp_process_stream (GstWebrtcDsp * self,
{ {
GstAudioBuffer abuf; GstAudioBuffer abuf;
webrtc::AudioProcessing * apm = self->apm; webrtc::AudioProcessing * apm = self->apm;
webrtc::StreamConfig config (self->info.rate, self->info.channels, false);
gint err; gint err;
if (!gst_audio_buffer_map (&abuf, &self->info, buffer, if (!gst_audio_buffer_map (&abuf, &self->info, buffer,
@ -490,19 +477,10 @@ gst_webrtc_dsp_process_stream (GstWebrtcDsp * self,
} }
if (self->interleaved) { if (self->interleaved) {
webrtc::AudioFrame frame; int16_t * const data = (int16_t * const) abuf.planes[0];
frame.num_channels_ = self->info.channels; err = apm->ProcessStream (data, config, config, data);
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);
} else { } else {
float * const * data = (float * const *) abuf.planes; float * const * data = (float * const *) abuf.planes;
webrtc::StreamConfig config (self->info.rate, self->info.channels, false);
err = apm->ProcessStream (data, config, config, data); err = apm->ProcessStream (data, config, config, data);
} }
@ -511,10 +489,13 @@ gst_webrtc_dsp_process_stream (GstWebrtcDsp * self,
webrtc_error_to_string (err)); webrtc_error_to_string (err));
} else { } else {
if (self->voice_detection) { 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) 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; self->stream_has_voice = stream_has_voice;
} }
@ -583,21 +564,9 @@ static gboolean
gst_webrtc_dsp_start (GstBaseTransform * btrans) gst_webrtc_dsp_start (GstBaseTransform * btrans)
{ {
GstWebrtcDsp *self = GST_WEBRTC_DSP (btrans); GstWebrtcDsp *self = GST_WEBRTC_DSP (btrans);
webrtc::Config config;
GST_OBJECT_LOCK (self); 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) { if (self->echo_cancel) {
self->probe = gst_webrtc_acquire_echo_probe (self->probe_name); 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) gst_webrtc_dsp_setup (GstAudioFilter * filter, const GstAudioInfo * info)
{ {
GstWebrtcDsp *self = GST_WEBRTC_DSP (filter); GstWebrtcDsp *self = GST_WEBRTC_DSP (filter);
webrtc::AudioProcessing * apm; webrtc::AudioProcessing::Config config;
webrtc::ProcessingConfig pconfig;
GstAudioInfo probe_info = *info; GstAudioInfo probe_info = *info;
gint err = 0;
GST_LOG_OBJECT (self, "setting format to %s with %i Hz and %i channels", GST_LOG_OBJECT (self, "setting format to %s with %i Hz and %i channels",
info->finfo->description, info->rate, info->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->info = *info;
self->interleaved = (info->layout == GST_AUDIO_LAYOUT_INTERLEAVED); self->interleaved = (info->layout == GST_AUDIO_LAYOUT_INTERLEAVED);
apm = self->apm; self->apm = webrtc::AudioProcessingBuilder().Create();
if (!self->interleaved) if (!self->interleaved)
gst_planar_audio_adapter_configure (self->padapter, info); 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_samples = info->rate / 100;
self->period_size = self->period_samples * info->bpf; self->period_size = self->period_samples * info->bpf;
if (self->interleaved && if (self->interleaved && (self->period_size > MAX_DATA_SIZE_SAMPLES * 2))
(webrtc::AudioFrame::kMaxDataSizeSamples * 2) < self->period_size)
goto period_too_big; goto period_too_big;
if (self->probe) { if (self->probe) {
@ -658,40 +624,31 @@ gst_webrtc_dsp_setup (GstAudioFilter * filter, const GstAudioInfo * info)
GST_WEBRTC_ECHO_PROBE_UNLOCK (self->probe); 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 */ /* Setup Filters */
// TODO: expose pre_amplifier
if (self->high_pass_filter) { if (self->high_pass_filter) {
GST_DEBUG_OBJECT (self, "Enabling 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) { if (self->echo_cancel) {
GST_DEBUG_OBJECT (self, "Enabling Echo Cancellation"); GST_DEBUG_OBJECT (self, "Enabling Echo Cancellation");
apm->echo_cancellation ()->enable_drift_compensation (false); config.echo_canceller.enabled = true;
apm->echo_cancellation ()
->set_suppression_level (self->echo_suppression_level);
apm->echo_cancellation ()->Enable (true);
} }
if (self->noise_suppression) { if (self->noise_suppression) {
GST_DEBUG_OBJECT (self, "Enabling Noise Suppression"); GST_DEBUG_OBJECT (self, "Enabling Noise Suppression");
apm->noise_suppression ()->set_level (self->noise_suppression_level); config.noise_suppression.enabled = true;
apm->noise_suppression ()->Enable (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) { if (self->gain_control) {
@ -706,30 +663,17 @@ gst_webrtc_dsp_setup (GstAudioFilter * filter, const GstAudioInfo * info)
g_type_class_unref (mode_class); g_type_class_unref (mode_class);
apm->gain_control ()->set_mode (self->gain_control_mode); config.gain_controller1.enabled = true;
apm->gain_control ()->set_target_level_dbfs (self->target_level_dbfs); config.gain_controller1.target_level_dbfs = self->target_level_dbfs;
apm->gain_control ()->set_compression_gain_db (self->compression_gain_db); config.gain_controller1.compression_gain_db = self->compression_gain_db;
apm->gain_control ()->enable_limiter (self->limiter); config.gain_controller1.enable_limiter = self->limiter;
apm->gain_control ()->Enable (true); config.level_estimation.enabled = true;
} }
if (self->voice_detection) { // TODO: expose gain controller 2
GEnumClass *likelihood_class = (GEnumClass *) // TODO: expose residual echo detector
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);
self->stream_has_voice = FALSE; self->apm->ApplyConfig (config);
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);
}
GST_OBJECT_UNLOCK (self); GST_OBJECT_UNLOCK (self);
@ -738,9 +682,9 @@ gst_webrtc_dsp_setup (GstAudioFilter * filter, const GstAudioInfo * info)
period_too_big: period_too_big:
GST_OBJECT_UNLOCK (self); GST_OBJECT_UNLOCK (self);
GST_WARNING_OBJECT (self, "webrtcdsp format produce too big period " 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.", "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; return FALSE;
probe_has_wrong_rate: probe_has_wrong_rate:
@ -751,14 +695,6 @@ probe_has_wrong_rate:
" use a caps filter to ensure those are the same.", " use a caps filter to ensure those are the same.",
probe_info.rate, info->rate), (NULL)); probe_info.rate, info->rate), (NULL));
return FALSE; 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 static gboolean
@ -803,8 +739,6 @@ gst_webrtc_dsp_set_property (GObject * object,
self->echo_cancel = g_value_get_boolean (value); self->echo_cancel = g_value_get_boolean (value);
break; break;
case PROP_ECHO_SUPPRESSION_LEVEL: case PROP_ECHO_SUPPRESSION_LEVEL:
self->echo_suppression_level =
(GstWebrtcEchoSuppressionLevel) g_value_get_enum (value);
break; break;
case PROP_NOISE_SUPPRESSION: case PROP_NOISE_SUPPRESSION:
self->noise_suppression = g_value_get_boolean (value); 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); self->gain_control = g_value_get_boolean (value);
break; break;
case PROP_EXPERIMENTAL_AGC: case PROP_EXPERIMENTAL_AGC:
self->experimental_agc = g_value_get_boolean (value);
break; break;
case PROP_EXTENDED_FILTER: case PROP_EXTENDED_FILTER:
self->extended_filter = g_value_get_boolean (value);
break; break;
case PROP_DELAY_AGNOSTIC: case PROP_DELAY_AGNOSTIC:
self->delay_agnostic = g_value_get_boolean (value);
break; break;
case PROP_TARGET_LEVEL_DBFS: case PROP_TARGET_LEVEL_DBFS:
self->target_level_dbfs = g_value_get_int (value); 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); self->voice_detection = g_value_get_boolean (value);
break; break;
case PROP_VOICE_DETECTION_FRAME_SIZE_MS: case PROP_VOICE_DETECTION_FRAME_SIZE_MS:
self->voice_detection_frame_size_ms = g_value_get_int (value);
break; break;
case PROP_VOICE_DETECTION_LIKELIHOOD: case PROP_VOICE_DETECTION_LIKELIHOOD:
self->voice_detection_likelihood =
(GstWebrtcVoiceDetectionLikelihood) g_value_get_enum (value);
break; break;
default: default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); 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); g_value_set_boolean (value, self->echo_cancel);
break; break;
case PROP_ECHO_SUPPRESSION_LEVEL: case PROP_ECHO_SUPPRESSION_LEVEL:
g_value_set_enum (value, self->echo_suppression_level); g_value_set_enum (value, (GstWebrtcEchoSuppressionLevel) 2);
break; break;
case PROP_NOISE_SUPPRESSION: case PROP_NOISE_SUPPRESSION:
g_value_set_boolean (value, self->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); g_value_set_boolean (value, self->gain_control);
break; break;
case PROP_EXPERIMENTAL_AGC: case PROP_EXPERIMENTAL_AGC:
g_value_set_boolean (value, self->experimental_agc); g_value_set_boolean (value, false);
break; break;
case PROP_EXTENDED_FILTER: case PROP_EXTENDED_FILTER:
g_value_set_boolean (value, self->extended_filter); g_value_set_boolean (value, false);
break; break;
case PROP_DELAY_AGNOSTIC: case PROP_DELAY_AGNOSTIC:
g_value_set_boolean (value, self->delay_agnostic); g_value_set_boolean (value, false);
break; break;
case PROP_TARGET_LEVEL_DBFS: case PROP_TARGET_LEVEL_DBFS:
g_value_set_int (value, self->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); g_value_set_boolean (value, self->voice_detection);
break; break;
case PROP_VOICE_DETECTION_FRAME_SIZE_MS: 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; break;
case PROP_VOICE_DETECTION_LIKELIHOOD: case PROP_VOICE_DETECTION_LIKELIHOOD:
g_value_set_enum (value, self->voice_detection_likelihood); g_value_set_enum (value, 2);
break; break;
default: default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); 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, g_object_class_install_property (gobject_class,
PROP_ECHO_SUPPRESSION_LEVEL, 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 " "Controls the aggressiveness of the suppressor. A higher level "
"trades off double-talk performance for increased echo suppression.", "trades off double-talk performance for increased echo suppression.",
GST_TYPE_WEBRTC_ECHO_SUPPRESSION_LEVEL, GST_TYPE_WEBRTC_ECHO_SUPPRESSION_LEVEL, 2,
webrtc::EchoCancellation::kModerateSuppression,
(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | (GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
G_PARAM_CONSTRUCT))); G_PARAM_CONSTRUCT | G_PARAM_DEPRECATED)));
g_object_class_install_property (gobject_class, g_object_class_install_property (gobject_class,
PROP_NOISE_SUPPRESSION, PROP_NOISE_SUPPRESSION,
@ -1026,7 +954,7 @@ gst_webrtc_dsp_class_init (GstWebrtcDspClass * klass)
"Controls the aggressiveness of the suppression. Increasing the " "Controls the aggressiveness of the suppression. Increasing the "
"level will reduce the noise level at the expense of a higher " "level will reduce the noise level at the expense of a higher "
"speech distortion.", GST_TYPE_WEBRTC_NOISE_SUPPRESSION_LEVEL, "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 | (GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
G_PARAM_CONSTRUCT))); G_PARAM_CONSTRUCT)));
@ -1039,24 +967,26 @@ gst_webrtc_dsp_class_init (GstWebrtcDspClass * klass)
g_object_class_install_property (gobject_class, g_object_class_install_property (gobject_class,
PROP_EXPERIMENTAL_AGC, 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.", "Enable or disable experimental automatic gain control.",
FALSE, (GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | 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, g_object_class_install_property (gobject_class,
PROP_EXTENDED_FILTER, PROP_EXTENDED_FILTER,
g_param_spec_boolean ("extended-filter", "Extended Filter", g_param_spec_boolean ("extended-filter", "Extended Filter",
"Enable or disable the extended filter.", "Enable or disable the extended filter.",
TRUE, (GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | 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, g_object_class_install_property (gobject_class,
PROP_DELAY_AGNOSTIC, 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.", "Enable or disable the delay agnostic mode.",
FALSE, (GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | 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, g_object_class_install_property (gobject_class,
PROP_TARGET_LEVEL_DBFS, PROP_TARGET_LEVEL_DBFS,
@ -1111,24 +1041,23 @@ gst_webrtc_dsp_class_init (GstWebrtcDspClass * klass)
g_object_class_install_property (gobject_class, g_object_class_install_property (gobject_class,
PROP_VOICE_DETECTION_FRAME_SIZE_MS, PROP_VOICE_DETECTION_FRAME_SIZE_MS,
g_param_spec_int ("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. " "Sets the |size| of the frames in ms on which the VAD will operate. "
"Larger frames will improve detection accuracy, but reduce the " "Larger frames will improve detection accuracy, but reduce the "
"frequency of updates", "frequency of updates",
10, 30, DEFAULT_VOICE_DETECTION_FRAME_SIZE_MS, 10, 30, DEFAULT_VOICE_DETECTION_FRAME_SIZE_MS,
(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | (GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
G_PARAM_CONSTRUCT))); G_PARAM_CONSTRUCT | G_PARAM_DEPRECATED)));
g_object_class_install_property (gobject_class, g_object_class_install_property (gobject_class,
PROP_VOICE_DETECTION_LIKELIHOOD, PROP_VOICE_DETECTION_LIKELIHOOD,
g_param_spec_enum ("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 " "Specifies the likelihood that a frame will be declared to contain "
"voice.", "voice.",
GST_TYPE_WEBRTC_VOICE_DETECTION_LIKELIHOOD, GST_TYPE_WEBRTC_VOICE_DETECTION_LIKELIHOOD, 2,
DEFAULT_VOICE_DETECTION_LIKELIHOOD,
(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | (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_GAIN_CONTROL_MODE, (GstPluginAPIFlags) 0);
gst_type_mark_as_plugin_api (GST_TYPE_WEBRTC_NOISE_SUPPRESSION_LEVEL, (GstPluginAPIFlags) 0); gst_type_mark_as_plugin_api (GST_TYPE_WEBRTC_NOISE_SUPPRESSION_LEVEL, (GstPluginAPIFlags) 0);

87
subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.cpp
View file

@ -33,7 +33,8 @@
#include "gstwebrtcechoprobe.h" #include "gstwebrtcechoprobe.h"
#include <webrtc/modules/interface/module_common_types.h> #include <modules/audio_processing/include/audio_processing.h>
#include <gst/audio/audio.h> #include <gst/audio/audio.h>
GST_DEBUG_CATEGORY_EXTERN (webrtc_dsp_debug); 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; self->period_size = self->period_samples * info->bpf;
if (self->interleaved && if (self->interleaved &&
(webrtc::AudioFrame::kMaxDataSizeSamples * 2) < self->period_size) (MAX_DATA_SIZE_SAMPLES * 2) < self->period_size)
goto period_too_big; goto period_too_big;
GST_WEBRTC_ECHO_PROBE_UNLOCK (self); GST_WEBRTC_ECHO_PROBE_UNLOCK (self);
@ -112,9 +113,9 @@ gst_webrtc_echo_probe_setup (GstAudioFilter * filter, const GstAudioInfo * info)
period_too_big: period_too_big:
GST_WEBRTC_ECHO_PROBE_UNLOCK (self); GST_WEBRTC_ECHO_PROBE_UNLOCK (self);
GST_WARNING_OBJECT (self, "webrtcdsp format produce too big period " 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.", "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; return FALSE;
} }
@ -303,18 +304,20 @@ gst_webrtc_release_echo_probe (GstWebrtcEchoProbe * probe)
gint gint
gst_webrtc_echo_probe_read (GstWebrtcEchoProbe * self, GstClockTime rec_time, gst_webrtc_echo_probe_read (GstWebrtcEchoProbe * self, GstClockTime rec_time,
gpointer _frame, GstBuffer ** buf) GstBuffer ** buf)
{ {
webrtc::AudioFrame * frame = (webrtc::AudioFrame *) _frame;
GstClockTimeDiff diff; GstClockTimeDiff diff;
gsize avail, skip, offset, size; gsize avail, skip, offset, size = 0;
gint delay = -1; gint delay = -1;
GST_WEBRTC_ECHO_PROBE_LOCK (self); 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) || if (!GST_CLOCK_TIME_IS_VALID (self->latency) ||
!GST_AUDIO_INFO_IS_VALID (&self->info)) !GST_AUDIO_INFO_IS_VALID (&self->info))
goto done; goto copy;
if (self->interleaved) if (self->interleaved)
avail = gst_adapter_available (self->adapter) / self->info.bpf; 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 */ /* In delay agnostic mode, just return 10ms of data */
if (!GST_CLOCK_TIME_IS_VALID (rec_time)) { if (!GST_CLOCK_TIME_IS_VALID (rec_time)) {
if (avail < self->period_samples) if (avail < self->period_samples)
goto done; goto copy;
size = self->period_samples; size = self->period_samples;
skip = 0; skip = 0;
@ -371,23 +374,51 @@ gst_webrtc_echo_probe_read (GstWebrtcEchoProbe * self, GstClockTime rec_time,
size = MIN (avail - offset, self->period_samples - skip); size = MIN (avail - offset, self->period_samples - skip);
copy: copy:
if (self->interleaved) { if (!size) {
skip *= self->info.bpf; /* No data, provide a period's worth of silence */
offset *= self->info.bpf; *buf = gst_buffer_new_allocate (NULL, self->period_size, NULL);
size *= self->info.bpf; gst_buffer_memset (*buf, 0, 0, self->period_size);
gst_buffer_add_audio_meta (*buf, &self->info, self->period_samples,
if (size < self->period_size) NULL);
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);
}
} else { } 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; 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); gst_planar_audio_adapter_flush (self->padapter, offset);
/* we need to fill silence at the beginning and/or the end of each /* 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, ret = gst_planar_audio_adapter_take_buffer (self->padapter, size,
GST_MAP_READWRITE); 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; *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; delay = self->delay;
done:
GST_WEBRTC_ECHO_PROBE_UNLOCK (self); GST_WEBRTC_ECHO_PROBE_UNLOCK (self);
return delay; return delay;

9
subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.h
View file

@ -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_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) #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 _GstWebrtcEchoProbe GstWebrtcEchoProbe;
typedef struct _GstWebrtcEchoProbeClass GstWebrtcEchoProbeClass; typedef struct _GstWebrtcEchoProbeClass GstWebrtcEchoProbeClass;
@ -71,6 +77,7 @@ struct _GstWebrtcEchoProbe
GstClockTime latency; GstClockTime latency;
gint delay; gint delay;
gboolean interleaved; gboolean interleaved;
gint extra_delay;
GstSegment segment; GstSegment segment;
GstAdapter *adapter; GstAdapter *adapter;
@ -92,7 +99,7 @@ GST_ELEMENT_REGISTER_DECLARE (webrtcechoprobe);
GstWebrtcEchoProbe *gst_webrtc_acquire_echo_probe (const gchar * name); GstWebrtcEchoProbe *gst_webrtc_acquire_echo_probe (const gchar * name);
void gst_webrtc_release_echo_probe (GstWebrtcEchoProbe * probe); void gst_webrtc_release_echo_probe (GstWebrtcEchoProbe * probe);
gint gst_webrtc_echo_probe_read (GstWebrtcEchoProbe * self, gint gst_webrtc_echo_probe_read (GstWebrtcEchoProbe * self,
GstClockTime rec_time, gpointer frame, GstBuffer ** buf); GstClockTime rec_time, GstBuffer ** buf);
G_END_DECLS G_END_DECLS
#endif /* __GST_WEBRTC_ECHO_PROBE_H__ */ #endif /* __GST_WEBRTC_ECHO_PROBE_H__ */

4
subprojects/gst-plugins-bad/ext/webrtcdsp/meson.build
View file

@ -4,7 +4,7 @@ webrtc_sources = [
'gstwebrtcdspplugin.cpp' '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')) required : get_option('webrtcdsp'))
if not gnustl_dep.found() and get_option('webrtcdsp').enabled() 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], dependencies : [gstbase_dep, gstaudio_dep, gstbadaudio_dep, webrtc_dep, gnustl_dep],
install : true, install : true,
install_dir : plugins_install_dir, install_dir : plugins_install_dir,
override_options : ['cpp_std=c++11'], override_options : ['cpp_std=c++17'],
) )
plugins += [gstwebrtcdsp] plugins += [gstwebrtcdsp]
endif endif