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webrtcdsp: Add WebRTC Audio Processing support

Browse source 
This DSP library can be used to enhance voice signal for real time
communication call. In implements multiple filters like noise reduction,
high pass filter, echo cancellation, automatic gain control, etc.

The webrtcdsp element can be used along, or with the help of the
webrtcechoprobe if echo cancellation is enabled. The echo probe should
be placed as close as possible to the audio sink, while the DSP is
generally place close to the audio capture. For local testing, one can
use an echo loop pipeline like the following:

  autoaudiosrc ! webrtcdsp ! webrtcechoprobe ! autoaudiosink

This pipeline should produce a single echo rather then repeated echo.
Those elements works if they are placed in the same top level pipeline.

https://bugzilla.gnome.org/show_bug.cgi?id=767800
This commit is contained in:
Nicolas Dufresne 2016-06-06 15:10:05 -04:00
parent cf6a517d41
commit 398f7059fc
7 changed files with 1293 additions and 2 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

14
configure.ac
View file

@ -3355,6 +3355,19 @@ AG_GST_CHECK_FEATURE(X265, [x265 plug-in], x265, [
AG_GST_PKG_CHECK_MODULES(X265, x265)
])
dnl *** WebRTC Audio Processing ***
dnl Note: We test for a specific version as there is no API stability
translit(dnm, m, l) AM_CONDITIONAL(USE_WEBRTCDSP, true)
AG_GST_CHECK_FEATURE(WEBRTCDSP, [WebRTC Audio Processing], webrtcdsp, [
AC_LANG_PUSH([C++])
old_CPPFLAGS=$CPPFLAGS
CPPFLAGS=-std=c++11 $CPPFLAGS
AG_GST_PKG_CHECK_MODULES(WEBRTCDSP, webrtc-audio-processing = 0.2)
AC_CHECK_HEADER([typeindex],[],[HAVE_WEBRTCDSP=no])
CPPFLAGS=$old_CPPFLAGS
AC_LANG_POP([C++])
])
else
dnl not building plugins with external dependencies,
@ -3751,6 +3764,7 @@ ext/x265/Makefile
ext/xvid/Makefile
ext/zbar/Makefile
ext/dtls/Makefile
ext/webrtcdsp/Makefile
po/Makefile.in
docs/Makefile
docs/plugins/Makefile

12
ext/Makefile.am
View file

@ -412,6 +412,12 @@ else
VULKAN_DIR=
endif
if USE_WEBRTCDSP
WEBRTCDSP_DIR=webrtcdsp
else
WEBRTCDSP_DIR=
endif
SUBDIRS=\
$(VOAACENC_DIR) \
$(ASSRENDER_DIR) \
@ -481,7 +487,8 @@ SUBDIRS=\
$(WEBP_DIR) \
$(X265_DIR) \
$(DTLS_DIR) \
$(VULKAN_DIR)
$(VULKAN_DIR) \
$(WEBRTCDSP_DIR)
DIST_SUBDIRS = \
assrender \
@ -549,6 +556,7 @@ DIST_SUBDIRS = \
webp \
x265 \
dtls \
vulkan
vulkan \
webrtcdsp
include $(top_srcdir)/common/parallel-subdirs.mak

17
ext/webrtcdsp/Makefile.am Normal file
View file

@ -0,0 +1,17 @@
plugin_LTLIBRARIES = libgstwebrtcdsp.la
libgstwebrtcdsp_la_SOURCES = gstwebrtcdsp.cpp gstwebrtcechoprobe.cpp
libgstwebrtcdsp_la_CXXFLAGS = \
-std=c++11 \
$(GST_CXXFLAGS) \
$(GST_BASE_CFLAGS) \
$(GST_PLUGINS_BASE_CFLAGS) \
$(WEBRTCDSP_CFLAGS)
libgstwebrtcdsp_la_LIBADD = \
-lgstaudio-$(GST_API_VERSION) \
$(GST_LIBS) $(GST_BASE_LIBS) \
$(GST_PLUGINS_BASE_LIBS) \
$(WEBRTCDSP_LIBS)
libgstwebrtcdsp_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
noinst_HEADERS = gstwebrtcdsp.h gstwebrtcechoprobe.h

858
ext/webrtcdsp/gstwebrtcdsp.cpp Normal file
View file

@ -0,0 +1,858 @@
/*
* WebRTC Audio Processing Elements
*
* Copyright 2016 Collabora Ltd
* @author: Nicolas Dufresne <nicolas.dufresne@collabora.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*
*/
/**
* SECTION:element-webrtcdsp
* @short_description: Audio Filter using WebRTC Audio Processing library
*
* A voice enhancement filter based on WebRTC Audio Processing library. This
* library provides a whide variety of enhancement algorithms. This element
* tries to enable as much as possible. The currently enabled enhancements are
* High Pass Filter, Echo Canceller, Noise Suppression, Automatic Gain Control,
* and some extended filters.
*
* While webrtcdsp element can be used alone, there is an exception for the
* echo canceller. The audio canceller need to be aware of the far end streams
* that are played to loud speakers. For this, you must place a webrtcechoprobe
* element at that far end. Note that the sample rate must match between
* webrtcdsp and the webrtechoprobe. Though, the number of channels can differ.
* The probe is found by the DSP element using it's object name. By default,
* webrtcdsp looks for webrtcechoprobe0, which means it just work if you have
* a single probe and DSP.
*
* The probe can only be used within the same top level GstPipeline.
* Additonally, to simplify the code, the probe element must be created
* before the DSP sink pad is activated. It does not need to be in any
* particular state and does not even need to be added to the pipeline yet.
*
* # Example launch line
*
* As a conveniance, the echo canceller can be tested using an echo loop. In
* this configuration, one would expect a single echo to be heard.
*
* |[
* gst-launch-1.0 pulsesrc ! webrtcdsp ! webrtcechoprobe ! pulsesink
* ]|
*
* In real environment, you'll place the probe before the playback, but only
* process the far end streams. The DSP should be placed as close as possible
* to the audio capture. The following pipeline is astracted and does not
* represent a real pipeline.
*
* |[
* gst-launch-1.0 far-end-src ! audio/x-raw,rate=48000 ! webrtcechoprobe ! pulsesink \
* pulsesrc ! audio/x-raw,rate=48000 ! webrtcdsp ! far-end-sink
* ]|
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstwebrtcdsp.h"
#include "gstwebrtcechoprobe.h"
#include <webrtc/modules/audio_processing/include/audio_processing.h>
#include <webrtc/modules/interface/module_common_types.h>
#include <webrtc/system_wrappers/include/trace.h>
GST_DEBUG_CATEGORY (webrtc_dsp_debug);
#define GST_CAT_DEFAULT (webrtc_dsp_debug)
static GstStaticPadTemplate gst_webrtc_dsp_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " GST_AUDIO_NE (S16) ", "
"layout = (string) interleaved, "
"rate = (int) { 48000, 32000, 16000, 8000 }, "
"channels = (int) [1, MAX]")
);
static GstStaticPadTemplate gst_webrtc_dsp_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " GST_AUDIO_NE (S16) ", "
"layout = (string) interleaved, "
"rate = (int) { 48000, 32000, 16000, 8000 }, "
"channels = (int) [1, MAX]")
);
typedef webrtc::EchoCancellation::SuppressionLevel GstWebrtcEchoSuppressionLevel;
#define GST_TYPE_WEBRTC_ECHO_SUPPRESSION_LEVEL \
(gst_webrtc_echo_suppression_level_get_type ())
static GType
gst_webrtc_echo_suppression_level_get_type (void)
{
static GType suppression_level_type = 0;
static const GEnumValue level_types[] = {
{webrtc::EchoCancellation::kLowSuppression, "Low Suppression", "low"},
{webrtc::EchoCancellation::kModerateSuppression,
"Moderate Suppression", "moderate"},
{webrtc::EchoCancellation::kHighSuppression, "high Suppression", "high"},
{0, NULL, NULL}
};
if (!suppression_level_type) {
suppression_level_type =
g_enum_register_static ("GstWebrtcEchoSuppressionLevel", level_types);
}
return suppression_level_type;
}
typedef webrtc::NoiseSuppression::Level GstWebrtcNoiseSuppressionLevel;
#define GST_TYPE_WEBRTC_NOISE_SUPPRESSION_LEVEL \
(gst_webrtc_noise_suppression_level_get_type ())
static GType
gst_webrtc_noise_suppression_level_get_type (void)
{
static GType suppression_level_type = 0;
static const GEnumValue level_types[] = {
{webrtc::NoiseSuppression::kLow, "Low Suppression", "low"},
{webrtc::NoiseSuppression::kModerate, "Moderate Suppression", "moderate"},
{webrtc::NoiseSuppression::kHigh, "High Suppression", "high"},
{webrtc::NoiseSuppression::kVeryHigh, "Very High Suppression",
"very-high"},
{0, NULL, NULL}
};
if (!suppression_level_type) {
suppression_level_type =
g_enum_register_static ("GstWebrtcNoiseSuppressionLevel", level_types);
}
return suppression_level_type;
}
enum
{
PROP_0,
PROP_PROBE,
PROP_HIGH_PASS_FILTER,
PROP_ECHO_CANCEL,
PROP_ECHO_SUPPRESSION_LEVEL,
PROP_NOISE_SUPPRESSION,
PROP_NOISE_SUPPRESSION_LEVEL,
PROP_GAIN_CONTROL,
PROP_EXPERIMENTAL_AGC,
PROP_EXTENDED_FILTER
};
/**
* GstWebrtcDSP:
*
* The adder object structure.
*/
struct _GstWebrtcDsp
{
GstAudioFilter element;
/* Protected by the object lock */
GstAudioInfo info;
guint period_size;
/* Protected by the stream lock */
GstClockTime timestamp;
GstAdapter *adapter;
webrtc::AudioProcessing * apm;
/* Protected by the object lock */
gchar *probe_name;
GstWebrtcEchoProbe *probe;
/* Properties */
gboolean high_pass_filter;
gboolean echo_cancel;
webrtc::EchoCancellation::SuppressionLevel echo_suppression_level;
gboolean noise_suppression;
webrtc::NoiseSuppression::Level noise_suppression_level;
gboolean gain_control;
gboolean experimental_agc;
gboolean extended_filter;
};
G_DEFINE_TYPE (GstWebrtcDsp, gst_webrtc_dsp, GST_TYPE_AUDIO_FILTER);
static const gchar *
webrtc_error_to_string (gint err)
{
const gchar *str = "unkown error";
switch (err) {
case webrtc::AudioProcessing::kNoError:
str = "success";
break;
case webrtc::AudioProcessing::kUnspecifiedError:
str = "unspecified error";
break;
case webrtc::AudioProcessing::kCreationFailedError:
str = "creating failed";
break;
case webrtc::AudioProcessing::kUnsupportedComponentError:
str = "unsupported component";
break;
case webrtc::AudioProcessing::kUnsupportedFunctionError:
str = "unsupported function";
break;
case webrtc::AudioProcessing::kNullPointerError:
str = "null pointer";
break;
case webrtc::AudioProcessing::kBadParameterError:
str = "bad parameter";
break;
case webrtc::AudioProcessing::kBadSampleRateError:
str = "bad sample rate";
break;
case webrtc::AudioProcessing::kBadDataLengthError:
str = "bad data length";
break;
case webrtc::AudioProcessing::kBadNumberChannelsError:
str = "bad number of channels";
break;
case webrtc::AudioProcessing::kFileError:
str = "file IO error";
break;
case webrtc::AudioProcessing::kStreamParameterNotSetError:
str = "stream parameter not set";
break;
case webrtc::AudioProcessing::kNotEnabledError:
str = "not enabled";
break;
default:
break;
}
return str;
}
/* with probe object lock */
static gboolean
gst_webrtc_dsp_sync_reverse_stream (GstWebrtcDsp * self,
GstWebrtcEchoProbe * probe)
{
GstClockTime probe_timestamp;
GstClockTimeDiff diff;
guint64 distance;
probe_timestamp = gst_adapter_prev_pts (probe->adapter, &distance);
if (!GST_CLOCK_TIME_IS_VALID (probe_timestamp)) {
GST_WARNING_OBJECT (self,
"Echo Probe is handling buffer without timestamp.");
return FALSE;
}
if (gst_adapter_pts_at_discont (probe->adapter) == probe_timestamp) {
if (distance == 0)
probe->synchronized = FALSE;
}
if (probe->synchronized)
return TRUE;
if (gst_adapter_available (probe->adapter) < probe->period_size
|| probe->latency == -1) {
GST_TRACE_OBJECT (self, "Echo Probe not ready yet");
return FALSE;
}
if (self->info.rate != probe->info.rate
|| self->info.channels != probe->info.channels) {
GST_WARNING_OBJECT (self,
"Echo Probe has rate %i and %i channels, while "
"the DSP is running at rate %i with %i channels, use a caps filter to "
"ensure those are the same.", probe->info.rate, probe->info.channels,
self->info.rate, self->info.channels);
return FALSE;
}
probe_timestamp += gst_util_uint64_scale_int (distance / probe->info.bpf,
GST_SECOND, probe->info.rate);
probe_timestamp += probe->latency;
diff = GST_CLOCK_DIFF (probe_timestamp, self->timestamp);
if (diff < 0) {
GST_TRACE_OBJECT (self,
"Echo cancellation will start in in %" GST_TIME_FORMAT,
GST_TIME_ARGS (-diff));
return FALSE;
}
distance = gst_util_uint64_scale_int ((guint64) diff,
probe->info.rate * probe->info.bpf, GST_SECOND);
if (gst_adapter_available (probe->adapter) < distance) {
GST_TRACE_OBJECT (self, "Not enough data to synchronize for now.");
return FALSE;
}
gst_adapter_flush (probe->adapter, (gsize) distance);
probe->synchronized = TRUE;
GST_DEBUG_OBJECT (probe, "Echo Probe is now synchronized");
return TRUE;
}
static void
gst_webrtc_dsp_analyze_reverse_stream (GstWebrtcDsp * self)
{
GstWebrtcEchoProbe *probe = NULL;
webrtc::AudioProcessing * apm;
webrtc::AudioFrame frame;
gint err;
GST_OBJECT_LOCK (self);
if (self->echo_cancel && self->probe)
probe = GST_WEBRTC_ECHO_PROBE (g_object_ref (self->probe));
GST_OBJECT_UNLOCK (self);
if (!probe)
return;
apm = self->apm;
GST_WEBRTC_ECHO_PROBE_LOCK (probe);
if (gst_adapter_available (probe->adapter) < probe->period_size) {
GST_LOG_OBJECT (self, "No echo data yet...");
goto beach;
}
if (!gst_webrtc_dsp_sync_reverse_stream (self, probe))
goto beach;
frame.num_channels_ = probe->info.channels;
frame.sample_rate_hz_ = probe->info.rate;
frame.samples_per_channel_ = probe->period_size / probe->info.bpf;
gst_adapter_copy (probe->adapter, (guint8 *) frame.data_, 0,
probe->period_size);
gst_adapter_flush (probe->adapter, self->period_size);
if ((err = apm->AnalyzeReverseStream (&frame)) < 0)
GST_WARNING_OBJECT (self, "Reverse stream analyses failed: %s.",
webrtc_error_to_string (err));
beach:
GST_WEBRTC_ECHO_PROBE_UNLOCK (probe);
gst_object_unref (probe);
}
static GstBuffer *
gst_webrtc_dsp_process_stream (GstWebrtcDsp * self)
{
GstBuffer *buffer;
GstMapInfo info;
webrtc::AudioProcessing * apm = self->apm;
webrtc::AudioFrame frame;
GstClockTime timestamp;
guint64 distance;
gint err;
frame.num_channels_ = self->info.channels;
frame.sample_rate_hz_ = self->info.rate;
frame.samples_per_channel_ = self->period_size / self->info.bpf;
timestamp = gst_adapter_prev_pts (self->adapter, &distance);
timestamp += gst_util_uint64_scale_int (distance / self->info.bpf,
GST_SECOND, self->info.rate);
buffer = gst_adapter_take_buffer (self->adapter, self->period_size);
if (!gst_buffer_map (buffer, &info, (GstMapFlags) GST_MAP_READWRITE)) {
gst_buffer_unref (buffer);
return NULL;
}
memcpy (frame.data_, info.data, self->period_size);
/* We synchronize in GStreamer */
apm->set_stream_delay_ms (5);
if ((err = apm->ProcessStream (&frame)) < 0) {
GST_WARNING_OBJECT (self, "Failed to filter the audio: %s.",
webrtc_error_to_string (err));
} else {
memcpy (info.data, frame.data_, self->period_size);
}
gst_buffer_unmap (buffer, &info);
GST_BUFFER_PTS (buffer) = timestamp;
GST_BUFFER_DURATION (buffer) = 10 * GST_MSECOND;
if (gst_adapter_pts_at_discont (self->adapter) == timestamp)
GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
else
GST_BUFFER_FLAG_UNSET (buffer, GST_BUFFER_FLAG_DISCONT);
self->timestamp = timestamp;
return buffer;
}
static GstFlowReturn
gst_webrtc_dsp_submit_input_buffer (GstBaseTransform * btrans,
gboolean is_discont, GstBuffer * buffer)
{
GstWebrtcDsp *self = GST_WEBRTC_DSP (btrans);
buffer = gst_buffer_make_writable (buffer);
GST_BUFFER_PTS (buffer) = gst_segment_to_running_time (&btrans->segment,
GST_FORMAT_TIME, GST_BUFFER_PTS (buffer));
if (!GST_CLOCK_TIME_IS_VALID (self->timestamp))
self->timestamp = GST_BUFFER_PTS (buffer);
if (is_discont) {
GST_OBJECT_LOCK (self);
if (self->echo_cancel && self->probe) {
GST_WEBRTC_ECHO_PROBE_LOCK (self->probe);
self->probe->synchronized = FALSE;
GST_WEBRTC_ECHO_PROBE_UNLOCK (self->probe);
}
GST_OBJECT_UNLOCK (self);
gst_adapter_clear (self->adapter);
}
gst_adapter_push (self->adapter, buffer);
return GST_FLOW_OK;
}
static GstFlowReturn
gst_webrtc_dsp_generate_output (GstBaseTransform * btrans, GstBuffer ** outbuf)
{
GstWebrtcDsp *self = GST_WEBRTC_DSP (btrans);
if (gst_adapter_available (self->adapter) >= self->period_size) {
gst_webrtc_dsp_analyze_reverse_stream (self);
*outbuf = gst_webrtc_dsp_process_stream (self);
} else {
*outbuf = NULL;
}
return GST_FLOW_OK;
}
static gboolean
gst_webrtc_dsp_start (GstBaseTransform * btrans)
{
GstWebrtcDsp *self = GST_WEBRTC_DSP (btrans);
webrtc::Config config;
GST_OBJECT_LOCK (self);
config.Set < webrtc::ExtendedFilter >
(new webrtc::ExtendedFilter (self->extended_filter));
config.Set < webrtc::ExperimentalAgc >
(new webrtc::ExperimentalAgc (self->experimental_agc));
/* TODO Intelligibility enhancer, Beamforming, etc. */
self->apm = webrtc::AudioProcessing::Create (config);
if (self->echo_cancel) {
self->probe = gst_webrtc_acquire_echo_probe (self->probe_name);
if (self->probe == NULL) {
GST_OBJECT_UNLOCK (self);
GST_ELEMENT_ERROR (self, RESOURCE, NOT_FOUND,
("No echo probe with name %s found.", self->probe_name), (NULL));
return FALSE;
}
}
GST_OBJECT_UNLOCK (self);
return TRUE;
}
static gboolean
gst_webrtc_dsp_setup (GstAudioFilter * filter, const GstAudioInfo * info)
{
GstWebrtcDsp *self = GST_WEBRTC_DSP (filter);
webrtc::AudioProcessing * apm;
webrtc::ProcessingConfig pconfig;
GstAudioInfo probe_info = *info;
gint err = 0;
GST_LOG_OBJECT (self, "setting format to %s with %i Hz and %i channels",
info->finfo->description, info->rate, info->channels);
GST_OBJECT_LOCK (self);
gst_adapter_clear (self->adapter);
self->info = *info;
apm = self->apm;
/* WebRTC library works with 10ms buffers, compute once this size */
self->period_size = info->bpf * info->rate / 100;
if ((webrtc::AudioFrame::kMaxDataSizeSamples * 2) < self->period_size)
goto period_too_big;
if (self->probe) {
GST_WEBRTC_ECHO_PROBE_LOCK (self->probe);
if (self->probe->info.rate != 0) {
if (self->probe->info.rate != info->rate)
goto probe_has_wrong_rate;
probe_info = self->probe->info;
}
self->probe->synchronized = FALSE;
GST_WEBRTC_ECHO_PROBE_UNLOCK (self->probe);
}
pconfig = {
/* input stream */
webrtc::StreamConfig (info->rate, info->channels, false),
/* output stream */
webrtc::StreamConfig (info->rate, info->channels, false),
/* reverse input stream */
webrtc::StreamConfig (probe_info.rate, probe_info.channels, false),
/* reverse output stream */
webrtc::StreamConfig (probe_info.rate, probe_info.channels, false),
};
if ((err = apm->Initialize (pconfig)) < 0)
goto initialize_failed;
/* Setup Filters */
if (self->high_pass_filter) {
GST_DEBUG_OBJECT (self, "Enabling High Pass filter");
apm->high_pass_filter ()->Enable (true);
}
if (self->echo_cancel && self->probe) {
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);
}
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);
}
if (self->gain_control) {
GST_DEBUG_OBJECT (self, "Enabling Digital Gain Control");
apm->gain_control ()->set_mode (webrtc::GainControl::kAdaptiveDigital);
apm->gain_control ()->Enable (true);
}
GST_OBJECT_UNLOCK (self);
return TRUE;
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), "
"reduce the number of channels or the rate.",
webrtc::AudioFrame::kMaxDataSizeSamples, self->period_size / 2);
return FALSE;
probe_has_wrong_rate:
GST_WEBRTC_ECHO_PROBE_UNLOCK (self->probe);
GST_OBJECT_UNLOCK (self);
GST_ELEMENT_ERROR (self, STREAM, FORMAT,
("Echo Probe has rate %i , while the DSP is running at rate %i,"
" use a caps filter to ensure those are the same.",
probe_info.rate, 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
gst_webrtc_dsp_stop (GstBaseTransform * btrans)
{
GstWebrtcDsp *self = GST_WEBRTC_DSP (btrans);
GST_OBJECT_LOCK (self);
gst_adapter_clear (self->adapter);
if (self->probe) {
gst_webrtc_release_echo_probe (self->probe);
self->probe = NULL;
}
delete self->apm;
self->apm = NULL;
GST_OBJECT_UNLOCK (self);
return TRUE;
}
static void
gst_webrtc_dsp_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstWebrtcDsp *self = GST_WEBRTC_DSP (object);
GST_OBJECT_LOCK (self);
switch (prop_id) {
case PROP_PROBE:
g_free (self->probe_name);
self->probe_name = g_value_dup_string (value);
break;
case PROP_HIGH_PASS_FILTER:
self->high_pass_filter = g_value_get_boolean (value);
break;
case PROP_ECHO_CANCEL:
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);
break;
case PROP_NOISE_SUPPRESSION_LEVEL:
self->noise_suppression_level =
(GstWebrtcNoiseSuppressionLevel) g_value_get_enum (value);
break;
case PROP_GAIN_CONTROL:
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;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_OBJECT_UNLOCK (self);
}
static void
gst_webrtc_dsp_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstWebrtcDsp *self = GST_WEBRTC_DSP (object);
GST_OBJECT_LOCK (self);
switch (prop_id) {
case PROP_PROBE:
g_value_set_string (value, self->probe_name);
break;
case PROP_HIGH_PASS_FILTER:
g_value_set_boolean (value, self->high_pass_filter);
break;
case PROP_ECHO_CANCEL:
g_value_set_boolean (value, self->echo_cancel);
break;
case PROP_ECHO_SUPPRESSION_LEVEL:
g_value_set_enum (value, self->echo_suppression_level);
break;
case PROP_NOISE_SUPPRESSION:
g_value_set_boolean (value, self->noise_suppression);
break;
case PROP_NOISE_SUPPRESSION_LEVEL:
g_value_set_enum (value, self->noise_suppression_level);
break;
case PROP_GAIN_CONTROL:
g_value_set_boolean (value, self->gain_control);
break;
case PROP_EXPERIMENTAL_AGC:
g_value_set_boolean (value, self->experimental_agc);
break;
case PROP_EXTENDED_FILTER:
g_value_set_boolean (value, self->extended_filter);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_OBJECT_UNLOCK (self);
}
static void
gst_webrtc_dsp_finalize (GObject * object)
{
GstWebrtcDsp *self = GST_WEBRTC_DSP (object);
gst_object_unref (self->adapter);
g_free (self->probe_name);
G_OBJECT_CLASS (gst_webrtc_dsp_parent_class)->finalize (object);
}
static void
gst_webrtc_dsp_init (GstWebrtcDsp * self)
{
self->adapter = gst_adapter_new ();
gst_audio_info_init (&self->info);
}
static void
gst_webrtc_dsp_class_init (GstWebrtcDspClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstBaseTransformClass *btrans_class = GST_BASE_TRANSFORM_CLASS (klass);
GstAudioFilterClass *audiofilter_class = GST_AUDIO_FILTER_CLASS (klass);
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_webrtc_dsp_finalize);
gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_webrtc_dsp_set_property);
gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_webrtc_dsp_get_property);
btrans_class->passthrough_on_same_caps = FALSE;
btrans_class->start = GST_DEBUG_FUNCPTR (gst_webrtc_dsp_start);
btrans_class->stop = GST_DEBUG_FUNCPTR (gst_webrtc_dsp_stop);
btrans_class->submit_input_buffer =
GST_DEBUG_FUNCPTR (gst_webrtc_dsp_submit_input_buffer);
btrans_class->generate_output =
GST_DEBUG_FUNCPTR (gst_webrtc_dsp_generate_output);
audiofilter_class->setup = GST_DEBUG_FUNCPTR (gst_webrtc_dsp_setup);
gst_element_class_add_static_pad_template (element_class,
&gst_webrtc_dsp_src_template);
gst_element_class_add_static_pad_template (element_class,
&gst_webrtc_dsp_sink_template);
gst_element_class_set_static_metadata (element_class,
"Voice Processor (AGC, AEC, filters, etc.)",
"Generic/Audio",
"Pre-processes voice with WebRTC Audio Processing Library",
"Nicolas Dufresne <nicolas.dufresne@collabora.com>");
g_object_class_install_property (gobject_class,
PROP_PROBE,
g_param_spec_string ("probe", "Echo Probe",
"The name of the webrtcechoprobe element that record the audio being "
"played through loud speakers. Must be set before PAUSED state.",
"webrtcechoprobe0",
(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
G_PARAM_CONSTRUCT)));
g_object_class_install_property (gobject_class,
PROP_HIGH_PASS_FILTER,
g_param_spec_boolean ("high-pass-filter", "High Pass Filter",
"Enable or disable high pass filtering", TRUE,
(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
G_PARAM_CONSTRUCT)));
g_object_class_install_property (gobject_class,
PROP_ECHO_CANCEL,
g_param_spec_boolean ("echo-cancel", "Echo Cancel",
"Enable or disable echo canceller, note that it will be disabled if "
"no webrtcechoprobe has been found", TRUE,
(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
G_PARAM_CONSTRUCT)));
g_object_class_install_property (gobject_class,
PROP_ECHO_SUPPRESSION_LEVEL,
g_param_spec_enum ("echo-suppression-level", "Echo Suppression Level",
"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,
(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
G_PARAM_CONSTRUCT)));
g_object_class_install_property (gobject_class,
PROP_NOISE_SUPPRESSION,
g_param_spec_boolean ("noise-suppression", "Noise Suppression",
"Enable or disable noise suppression", TRUE,
(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
G_PARAM_CONSTRUCT)));
g_object_class_install_property (gobject_class,
PROP_NOISE_SUPPRESSION_LEVEL,
g_param_spec_enum ("noise-suppression-level", "Noise Suppression Level",
"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,
(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
G_PARAM_CONSTRUCT)));
g_object_class_install_property (gobject_class,
PROP_GAIN_CONTROL,
g_param_spec_boolean ("gain-control", "Gain Control",
"Enable or disable automatic digital gain control",
TRUE, (GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
G_PARAM_CONSTRUCT)));
g_object_class_install_property (gobject_class,
PROP_EXPERIMENTAL_AGC,
g_param_spec_boolean ("experimental-agc", "Experimental AGC",
"Enable or disable experimental automatic gain control.",
FALSE, (GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
G_PARAM_CONSTRUCT)));
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)));
}
static gboolean
plugin_init (GstPlugin * plugin)
{
GST_DEBUG_CATEGORY_INIT
(webrtc_dsp_debug, "webrtcdsp", 0, "libwebrtcdsp wrapping elements");
if (!gst_element_register (plugin, "webrtcdsp", GST_RANK_NONE,
GST_TYPE_WEBRTC_DSP)) {
return FALSE;
}
if (!gst_element_register (plugin, "webrtcechoprobe", GST_RANK_NONE,
GST_TYPE_WEBRTC_ECHO_PROBE)) {
return FALSE;
}
return TRUE;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
webrtcdsp,
"Voice pre-processing using WebRTC Audio Processing Library",
plugin_init, VERSION, "LGPL", "WebRTCDsp", "http://git.collabora.com")

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/*
* WebRTC Audio Processing Elements
*
* Copyright 2016 Collabora Ltd
* @author: Nicolas Dufresne <nicolas.dufresne@collabora.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*
*/
#ifndef __GST_WEBRTC_DSP_H__
#define __GST_WEBRTC_DSP_H__
#include <gst/gst.h>
#include <gst/base/gstadapter.h>
#include <gst/base/gstbasetransform.h>
#include <gst/audio/audio.h>
G_BEGIN_DECLS
#define GST_TYPE_WEBRTC_DSP (gst_webrtc_dsp_get_type())
#define GST_WEBRTC_DSP(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_DSP,GstWebrtcDsp))
#define GST_IS_WEBRTC_DSP(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_DSP))
#define GST_WEBRTC_DSP_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_DSP,GstWebrtcDspClass))
#define GST_IS_WEBRTC_DSP_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_DSP))
#define GST_WEBRTC_DSP_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_DSP,GstWebrtcDspClass))
typedef struct _GstWebrtcDsp GstWebrtcDsp;
typedef struct _GstWebrtcDspClass GstWebrtcDspClass;
struct _GstWebrtcDspClass
{
GstAudioFilterClass parent_class;
};
GType gst_webrtc_dsp_get_type (void);
G_END_DECLS
#endif /* __GST_WEBRTC_DSP_H__ */

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/*
* WebRTC Audio Processing Elements
*
* Copyright 2016 Collabora Ltd
* @author: Nicolas Dufresne <nicolas.dufresne@collabora.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*
*/
/**
* SECTION:element-webrtcechoprobe
*
* This echo probe is to be used with the webrtcdsp element. See #gst-plugins-bad-plugins-webrtcdsp
* documentation for more details.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstwebrtcechoprobe.h"
#include <webrtc/modules/interface/module_common_types.h>
#include <gst/audio/audio.h>
GST_DEBUG_CATEGORY_EXTERN (webrtc_dsp_debug);
#define GST_CAT_DEFAULT (webrtc_dsp_debug)
#define MAX_ADAPTER_SIZE (1*1024*1024)
static GstStaticPadTemplate gst_webrtc_echo_probe_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " GST_AUDIO_NE (S16) ", "
"layout = (string) interleaved, "
"rate = (int) { 48000, 32000, 16000, 8000 }, "
"channels = (int) [1, MAX]")
);
static GstStaticPadTemplate gst_webrtc_echo_probe_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " GST_AUDIO_NE (S16) ", "
"layout = (string) interleaved, "
"rate = (int) { 48000, 32000, 16000, 8000 }, "
"channels = (int) [1, MAX]")
);
G_LOCK_DEFINE_STATIC (gst_aec_probes);
static GList *gst_aec_probes = NULL;
G_DEFINE_TYPE (GstWebrtcEchoProbe, gst_webrtc_echo_probe,
GST_TYPE_AUDIO_FILTER);
static gboolean
gst_webrtc_echo_probe_setup (GstAudioFilter * filter, const GstAudioInfo * info)
{
GstWebrtcEchoProbe *self = GST_WEBRTC_ECHO_PROBE (filter);
GST_LOG_OBJECT (self, "setting format to %s with %i Hz and %i channels",
info->finfo->description, info->rate, info->channels);
GST_WEBRTC_ECHO_PROBE_LOCK (self);
self->info = *info;
self->synchronized = FALSE;
/* WebRTC library works with 10ms buffers, compute once this size */
self->period_size = info->bpf * info->rate / 100;
if ((webrtc::AudioFrame::kMaxDataSizeSamples * 2) < self->period_size)
goto period_too_big;
GST_WEBRTC_ECHO_PROBE_UNLOCK (self);
return TRUE;
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), "
"reduce the number of channels or the rate.",
webrtc::AudioFrame::kMaxDataSizeSamples, self->period_size / 2);
return FALSE;
}
static gboolean
gst_webrtc_echo_probe_stop (GstBaseTransform * btrans)
{
GstWebrtcEchoProbe *self = GST_WEBRTC_ECHO_PROBE (btrans);
GST_WEBRTC_ECHO_PROBE_LOCK (self);
gst_adapter_clear (self->adapter);
GST_WEBRTC_ECHO_PROBE_UNLOCK (self);
return TRUE;
}
static gboolean
gst_webrtc_echo_probe_src_event (GstBaseTransform * btrans, GstEvent * event)
{
GstBaseTransformClass *klass;
GstWebrtcEchoProbe *self = GST_WEBRTC_ECHO_PROBE (btrans);
GstClockTime latency;
klass = GST_BASE_TRANSFORM_CLASS (gst_webrtc_echo_probe_parent_class);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_LATENCY:
gst_event_parse_latency (event, &latency);
GST_WEBRTC_ECHO_PROBE_LOCK (self);
self->latency = latency;
GST_WEBRTC_ECHO_PROBE_UNLOCK (self);
GST_DEBUG_OBJECT (self, "We have a latency of %" GST_TIME_FORMAT,
GST_TIME_ARGS (latency));
break;
default:
break;
}
return klass->src_event (btrans, event);
}
static GstFlowReturn
gst_webrtc_echo_probe_transform_ip (GstBaseTransform * btrans,
GstBuffer * buffer)
{
GstWebrtcEchoProbe *self = GST_WEBRTC_ECHO_PROBE (btrans);
GstBuffer *newbuf = NULL;
GST_WEBRTC_ECHO_PROBE_LOCK (self);
newbuf = gst_buffer_copy (buffer);
/* Moves the buffer timestamp to be in Running time */
GST_BUFFER_PTS (newbuf) = gst_segment_to_running_time (&btrans->segment,
GST_FORMAT_TIME, GST_BUFFER_PTS (buffer));
gst_adapter_push (self->adapter, newbuf);
if (gst_adapter_available (self->adapter) > MAX_ADAPTER_SIZE)
gst_adapter_flush (self->adapter,
gst_adapter_available (self->adapter) - MAX_ADAPTER_SIZE);
GST_WEBRTC_ECHO_PROBE_UNLOCK (self);
return GST_FLOW_OK;
}
static void
gst_webrtc_echo_probe_finalize (GObject * object)
{
GstWebrtcEchoProbe *self = GST_WEBRTC_ECHO_PROBE (object);
G_LOCK (gst_aec_probes);
gst_aec_probes = g_list_remove (gst_aec_probes, self);
G_UNLOCK (gst_aec_probes);
gst_object_unref (self->adapter);
self->adapter = NULL;
G_OBJECT_CLASS (gst_webrtc_echo_probe_parent_class)->finalize (object);
}
static void
gst_webrtc_echo_probe_init (GstWebrtcEchoProbe * self)
{
self->adapter = gst_adapter_new ();
gst_audio_info_init (&self->info);
g_mutex_init (&self->lock);
self->latency = -1;
G_LOCK (gst_aec_probes);
gst_aec_probes = g_list_prepend (gst_aec_probes, self);
G_UNLOCK (gst_aec_probes);
}
static void
gst_webrtc_echo_probe_class_init (GstWebrtcEchoProbeClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstBaseTransformClass *btrans_class = GST_BASE_TRANSFORM_CLASS (klass);
GstAudioFilterClass *audiofilter_class = GST_AUDIO_FILTER_CLASS (klass);
gobject_class->finalize = gst_webrtc_echo_probe_finalize;
btrans_class->passthrough_on_same_caps = TRUE;
btrans_class->src_event = GST_DEBUG_FUNCPTR (gst_webrtc_echo_probe_src_event);
btrans_class->transform_ip =
GST_DEBUG_FUNCPTR (gst_webrtc_echo_probe_transform_ip);
btrans_class->stop = GST_DEBUG_FUNCPTR (gst_webrtc_echo_probe_stop);
audiofilter_class->setup = GST_DEBUG_FUNCPTR (gst_webrtc_echo_probe_setup);
gst_element_class_add_static_pad_template (element_class,
&gst_webrtc_echo_probe_src_template);
gst_element_class_add_static_pad_template (element_class,
&gst_webrtc_echo_probe_sink_template);
gst_element_class_set_static_metadata (element_class,
"Accoustic Echo Canceller probe",
"Generic/Audio",
"Gathers playback buffers for webrtcdsp",
"Nicolas Dufresne <nicolas.dufrsesne@collabora.com>");
}
GstWebrtcEchoProbe *
gst_webrtc_acquire_echo_probe (const gchar * name)
{
GstWebrtcEchoProbe *ret = NULL;
GList *l;
G_LOCK (gst_aec_probes);
for (l = gst_aec_probes; l; l = l->next) {
GstWebrtcEchoProbe *probe = GST_WEBRTC_ECHO_PROBE (l->data);
GST_WEBRTC_ECHO_PROBE_LOCK (probe);
if (!probe->acquired && g_strcmp0 (GST_OBJECT_NAME (probe), name) == 0) {
probe->acquired = TRUE;
ret = GST_WEBRTC_ECHO_PROBE (gst_object_ref (probe));
GST_WEBRTC_ECHO_PROBE_UNLOCK (probe);
break;
}
GST_WEBRTC_ECHO_PROBE_UNLOCK (probe);
}
G_UNLOCK (gst_aec_probes);
return ret;
}
void
gst_webrtc_release_echo_probe (GstWebrtcEchoProbe * probe)
{
GST_WEBRTC_ECHO_PROBE_LOCK (probe);
probe->acquired = FALSE;
GST_WEBRTC_ECHO_PROBE_UNLOCK (probe);
gst_object_unref (probe);
}

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/*
* WebRTC Audio Processing Elements
*
* Copyright 2016 Collabora Ltd
* @author: Nicolas Dufresne <nicolas.dufresne@collabora.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*
*/
#ifndef __GST_WEBRTC_ECHO_PROBE_H__
#define __GST_WEBRTC_ECHO_PROBE_H__
#include <gst/gst.h>
#include <gst/base/gstadapter.h>
#include <gst/base/gstbasetransform.h>
#include <gst/audio/audio.h>
G_BEGIN_DECLS
#define GST_TYPE_WEBRTC_ECHO_PROBE (gst_webrtc_echo_probe_get_type())
#define GST_WEBRTC_ECHO_PROBE(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_ECHO_PROBE,GstWebrtcEchoProbe))
#define GST_IS_WEBRTC_ECHO_PROBE(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_ECHO_PROBE))
#define GST_WEBRTC_ECHO_PROBE_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_ECHO_PROBE,GstWebrtcEchoProbeClass))
#define GST_IS_WEBRTC_ECHO_PROBE_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_ECHO_PROBE))
#define GST_WEBRTC_ECHO_PROBE_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_ECHO_PROBE,GstWebrtcEchoProbeClass))
#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)
typedef struct _GstWebrtcEchoProbe GstWebrtcEchoProbe;
typedef struct _GstWebrtcEchoProbeClass GstWebrtcEchoProbeClass;
/**
* GstWebrtcEchoProbe:
*
* The adder object structure.
*/
struct _GstWebrtcEchoProbe
{
GstAudioFilter parent;
/* This lock is required as the DSP may need to lock itself using it's
* object lock and also lock the probe. The natural order for the DSP is
* to lock the DSP and then the echo probe. If we where using the probe
* object lock, we'd be racing with GstBin which will lock sink to src,
* and may accidently reverse the order. */
GMutex lock;
/* Protected by the lock */
GstAudioInfo info;
guint period_size;
gint latency;
gboolean synchronized;
GstSegment segment;
GstAdapter *adapter;
/* Private */
gboolean acquired;
};
struct _GstWebrtcEchoProbeClass
{
GstAudioFilterClass parent_class;
};
GType gst_webrtc_echo_probe_get_type (void);
GstWebrtcEchoProbe *gst_webrtc_acquire_echo_probe (const gchar * name);
void gst_webrtc_release_echo_probe (GstWebrtcEchoProbe * probe);
G_END_DECLS
#endif /* __GST_WEBRTC_ECHO_PROBE_H__ */