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gstreamer/ext/webrtcdsp/gstwebrtcdsp.cpp

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webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
/*
* 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,
webrtcdsp: Add delay-agnostic property In this mode, we let WebRTC Audio Processing figure-out the delay. This is useful when the latency reported by the stack cannot be trusted. Note that in this mode, the leaking of echo during packet lost is much worst. It is recommanded to use PLC (e.g. spanplc, or opus built-in plc). In this mode, we don't do any synchronization. Instead, we simply process all the available reverse stream data as it comes.
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PROP_EXTENDED_FILTER,
PROP_DELAY_AGNOSTIC
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
};
/**
* GstWebrtcDSP:
*
* The adder object structure.
*/
struct _GstWebrtcDsp
{
GstAudioFilter element;
/* Protected by the object lock */
GstAudioInfo info;
guint period_size;
/* Protected by the stream lock */
GstAdapter *adapter;
webrtcdsp: Style fix
2016-06-23 01:44:35 +00:00
webrtc::AudioProcessing * apm;
webrtcdsp: Add WebRTC Audio Processing support 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
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/* 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;
webrtcdsp: Add delay-agnostic property In this mode, we let WebRTC Audio Processing figure-out the delay. This is useful when the latency reported by the stack cannot be trusted. Note that in this mode, the leaking of echo during packet lost is much worst. It is recommanded to use PLC (e.g. spanplc, or opus built-in plc). In this mode, we don't do any synchronization. Instead, we simply process all the available reverse stream data as it comes.
2016-07-14 03:00:51 +00:00
gboolean delay_agnostic;
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
};
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;
}
webrtcdsp: Rewrite echo data synchronization The previous code would run out of sync if there was packet lost or clock skews. When that happened, the echo cancellation feature would completely stop working. As this is crucial for audio calls, this patch re-implement synchronization completely. Instead of letting it drift until next discont, we now synchronize against the record data at every iteration. This way we simply never let the stream drift for longer then 10ms period. We also shorter the delay by using the latency up the probe (basically excluding the sink latency. This is a decent delay to avoid starving in the probe queue. https://bugzilla.gnome.org/show_bug.cgi?id=768009
2016-06-29 19:56:47 +00:00
static GstBuffer *
gst_webrtc_dsp_take_buffer (GstWebrtcDsp * self)
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
{
webrtcdsp: Rewrite echo data synchronization The previous code would run out of sync if there was packet lost or clock skews. When that happened, the echo cancellation feature would completely stop working. As this is crucial for audio calls, this patch re-implement synchronization completely. Instead of letting it drift until next discont, we now synchronize against the record data at every iteration. This way we simply never let the stream drift for longer then 10ms period. We also shorter the delay by using the latency up the probe (basically excluding the sink latency. This is a decent delay to avoid starving in the probe queue. https://bugzilla.gnome.org/show_bug.cgi?id=768009
2016-06-29 19:56:47 +00:00
GstBuffer *buffer;
GstClockTime timestamp;
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
guint64 distance;
webrtcdsp: Rewrite echo data synchronization The previous code would run out of sync if there was packet lost or clock skews. When that happened, the echo cancellation feature would completely stop working. As this is crucial for audio calls, this patch re-implement synchronization completely. Instead of letting it drift until next discont, we now synchronize against the record data at every iteration. This way we simply never let the stream drift for longer then 10ms period. We also shorter the delay by using the latency up the probe (basically excluding the sink latency. This is a decent delay to avoid starving in the probe queue. https://bugzilla.gnome.org/show_bug.cgi?id=768009
2016-06-29 19:56:47 +00:00
timestamp = gst_adapter_prev_pts (self->adapter, &distance);
timestamp += gst_util_uint64_scale_int (distance / self->info.bpf,
GST_SECOND, self->info.rate);
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
webrtcdsp: Rewrite echo data synchronization The previous code would run out of sync if there was packet lost or clock skews. When that happened, the echo cancellation feature would completely stop working. As this is crucial for audio calls, this patch re-implement synchronization completely. Instead of letting it drift until next discont, we now synchronize against the record data at every iteration. This way we simply never let the stream drift for longer then 10ms period. We also shorter the delay by using the latency up the probe (basically excluding the sink latency. This is a decent delay to avoid starving in the probe queue. https://bugzilla.gnome.org/show_bug.cgi?id=768009
2016-06-29 19:56:47 +00:00
buffer = gst_adapter_take_buffer (self->adapter, self->period_size);
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
webrtcdsp: Rewrite echo data synchronization The previous code would run out of sync if there was packet lost or clock skews. When that happened, the echo cancellation feature would completely stop working. As this is crucial for audio calls, this patch re-implement synchronization completely. Instead of letting it drift until next discont, we now synchronize against the record data at every iteration. This way we simply never let the stream drift for longer then 10ms period. We also shorter the delay by using the latency up the probe (basically excluding the sink latency. This is a decent delay to avoid starving in the probe queue. https://bugzilla.gnome.org/show_bug.cgi?id=768009
2016-06-29 19:56:47 +00:00
GST_BUFFER_PTS (buffer) = timestamp;
GST_BUFFER_DURATION (buffer) = 10 * GST_MSECOND;
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
webrtcdsp: Rewrite echo data synchronization The previous code would run out of sync if there was packet lost or clock skews. When that happened, the echo cancellation feature would completely stop working. As this is crucial for audio calls, this patch re-implement synchronization completely. Instead of letting it drift until next discont, we now synchronize against the record data at every iteration. This way we simply never let the stream drift for longer then 10ms period. We also shorter the delay by using the latency up the probe (basically excluding the sink latency. This is a decent delay to avoid starving in the probe queue. https://bugzilla.gnome.org/show_bug.cgi?id=768009
2016-06-29 19:56:47 +00:00
if (gst_adapter_pts_at_discont (self->adapter) == timestamp && distance == 0) {
GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
} else
GST_BUFFER_FLAG_UNSET (buffer, GST_BUFFER_FLAG_DISCONT);
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
webrtcdsp: Rewrite echo data synchronization The previous code would run out of sync if there was packet lost or clock skews. When that happened, the echo cancellation feature would completely stop working. As this is crucial for audio calls, this patch re-implement synchronization completely. Instead of letting it drift until next discont, we now synchronize against the record data at every iteration. This way we simply never let the stream drift for longer then 10ms period. We also shorter the delay by using the latency up the probe (basically excluding the sink latency. This is a decent delay to avoid starving in the probe queue. https://bugzilla.gnome.org/show_bug.cgi?id=768009
2016-06-29 19:56:47 +00:00
return buffer;
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
}
webrtcdsp: Rewrite echo data synchronization The previous code would run out of sync if there was packet lost or clock skews. When that happened, the echo cancellation feature would completely stop working. As this is crucial for audio calls, this patch re-implement synchronization completely. Instead of letting it drift until next discont, we now synchronize against the record data at every iteration. This way we simply never let the stream drift for longer then 10ms period. We also shorter the delay by using the latency up the probe (basically excluding the sink latency. This is a decent delay to avoid starving in the probe queue. https://bugzilla.gnome.org/show_bug.cgi?id=768009
2016-06-29 19:56:47 +00:00
static GstFlowReturn
gst_webrtc_dsp_analyze_reverse_stream (GstWebrtcDsp * self,
GstClockTime rec_time)
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
{
GstWebrtcEchoProbe *probe = NULL;
webrtc::AudioProcessing * apm;
webrtc::AudioFrame frame;
webrtcdsp: Rewrite echo data synchronization The previous code would run out of sync if there was packet lost or clock skews. When that happened, the echo cancellation feature would completely stop working. As this is crucial for audio calls, this patch re-implement synchronization completely. Instead of letting it drift until next discont, we now synchronize against the record data at every iteration. This way we simply never let the stream drift for longer then 10ms period. We also shorter the delay by using the latency up the probe (basically excluding the sink latency. This is a decent delay to avoid starving in the probe queue. https://bugzilla.gnome.org/show_bug.cgi?id=768009
2016-06-29 19:56:47 +00:00
GstFlowReturn ret = GST_FLOW_OK;
gint err, delay;
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
GST_OBJECT_LOCK (self);
webrtcdsp: We now fail if there is no echo probe When echo cancel is enabled, we now fail the pipeline if there is not echo probe. For this reason there is no need to check if probe pointer is set anymore.
2016-06-29 19:55:19 +00:00
if (self->echo_cancel)
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
probe = GST_WEBRTC_ECHO_PROBE (g_object_ref (self->probe));
GST_OBJECT_UNLOCK (self);
webrtcdsp: Rewrite echo data synchronization The previous code would run out of sync if there was packet lost or clock skews. When that happened, the echo cancellation feature would completely stop working. As this is crucial for audio calls, this patch re-implement synchronization completely. Instead of letting it drift until next discont, we now synchronize against the record data at every iteration. This way we simply never let the stream drift for longer then 10ms period. We also shorter the delay by using the latency up the probe (basically excluding the sink latency. This is a decent delay to avoid starving in the probe queue. https://bugzilla.gnome.org/show_bug.cgi?id=768009
2016-06-29 19:56:47 +00:00
/* If echo cancellation is disabled */
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
if (!probe)
webrtcdsp: Rewrite echo data synchronization The previous code would run out of sync if there was packet lost or clock skews. When that happened, the echo cancellation feature would completely stop working. As this is crucial for audio calls, this patch re-implement synchronization completely. Instead of letting it drift until next discont, we now synchronize against the record data at every iteration. This way we simply never let the stream drift for longer then 10ms period. We also shorter the delay by using the latency up the probe (basically excluding the sink latency. This is a decent delay to avoid starving in the probe queue. https://bugzilla.gnome.org/show_bug.cgi?id=768009
2016-06-29 19:56:47 +00:00
return GST_FLOW_OK;
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
apm = self->apm;
webrtcdsp: Add delay-agnostic property In this mode, we let WebRTC Audio Processing figure-out the delay. This is useful when the latency reported by the stack cannot be trusted. Note that in this mode, the leaking of echo during packet lost is much worst. It is recommanded to use PLC (e.g. spanplc, or opus built-in plc). In this mode, we don't do any synchronization. Instead, we simply process all the available reverse stream data as it comes.
2016-07-14 03:00:51 +00:00
if (self->delay_agnostic)
rec_time = GST_CLOCK_TIME_NONE;
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
webrtcdsp: Add delay-agnostic property In this mode, we let WebRTC Audio Processing figure-out the delay. This is useful when the latency reported by the stack cannot be trusted. Note that in this mode, the leaking of echo during packet lost is much worst. It is recommanded to use PLC (e.g. spanplc, or opus built-in plc). In this mode, we don't do any synchronization. Instead, we simply process all the available reverse stream data as it comes.
2016-07-14 03:00:51 +00:00
again:
delay = gst_webrtc_echo_probe_read (probe, rec_time, (gpointer) &frame);
webrtcdsp: Rewrite echo data synchronization The previous code would run out of sync if there was packet lost or clock skews. When that happened, the echo cancellation feature would completely stop working. As this is crucial for audio calls, this patch re-implement synchronization completely. Instead of letting it drift until next discont, we now synchronize against the record data at every iteration. This way we simply never let the stream drift for longer then 10ms period. We also shorter the delay by using the latency up the probe (basically excluding the sink latency. This is a decent delay to avoid starving in the probe queue. https://bugzilla.gnome.org/show_bug.cgi?id=768009
2016-06-29 19:56:47 +00:00
apm->set_stream_delay_ms (delay);
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
webrtcdsp: Rewrite echo data synchronization The previous code would run out of sync if there was packet lost or clock skews. When that happened, the echo cancellation feature would completely stop working. As this is crucial for audio calls, this patch re-implement synchronization completely. Instead of letting it drift until next discont, we now synchronize against the record data at every iteration. This way we simply never let the stream drift for longer then 10ms period. We also shorter the delay by using the latency up the probe (basically excluding the sink latency. This is a decent delay to avoid starving in the probe queue. https://bugzilla.gnome.org/show_bug.cgi?id=768009
2016-06-29 19:56:47 +00:00
if (delay < 0)
goto done;
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
webrtcdsp: Rewrite echo data synchronization The previous code would run out of sync if there was packet lost or clock skews. When that happened, the echo cancellation feature would completely stop working. As this is crucial for audio calls, this patch re-implement synchronization completely. Instead of letting it drift until next discont, we now synchronize against the record data at every iteration. This way we simply never let the stream drift for longer then 10ms period. We also shorter the delay by using the latency up the probe (basically excluding the sink latency. This is a decent delay to avoid starving in the probe queue. https://bugzilla.gnome.org/show_bug.cgi?id=768009
2016-06-29 19:56:47 +00:00
if (frame.sample_rate_hz_ != self->info.rate) {
GST_ELEMENT_ERROR (self, STREAM, FORMAT,
("Echo Probe has rate %i , while the DSP is running at rate %i,"
" use a caps filter to ensure those are the same.",
frame.sample_rate_hz_, self->info.rate), (NULL));
ret = GST_FLOW_ERROR;
goto done;
}
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
if ((err = apm->AnalyzeReverseStream (&frame)) < 0)
GST_WARNING_OBJECT (self, "Reverse stream analyses failed: %s.",
webrtc_error_to_string (err));
webrtcdsp: Add delay-agnostic property In this mode, we let WebRTC Audio Processing figure-out the delay. This is useful when the latency reported by the stack cannot be trusted. Note that in this mode, the leaking of echo during packet lost is much worst. It is recommanded to use PLC (e.g. spanplc, or opus built-in plc). In this mode, we don't do any synchronization. Instead, we simply process all the available reverse stream data as it comes.
2016-07-14 03:00:51 +00:00
if (self->delay_agnostic)
goto again;
webrtcdsp: Rewrite echo data synchronization The previous code would run out of sync if there was packet lost or clock skews. When that happened, the echo cancellation feature would completely stop working. As this is crucial for audio calls, this patch re-implement synchronization completely. Instead of letting it drift until next discont, we now synchronize against the record data at every iteration. This way we simply never let the stream drift for longer then 10ms period. We also shorter the delay by using the latency up the probe (basically excluding the sink latency. This is a decent delay to avoid starving in the probe queue. https://bugzilla.gnome.org/show_bug.cgi?id=768009
2016-06-29 19:56:47 +00:00
done:
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
gst_object_unref (probe);
webrtcdsp: Rewrite echo data synchronization The previous code would run out of sync if there was packet lost or clock skews. When that happened, the echo cancellation feature would completely stop working. As this is crucial for audio calls, this patch re-implement synchronization completely. Instead of letting it drift until next discont, we now synchronize against the record data at every iteration. This way we simply never let the stream drift for longer then 10ms period. We also shorter the delay by using the latency up the probe (basically excluding the sink latency. This is a decent delay to avoid starving in the probe queue. https://bugzilla.gnome.org/show_bug.cgi?id=768009
2016-06-29 19:56:47 +00:00
return ret;
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
}
webrtcdsp: Add delay-agnostic property In this mode, we let WebRTC Audio Processing figure-out the delay. This is useful when the latency reported by the stack cannot be trusted. Note that in this mode, the leaking of echo during packet lost is much worst. It is recommanded to use PLC (e.g. spanplc, or opus built-in plc). In this mode, we don't do any synchronization. Instead, we simply process all the available reverse stream data as it comes.
2016-07-14 03:00:51 +00:00
static GstFlowReturn
webrtcdsp: Rewrite echo data synchronization The previous code would run out of sync if there was packet lost or clock skews. When that happened, the echo cancellation feature would completely stop working. As this is crucial for audio calls, this patch re-implement synchronization completely. Instead of letting it drift until next discont, we now synchronize against the record data at every iteration. This way we simply never let the stream drift for longer then 10ms period. We also shorter the delay by using the latency up the probe (basically excluding the sink latency. This is a decent delay to avoid starving in the probe queue. https://bugzilla.gnome.org/show_bug.cgi?id=768009
2016-06-29 19:56:47 +00:00
gst_webrtc_dsp_process_stream (GstWebrtcDsp * self,
GstBuffer * buffer)
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
{
GstMapInfo info;
webrtc::AudioProcessing * apm = self->apm;
webrtc::AudioFrame frame;
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;
if (!gst_buffer_map (buffer, &info, (GstMapFlags) GST_MAP_READWRITE)) {
gst_buffer_unref (buffer);
webrtcdsp: Rewrite echo data synchronization The previous code would run out of sync if there was packet lost or clock skews. When that happened, the echo cancellation feature would completely stop working. As this is crucial for audio calls, this patch re-implement synchronization completely. Instead of letting it drift until next discont, we now synchronize against the record data at every iteration. This way we simply never let the stream drift for longer then 10ms period. We also shorter the delay by using the latency up the probe (basically excluding the sink latency. This is a decent delay to avoid starving in the probe queue. https://bugzilla.gnome.org/show_bug.cgi?id=768009
2016-06-29 19:56:47 +00:00
return GST_FLOW_ERROR;
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
}
memcpy (frame.data_, info.data, self->period_size);
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);
webrtcdsp: Rewrite echo data synchronization The previous code would run out of sync if there was packet lost or clock skews. When that happened, the echo cancellation feature would completely stop working. As this is crucial for audio calls, this patch re-implement synchronization completely. Instead of letting it drift until next discont, we now synchronize against the record data at every iteration. This way we simply never let the stream drift for longer then 10ms period. We also shorter the delay by using the latency up the probe (basically excluding the sink latency. This is a decent delay to avoid starving in the probe queue. https://bugzilla.gnome.org/show_bug.cgi?id=768009
2016-06-29 19:56:47 +00:00
return GST_FLOW_OK;
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
}
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 (is_discont) {
webrtcdsp: Rewrite echo data synchronization The previous code would run out of sync if there was packet lost or clock skews. When that happened, the echo cancellation feature would completely stop working. As this is crucial for audio calls, this patch re-implement synchronization completely. Instead of letting it drift until next discont, we now synchronize against the record data at every iteration. This way we simply never let the stream drift for longer then 10ms period. We also shorter the delay by using the latency up the probe (basically excluding the sink latency. This is a decent delay to avoid starving in the probe queue. https://bugzilla.gnome.org/show_bug.cgi?id=768009
2016-06-29 19:56:47 +00:00
GST_DEBUG_OBJECT (self,
"Received discont, clearing adapter.");
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
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);
webrtcdsp: Rewrite echo data synchronization The previous code would run out of sync if there was packet lost or clock skews. When that happened, the echo cancellation feature would completely stop working. As this is crucial for audio calls, this patch re-implement synchronization completely. Instead of letting it drift until next discont, we now synchronize against the record data at every iteration. This way we simply never let the stream drift for longer then 10ms period. We also shorter the delay by using the latency up the probe (basically excluding the sink latency. This is a decent delay to avoid starving in the probe queue. https://bugzilla.gnome.org/show_bug.cgi?id=768009
2016-06-29 19:56:47 +00:00
GstFlowReturn ret;
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
webrtcdsp: Rewrite echo data synchronization The previous code would run out of sync if there was packet lost or clock skews. When that happened, the echo cancellation feature would completely stop working. As this is crucial for audio calls, this patch re-implement synchronization completely. Instead of letting it drift until next discont, we now synchronize against the record data at every iteration. This way we simply never let the stream drift for longer then 10ms period. We also shorter the delay by using the latency up the probe (basically excluding the sink latency. This is a decent delay to avoid starving in the probe queue. https://bugzilla.gnome.org/show_bug.cgi?id=768009
2016-06-29 19:56:47 +00:00
if (gst_adapter_available (self->adapter) < self->period_size) {
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
*outbuf = NULL;
webrtcdsp: Rewrite echo data synchronization The previous code would run out of sync if there was packet lost or clock skews. When that happened, the echo cancellation feature would completely stop working. As this is crucial for audio calls, this patch re-implement synchronization completely. Instead of letting it drift until next discont, we now synchronize against the record data at every iteration. This way we simply never let the stream drift for longer then 10ms period. We also shorter the delay by using the latency up the probe (basically excluding the sink latency. This is a decent delay to avoid starving in the probe queue. https://bugzilla.gnome.org/show_bug.cgi?id=768009
2016-06-29 19:56:47 +00:00
return GST_FLOW_OK;
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
}
webrtcdsp: Rewrite echo data synchronization The previous code would run out of sync if there was packet lost or clock skews. When that happened, the echo cancellation feature would completely stop working. As this is crucial for audio calls, this patch re-implement synchronization completely. Instead of letting it drift until next discont, we now synchronize against the record data at every iteration. This way we simply never let the stream drift for longer then 10ms period. We also shorter the delay by using the latency up the probe (basically excluding the sink latency. This is a decent delay to avoid starving in the probe queue. https://bugzilla.gnome.org/show_bug.cgi?id=768009
2016-06-29 19:56:47 +00:00
*outbuf = gst_webrtc_dsp_take_buffer (self);
ret = gst_webrtc_dsp_analyze_reverse_stream (self, GST_BUFFER_PTS (*outbuf));
if (ret == GST_FLOW_OK)
ret = gst_webrtc_dsp_process_stream (self, *outbuf);
return ret;
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
}
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));
webrtcdsp: Add delay-agnostic property In this mode, we let WebRTC Audio Processing figure-out the delay. This is useful when the latency reported by the stack cannot be trusted. Note that in this mode, the leaking of echo during packet lost is much worst. It is recommanded to use PLC (e.g. spanplc, or opus built-in plc). In this mode, we don't do any synchronization. Instead, we simply process all the available reverse stream data as it comes.
2016-07-14 03:00:51 +00:00
config.Set < webrtc::DelayAgnostic >
(new webrtc::DelayAgnostic (self->delay_agnostic));
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
/* 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;
}
GST_WEBRTC_ECHO_PROBE_UNLOCK (self->probe);
}
webrtcdsp: Workaround GCC/MingW 4.7 limitations It refuses to initialize a classes using brackets notation. This is to allow building using our mingw version.
2016-07-05 13:21:58 +00:00
/* 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);
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
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);
}
webrtcdsp: Rewrite echo data synchronization The previous code would run out of sync if there was packet lost or clock skews. When that happened, the echo cancellation feature would completely stop working. As this is crucial for audio calls, this patch re-implement synchronization completely. Instead of letting it drift until next discont, we now synchronize against the record data at every iteration. This way we simply never let the stream drift for longer then 10ms period. We also shorter the delay by using the latency up the probe (basically excluding the sink latency. This is a decent delay to avoid starving in the probe queue. https://bugzilla.gnome.org/show_bug.cgi?id=768009
2016-06-29 19:56:47 +00:00
if (self->echo_cancel) {
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
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;
webrtcdsp: Add delay-agnostic property In this mode, we let WebRTC Audio Processing figure-out the delay. This is useful when the latency reported by the stack cannot be trusted. Note that in this mode, the leaking of echo during packet lost is much worst. It is recommanded to use PLC (e.g. spanplc, or opus built-in plc). In this mode, we don't do any synchronization. Instead, we simply process all the available reverse stream data as it comes.
2016-07-14 03:00:51 +00:00
case PROP_DELAY_AGNOSTIC:
self->delay_agnostic = g_value_get_boolean (value);
break;
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
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;
webrtcdsp: Add delay-agnostic property In this mode, we let WebRTC Audio Processing figure-out the delay. This is useful when the latency reported by the stack cannot be trusted. Note that in this mode, the leaking of echo during packet lost is much worst. It is recommanded to use PLC (e.g. spanplc, or opus built-in plc). In this mode, we don't do any synchronization. Instead, we simply process all the available reverse stream data as it comes.
2016-07-14 03:00:51 +00:00
case PROP_DELAY_AGNOSTIC:
g_value_set_boolean (value, self->delay_agnostic);
break;
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
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)));
webrtcdsp: Add delay-agnostic property In this mode, we let WebRTC Audio Processing figure-out the delay. This is useful when the latency reported by the stack cannot be trusted. Note that in this mode, the leaking of echo during packet lost is much worst. It is recommanded to use PLC (e.g. spanplc, or opus built-in plc). In this mode, we don't do any synchronization. Instead, we simply process all the available reverse stream data as it comes.
2016-07-14 03:00:51 +00:00
g_object_class_install_property (gobject_class,
PROP_DELAY_AGNOSTIC,
g_param_spec_boolean ("delay-agnostic", "Delay Agnostic",
"Enable or disable the delay agnostic mode.",
FALSE, (GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
G_PARAM_CONSTRUCT)));
webrtcdsp: Add WebRTC Audio Processing support 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
2016-06-06 19:10:05 +00:00
}
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")
Reference in a new issue Copy permalink