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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>
This commit is contained in:
Arun Raghavan
GStreamer Marge Bot
parent
b3d27da397
commit
d5755744c3
4 changed files with 164 additions and 207 deletions
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@ -71,9 +71,7 @@
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#include "gstwebrtcdsp.h"
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#include "gstwebrtcechoprobe.h"
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#include <webrtc/modules/audio_processing/include/audio_processing.h>
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#include <webrtc/modules/interface/module_common_types.h>
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#include <webrtc/system_wrappers/include/trace.h>
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#include <modules/audio_processing/include/audio_processing.h>
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GST_DEBUG_CATEGORY (webrtc_dsp_debug);
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#define GST_CAT_DEFAULT (webrtc_dsp_debug)
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@ -82,10 +80,9 @@ GST_DEBUG_CATEGORY (webrtc_dsp_debug);
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#define DEFAULT_COMPRESSION_GAIN_DB 9
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#define DEFAULT_STARTUP_MIN_VOLUME 12
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#define DEFAULT_LIMITER TRUE
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#define DEFAULT_GAIN_CONTROL_MODE webrtc::GainControl::kAdaptiveDigital
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#define DEFAULT_GAIN_CONTROL_MODE webrtc::AudioProcessing::Config::GainController1::Mode::kAdaptiveDigital
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#define DEFAULT_VOICE_DETECTION FALSE
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#define DEFAULT_VOICE_DETECTION_FRAME_SIZE_MS 10
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#define DEFAULT_VOICE_DETECTION_LIKELIHOOD webrtc::VoiceDetection::kLowLikelihood
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static GstStaticPadTemplate gst_webrtc_dsp_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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@ -119,7 +116,7 @@ GST_STATIC_PAD_TEMPLATE ("src",
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"channels = (int) [1, MAX]")
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);
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typedef webrtc::EchoCancellation::SuppressionLevel GstWebrtcEchoSuppressionLevel;
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typedef int GstWebrtcEchoSuppressionLevel;
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#define GST_TYPE_WEBRTC_ECHO_SUPPRESSION_LEVEL \
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(gst_webrtc_echo_suppression_level_get_type ())
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static GType
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@ -127,10 +124,9 @@ gst_webrtc_echo_suppression_level_get_type (void)
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{
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static GType suppression_level_type = 0;
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static const GEnumValue level_types[] = {
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{webrtc::EchoCancellation::kLowSuppression, "Low Suppression", "low"},
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{webrtc::EchoCancellation::kModerateSuppression,
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"Moderate Suppression", "moderate"},
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{webrtc::EchoCancellation::kHighSuppression, "high Suppression", "high"},
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{1, "Low Suppression", "low"},
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{2, "Moderate Suppression", "moderate"},
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{3, "high Suppression", "high"},
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{0, NULL, NULL}
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};
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@ -141,7 +137,7 @@ gst_webrtc_echo_suppression_level_get_type (void)
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return suppression_level_type;
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}
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typedef webrtc::NoiseSuppression::Level GstWebrtcNoiseSuppressionLevel;
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typedef webrtc::AudioProcessing::Config::NoiseSuppression::Level GstWebrtcNoiseSuppressionLevel;
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#define GST_TYPE_WEBRTC_NOISE_SUPPRESSION_LEVEL \
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(gst_webrtc_noise_suppression_level_get_type ())
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static GType
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@ -149,10 +145,10 @@ gst_webrtc_noise_suppression_level_get_type (void)
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{
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static GType suppression_level_type = 0;
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static const GEnumValue level_types[] = {
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{webrtc::NoiseSuppression::kLow, "Low Suppression", "low"},
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{webrtc::NoiseSuppression::kModerate, "Moderate Suppression", "moderate"},
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{webrtc::NoiseSuppression::kHigh, "High Suppression", "high"},
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{webrtc::NoiseSuppression::kVeryHigh, "Very High Suppression",
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{webrtc::AudioProcessing::Config::NoiseSuppression::Level::kLow, "Low Suppression", "low"},
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{webrtc::AudioProcessing::Config::NoiseSuppression::Level::kModerate, "Moderate Suppression", "moderate"},
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{webrtc::AudioProcessing::Config::NoiseSuppression::Level::kHigh, "High Suppression", "high"},
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{webrtc::AudioProcessing::Config::NoiseSuppression::Level::kVeryHigh, "Very High Suppression",
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"very-high"},
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{0, NULL, NULL}
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};
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@ -164,7 +160,7 @@ gst_webrtc_noise_suppression_level_get_type (void)
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return suppression_level_type;
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}
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typedef webrtc::GainControl::Mode GstWebrtcGainControlMode;
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typedef webrtc::AudioProcessing::Config::GainController1::Mode GstWebrtcGainControlMode;
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#define GST_TYPE_WEBRTC_GAIN_CONTROL_MODE \
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(gst_webrtc_gain_control_mode_get_type ())
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static GType
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@ -172,8 +168,9 @@ gst_webrtc_gain_control_mode_get_type (void)
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{
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static GType gain_control_mode_type = 0;
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static const GEnumValue mode_types[] = {
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{webrtc::GainControl::kAdaptiveDigital, "Adaptive Digital", "adaptive-digital"},
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{webrtc::GainControl::kFixedDigital, "Fixed Digital", "fixed-digital"},
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{webrtc::AudioProcessing::Config::GainController1::kAdaptiveDigital, "Adaptive Digital", "adaptive-digital"},
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{webrtc::AudioProcessing::Config::GainController1::kFixedDigital, "Fixed Digital", "fixed-digital"},
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{webrtc::AudioProcessing::Config::GainController1::kAdaptiveAnalog, "Adaptive Analog", "adaptive-analog"},
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{0, NULL, NULL}
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};
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@ -184,7 +181,7 @@ gst_webrtc_gain_control_mode_get_type (void)
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return gain_control_mode_type;
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}
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typedef webrtc::VoiceDetection::Likelihood GstWebrtcVoiceDetectionLikelihood;
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typedef int GstWebrtcVoiceDetectionLikelihood;
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#define GST_TYPE_WEBRTC_VOICE_DETECTION_LIKELIHOOD \
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(gst_webrtc_voice_detection_likelihood_get_type ())
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static GType
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@ -192,10 +189,10 @@ gst_webrtc_voice_detection_likelihood_get_type (void)
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{
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static GType likelihood_type = 0;
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static const GEnumValue likelihood_types[] = {
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{webrtc::VoiceDetection::kVeryLowLikelihood, "Very Low Likelihood", "very-low"},
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{webrtc::VoiceDetection::kLowLikelihood, "Low Likelihood", "low"},
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{webrtc::VoiceDetection::kModerateLikelihood, "Moderate Likelihood", "moderate"},
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{webrtc::VoiceDetection::kHighLikelihood, "High Likelihood", "high"},
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{1, "Very Low Likelihood", "very-low"},
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{2, "Low Likelihood", "low"},
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{3, "Moderate Likelihood", "moderate"},
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{4, "High Likelihood", "high"},
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{0, NULL, NULL}
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};
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@ -227,6 +224,7 @@ enum
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PROP_VOICE_DETECTION,
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PROP_VOICE_DETECTION_FRAME_SIZE_MS,
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PROP_VOICE_DETECTION_LIKELIHOOD,
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PROP_EXTRA_DELAY_MS,
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};
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/**
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@ -248,7 +246,7 @@ struct _GstWebrtcDsp
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/* Protected by the stream lock */
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GstAdapter *adapter;
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GstPlanarAudioAdapter *padapter;
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webrtc::AudioProcessing * apm;
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webrtc::AudioProcessing *apm;
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/* Protected by the object lock */
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gchar *probe_name;
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@ -257,21 +255,15 @@ struct _GstWebrtcDsp
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/* Properties */
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gboolean high_pass_filter;
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gboolean echo_cancel;
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webrtc::EchoCancellation::SuppressionLevel echo_suppression_level;
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gboolean noise_suppression;
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webrtc::NoiseSuppression::Level noise_suppression_level;
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webrtc::AudioProcessing::Config::NoiseSuppression::Level noise_suppression_level;
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gboolean gain_control;
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gboolean experimental_agc;
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gboolean extended_filter;
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gboolean delay_agnostic;
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gint target_level_dbfs;
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gint compression_gain_db;
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gint startup_min_volume;
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gboolean limiter;
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webrtc::GainControl::Mode gain_control_mode;
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webrtc::AudioProcessing::Config::GainController1::Mode gain_control_mode;
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gboolean voice_detection;
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gint voice_detection_frame_size_ms;
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webrtc::VoiceDetection::Likelihood voice_detection_likelihood;
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};
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G_DEFINE_TYPE_WITH_CODE (GstWebrtcDsp, gst_webrtc_dsp, GST_TYPE_AUDIO_FILTER,
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@ -376,9 +368,9 @@ gst_webrtc_dsp_analyze_reverse_stream (GstWebrtcDsp * self,
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GstClockTime rec_time)
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{
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GstWebrtcEchoProbe *probe = NULL;
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webrtc::AudioProcessing * apm;
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webrtc::AudioFrame frame;
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webrtc::AudioProcessing *apm;
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GstBuffer *buf = NULL;
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GstAudioBuffer abuf;
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GstFlowReturn ret = GST_FLOW_OK;
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gint err, delay;
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@ -391,48 +383,44 @@ gst_webrtc_dsp_analyze_reverse_stream (GstWebrtcDsp * self,
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if (!probe)
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return GST_FLOW_OK;
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webrtc::StreamConfig config (probe->info.rate, probe->info.channels,
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false);
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apm = self->apm;
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if (self->delay_agnostic)
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rec_time = GST_CLOCK_TIME_NONE;
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again:
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delay = gst_webrtc_echo_probe_read (probe, rec_time, (gpointer) &frame, &buf);
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delay = gst_webrtc_echo_probe_read (probe, rec_time, &buf);
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apm->set_stream_delay_ms (delay);
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g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR);
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if (delay < 0)
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goto done;
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if (frame.sample_rate_hz_ != self->info.rate) {
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if (probe->info.rate != self->info.rate) {
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GST_ELEMENT_ERROR (self, STREAM, FORMAT,
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("Echo Probe has rate %i , while the DSP is running at rate %i,"
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" use a caps filter to ensure those are the same.",
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frame.sample_rate_hz_, self->info.rate), (NULL));
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probe->info.rate, self->info.rate), (NULL));
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ret = GST_FLOW_ERROR;
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goto done;
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}
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if (buf) {
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webrtc::StreamConfig config (frame.sample_rate_hz_, frame.num_channels_,
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false);
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GstAudioBuffer abuf;
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float * const * data;
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gst_audio_buffer_map (&abuf, &self->info, buf, GST_MAP_READWRITE);
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if (probe->interleaved) {
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int16_t * const data = (int16_t * const) abuf.planes[0];
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gst_audio_buffer_map (&abuf, &self->info, buf, GST_MAP_READWRITE);
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data = (float * const *) abuf.planes;
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if ((err = apm->ProcessReverseStream (data, config, config, data)) < 0)
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GST_WARNING_OBJECT (self, "Reverse stream analyses failed: %s.",
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webrtc_error_to_string (err));
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gst_audio_buffer_unmap (&abuf);
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gst_buffer_replace (&buf, NULL);
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} else {
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if ((err = apm->AnalyzeReverseStream (&frame)) < 0)
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float * const * data = (float * const *) abuf.planes;
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if ((err = apm->ProcessReverseStream (data, config, config, data)) < 0)
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GST_WARNING_OBJECT (self, "Reverse stream analyses failed: %s.",
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webrtc_error_to_string (err));
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}
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if (self->delay_agnostic)
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goto again;
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gst_audio_buffer_unmap (&abuf);
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done:
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gst_object_unref (probe);
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@ -443,16 +431,14 @@ done:
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static void
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gst_webrtc_vad_post_activity (GstWebrtcDsp *self, GstBuffer *buffer,
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gboolean stream_has_voice)
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gboolean stream_has_voice, guint8 level)
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{
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GstClockTime timestamp = GST_BUFFER_PTS (buffer);
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GstBaseTransform *trans = GST_BASE_TRANSFORM_CAST (self);
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GstStructure *s;
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GstClockTime stream_time;
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GstAudioLevelMeta *meta;
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guint8 level;
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level = self->apm->level_estimator ()->RMS ();
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meta = gst_buffer_get_audio_level_meta (buffer);
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if (meta) {
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meta->voice_activity = stream_has_voice;
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@ -481,6 +467,7 @@ gst_webrtc_dsp_process_stream (GstWebrtcDsp * self,
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{
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GstAudioBuffer abuf;
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webrtc::AudioProcessing * apm = self->apm;
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webrtc::StreamConfig config (self->info.rate, self->info.channels, false);
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gint err;
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if (!gst_audio_buffer_map (&abuf, &self->info, buffer,
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@ -490,19 +477,10 @@ gst_webrtc_dsp_process_stream (GstWebrtcDsp * self,
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}
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if (self->interleaved) {
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webrtc::AudioFrame frame;
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frame.num_channels_ = self->info.channels;
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frame.sample_rate_hz_ = self->info.rate;
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frame.samples_per_channel_ = self->period_samples;
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memcpy (frame.data_, abuf.planes[0], self->period_size);
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err = apm->ProcessStream (&frame);
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if (err >= 0)
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memcpy (abuf.planes[0], frame.data_, self->period_size);
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int16_t * const data = (int16_t * const) abuf.planes[0];
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err = apm->ProcessStream (data, config, config, data);
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} else {
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float * const * data = (float * const *) abuf.planes;
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webrtc::StreamConfig config (self->info.rate, self->info.channels, false);
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err = apm->ProcessStream (data, config, config, data);
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}
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@ -511,10 +489,13 @@ gst_webrtc_dsp_process_stream (GstWebrtcDsp * self,
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webrtc_error_to_string (err));
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} else {
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if (self->voice_detection) {
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gboolean stream_has_voice = apm->voice_detection ()->stream_has_voice ();
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webrtc::AudioProcessingStats stats = apm->GetStatistics ();
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gboolean stream_has_voice = stats.voice_detected && *stats.voice_detected;
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// The meta takes the value as -dbov, so we negate
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guint8 level = stats.output_rms_dbfs ? (guint8) -(*stats.output_rms_dbfs) : 127;
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if (stream_has_voice != self->stream_has_voice)
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gst_webrtc_vad_post_activity (self, buffer, stream_has_voice);
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gst_webrtc_vad_post_activity (self, buffer, stream_has_voice, level);
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self->stream_has_voice = stream_has_voice;
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}
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@ -583,21 +564,9 @@ static gboolean
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gst_webrtc_dsp_start (GstBaseTransform * btrans)
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{
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GstWebrtcDsp *self = GST_WEBRTC_DSP (btrans);
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webrtc::Config config;
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GST_OBJECT_LOCK (self);
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config.Set < webrtc::ExtendedFilter >
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(new webrtc::ExtendedFilter (self->extended_filter));
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config.Set < webrtc::ExperimentalAgc >
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(new webrtc::ExperimentalAgc (self->experimental_agc, self->startup_min_volume));
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config.Set < webrtc::DelayAgnostic >
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(new webrtc::DelayAgnostic (self->delay_agnostic));
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/* TODO Intelligibility enhancer, Beamforming, etc. */
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self->apm = webrtc::AudioProcessing::Create (config);
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if (self->echo_cancel) {
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self->probe = gst_webrtc_acquire_echo_probe (self->probe_name);
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@ -618,10 +587,8 @@ static gboolean
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gst_webrtc_dsp_setup (GstAudioFilter * filter, const GstAudioInfo * info)
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{
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GstWebrtcDsp *self = GST_WEBRTC_DSP (filter);
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webrtc::AudioProcessing * apm;
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webrtc::ProcessingConfig pconfig;
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webrtc::AudioProcessing::Config config;
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GstAudioInfo probe_info = *info;
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gint err = 0;
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GST_LOG_OBJECT (self, "setting format to %s with %i Hz and %i channels",
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info->finfo->description, info->rate, info->channels);
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@ -633,7 +600,7 @@ gst_webrtc_dsp_setup (GstAudioFilter * filter, const GstAudioInfo * info)
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self->info = *info;
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self->interleaved = (info->layout == GST_AUDIO_LAYOUT_INTERLEAVED);
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apm = self->apm;
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self->apm = webrtc::AudioProcessingBuilder().Create();
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if (!self->interleaved)
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gst_planar_audio_adapter_configure (self->padapter, info);
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@ -642,8 +609,7 @@ gst_webrtc_dsp_setup (GstAudioFilter * filter, const GstAudioInfo * info)
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self->period_samples = info->rate / 100;
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self->period_size = self->period_samples * info->bpf;
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if (self->interleaved &&
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(webrtc::AudioFrame::kMaxDataSizeSamples * 2) < self->period_size)
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if (self->interleaved && (self->period_size > MAX_DATA_SIZE_SAMPLES * 2))
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goto period_too_big;
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if (self->probe) {
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@ -658,40 +624,31 @@ gst_webrtc_dsp_setup (GstAudioFilter * filter, const GstAudioInfo * info)
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GST_WEBRTC_ECHO_PROBE_UNLOCK (self->probe);
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}
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/* input stream */
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pconfig.streams[webrtc::ProcessingConfig::kInputStream] =
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webrtc::StreamConfig (info->rate, info->channels, false);
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/* output stream */
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pconfig.streams[webrtc::ProcessingConfig::kOutputStream] =
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webrtc::StreamConfig (info->rate, info->channels, false);
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/* reverse input stream */
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pconfig.streams[webrtc::ProcessingConfig::kReverseInputStream] =
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webrtc::StreamConfig (probe_info.rate, probe_info.channels, false);
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/* reverse output stream */
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pconfig.streams[webrtc::ProcessingConfig::kReverseOutputStream] =
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webrtc::StreamConfig (probe_info.rate, probe_info.channels, false);
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if ((err = apm->Initialize (pconfig)) < 0)
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goto initialize_failed;
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/* Setup Filters */
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||||
// 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);
|
||||
|
|
|
|||
|
|
@ -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;
|
||||
|
|
|
|||
|
|
@ -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__ */
|
||||
|
|
|
|||
|
|
@ -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
|
||||
|
|
|
|||
Loading…
Reference in a new issue