mirrors/pipewire
1
0
Fork 0
mirror of https://gitlab.freedesktop.org/pipewire/pipewire.git synced 2025-10-29 05:40:27 -04:00
Code Issues Projects Releases Packages Wiki Activity Actions

spa: aec: Add support for webrtc-audio-processing-2

Browse source
This commit is contained in:
Arun Raghavan 2024-12-27 03:22:59 -05:00 • committed by Wim Taymans
parent 3752535171
commit 71e403bbdb
3 changed files with 65 additions and 20 deletions
Download patch file Download diff file Expand all files Collapse all files

28
meson.build
View file

@ -431,20 +431,28 @@ cdata.set('HAVE_GSTREAMER_DMA_DRM', gst_dma_drm_found)
if get_option('echo-cancel-webrtc').disabled() if get_option('echo-cancel-webrtc').disabled()
webrtc_dep = dependency('', required: false) webrtc_dep = dependency('', required: false)
summary({'WebRTC Echo Canceling >= 1.2': webrtc_dep.found()}, bool_yn: true, section: 'Misc dependencies') summary({'WebRTC Echo Canceling': webrtc_dep.found()}, bool_yn: false, section: 'Misc dependencies')
else else
webrtc_dep = dependency('webrtc-audio-processing-1', webrtc_dep = dependency('webrtc-audio-processing-2',
version : ['>= 1.2' ], version : ['>= 2.0' ],
required : false) required : false)
cdata.set('HAVE_WEBRTC1', webrtc_dep.found()) cdata.set('HAVE_WEBRTC2', webrtc_dep.found())
if webrtc_dep.found() if webrtc_dep.found()
summary({'WebRTC Echo Canceling >= 1.2': webrtc_dep.found()}, bool_yn: true, section: 'Misc dependencies') summary({'WebRTC Echo Canceling >= 2.0': webrtc_dep.found()}, bool_yn: true, section: 'Misc dependencies')
else else
webrtc_dep = dependency('webrtc-audio-processing', webrtc_dep = dependency('webrtc-audio-processing-1',
version : ['>= 0.2', '< 1.0'], version : ['>= 1.2' ],
required : get_option('echo-cancel-webrtc')) required : false)
cdata.set('HAVE_WEBRTC', webrtc_dep.found()) cdata.set('HAVE_WEBRTC1', webrtc_dep.found())
summary({'WebRTC Echo Canceling < 1.0': webrtc_dep.found()}, bool_yn: true, section: 'Misc dependencies') if webrtc_dep.found()
summary({'WebRTC Echo Canceling >= 1.2': webrtc_dep.found()}, bool_yn: true, section: 'Misc dependencies')
else
webrtc_dep = dependency('webrtc-audio-processing',
version : ['>= 0.2', '< 1.0'],
required : get_option('echo-cancel-webrtc'))
cdata.set('HAVE_WEBRTC', webrtc_dep.found())
summary({'WebRTC Echo Canceling < 1.0': webrtc_dep.found()}, bool_yn: true, section: 'Misc dependencies')
endif
endif endif
endif endif

53
spa/plugins/aec/aec-webrtc.cpp
View file

@ -28,7 +28,11 @@ struct impl_data {
struct spa_audio_aec aec; struct spa_audio_aec aec;
struct spa_log *log; struct spa_log *log;
#if defined(HAVE_WEBRTC) || defined(HAVE_WEBRTC1)
std::unique_ptr<webrtc::AudioProcessing> apm; std::unique_ptr<webrtc::AudioProcessing> apm;
#elif defined(HAVE_WEBRTC2)
rtc::scoped_refptr<webrtc::AudioProcessing> apm;
#endif
spa_audio_info_raw rec_info; spa_audio_info_raw rec_info;
spa_audio_info_raw out_info; spa_audio_info_raw out_info;
spa_audio_info_raw play_info; spa_audio_info_raw play_info;
@ -103,16 +107,17 @@ static int webrtc_init2(void *object, const struct spa_dict *args,
bool high_pass_filter = webrtc_get_spa_bool(args, "webrtc.high_pass_filter", true); bool high_pass_filter = webrtc_get_spa_bool(args, "webrtc.high_pass_filter", true);
bool noise_suppression = webrtc_get_spa_bool(args, "webrtc.noise_suppression", true); bool noise_suppression = webrtc_get_spa_bool(args, "webrtc.noise_suppression", true);
bool voice_detection = webrtc_get_spa_bool(args, "webrtc.voice_detection", true); #if defined(HAVE_WEBRTC)
#ifdef HAVE_WEBRTC
bool extended_filter = webrtc_get_spa_bool(args, "webrtc.extended_filter", true); bool extended_filter = webrtc_get_spa_bool(args, "webrtc.extended_filter", true);
bool delay_agnostic = webrtc_get_spa_bool(args, "webrtc.delay_agnostic", true); bool delay_agnostic = webrtc_get_spa_bool(args, "webrtc.delay_agnostic", true);
// Disable experimental flags by default // Disable experimental flags by default
bool experimental_agc = webrtc_get_spa_bool(args, "webrtc.experimental_agc", false); bool experimental_agc = webrtc_get_spa_bool(args, "webrtc.experimental_agc", false);
bool experimental_ns = webrtc_get_spa_bool(args, "webrtc.experimental_ns", false); bool experimental_ns = webrtc_get_spa_bool(args, "webrtc.experimental_ns", false);
bool voice_detection = webrtc_get_spa_bool(args, "webrtc.voice_detection", true);
bool beamforming = webrtc_get_spa_bool(args, "webrtc.beamforming", false); bool beamforming = webrtc_get_spa_bool(args, "webrtc.beamforming", false);
#else #elif defined(HAVE_WEBRTC1)
bool voice_detection = webrtc_get_spa_bool(args, "webrtc.voice_detection", true);
bool transient_suppression = webrtc_get_spa_bool(args, "webrtc.transient_suppression", true); bool transient_suppression = webrtc_get_spa_bool(args, "webrtc.transient_suppression", true);
#endif #endif
// Note: AGC seems to mess up with Agnostic Delay Detection, especially with speech, // Note: AGC seems to mess up with Agnostic Delay Detection, especially with speech,
@ -123,7 +128,7 @@ static int webrtc_init2(void *object, const struct spa_dict *args,
// This filter will modify playback buffer (when calling ProcessReverseStream), but now // This filter will modify playback buffer (when calling ProcessReverseStream), but now
// playback buffer modifications are discarded. // playback buffer modifications are discarded.
#ifdef HAVE_WEBRTC #if defined(HAVE_WEBRTC)
webrtc::Config config; webrtc::Config config;
config.Set<webrtc::ExtendedFilter>(new webrtc::ExtendedFilter(extended_filter)); config.Set<webrtc::ExtendedFilter>(new webrtc::ExtendedFilter(extended_filter));
config.Set<webrtc::DelayAgnostic>(new webrtc::DelayAgnostic(delay_agnostic)); config.Set<webrtc::DelayAgnostic>(new webrtc::DelayAgnostic(delay_agnostic));
@ -167,7 +172,7 @@ static int webrtc_init2(void *object, const struct spa_dict *args,
config.Set<webrtc::Beamforming>(new webrtc::Beamforming(true, geometry)); config.Set<webrtc::Beamforming>(new webrtc::Beamforming(true, geometry));
} }
} }
#else #elif defined(HAVE_WEBRTC1)
webrtc::AudioProcessing::Config config; webrtc::AudioProcessing::Config config;
config.echo_canceller.enabled = true; config.echo_canceller.enabled = true;
config.pipeline.multi_channel_capture = rec_info->channels > 1; config.pipeline.multi_channel_capture = rec_info->channels > 1;
@ -184,20 +189,43 @@ static int webrtc_init2(void *object, const struct spa_dict *args,
// FIXME: expose pre/postamp gain // FIXME: expose pre/postamp gain
config.transient_suppression.enabled = transient_suppression; config.transient_suppression.enabled = transient_suppression;
config.voice_detection.enabled = voice_detection; config.voice_detection.enabled = voice_detection;
#elif defined(HAVE_WEBRTC2)
webrtc::AudioProcessing::Config config;
config.pipeline.multi_channel_capture = rec_info->channels > 1;
config.pipeline.multi_channel_render = play_info->channels > 1;
// FIXME: Example code enables both gain controllers, but that seems sus
config.gain_controller1.enabled = gain_control;
config.gain_controller1.mode = webrtc::AudioProcessing::Config::GainController1::Mode::kAdaptiveDigital;
config.gain_controller2.enabled = gain_control;
config.high_pass_filter.enabled = high_pass_filter;
config.noise_suppression.enabled = noise_suppression;
config.noise_suppression.level = webrtc::AudioProcessing::Config::NoiseSuppression::kHigh;
// FIXME: expose pre/postamp gain
#endif #endif
#if defined(HAVE_WEBRTC) || defined(HAVE_WEBRTC1)
webrtc::ProcessingConfig pconfig = {{ webrtc::ProcessingConfig pconfig = {{
webrtc::StreamConfig(rec_info->rate, rec_info->channels, false), /* input stream */ webrtc::StreamConfig(rec_info->rate, rec_info->channels, false), /* input stream */
webrtc::StreamConfig(out_info->rate, out_info->channels, false), /* output stream */ webrtc::StreamConfig(out_info->rate, out_info->channels, false), /* output stream */
webrtc::StreamConfig(play_info->rate, play_info->channels, false), /* reverse input stream */ webrtc::StreamConfig(play_info->rate, play_info->channels, false), /* reverse input stream */
webrtc::StreamConfig(play_info->rate, play_info->channels, false), /* reverse output stream */ webrtc::StreamConfig(play_info->rate, play_info->channels, false), /* reverse output stream */
}}; }};
#elif defined(HAVE_WEBRTC2)
webrtc::ProcessingConfig pconfig = {{
webrtc::StreamConfig(rec_info->rate, rec_info->channels), /* input stream */
webrtc::StreamConfig(out_info->rate, out_info->channels), /* output stream */
webrtc::StreamConfig(play_info->rate, play_info->channels), /* reverse input stream */
webrtc::StreamConfig(play_info->rate, play_info->channels), /* reverse output stream */
}};
#endif
#ifdef HAVE_WEBRTC #if defined(HAVE_WEBRTC)
auto apm = std::unique_ptr<webrtc::AudioProcessing>(webrtc::AudioProcessing::Create(config)); auto apm = std::unique_ptr<webrtc::AudioProcessing>(webrtc::AudioProcessing::Create(config));
#else #elif defined(HAVE_WEBRTC1)
auto apm = std::unique_ptr<webrtc::AudioProcessing>(webrtc::AudioProcessingBuilder().Create()); auto apm = std::unique_ptr<webrtc::AudioProcessing>(webrtc::AudioProcessingBuilder().Create());
apm->ApplyConfig(config);
#elif defined(HAVE_WEBRTC2)
auto apm = webrtc::AudioProcessingBuilder().Create();
apm->ApplyConfig(config); apm->ApplyConfig(config);
#endif #endif
@ -250,12 +278,21 @@ static int webrtc_run(void *object, const float *rec[], const float *play[], flo
auto impl = static_cast<struct impl_data*>(object); auto impl = static_cast<struct impl_data*>(object);
int res; int res;
#if defined(HAVE_WEBRTC) || defined(HAVE_WEBRTC1)
webrtc::StreamConfig play_config = webrtc::StreamConfig play_config =
webrtc::StreamConfig(impl->play_info.rate, impl->play_info.channels, false); webrtc::StreamConfig(impl->play_info.rate, impl->play_info.channels, false);
webrtc::StreamConfig rec_config = webrtc::StreamConfig rec_config =
webrtc::StreamConfig(impl->rec_info.rate, impl->rec_info.channels, false); webrtc::StreamConfig(impl->rec_info.rate, impl->rec_info.channels, false);
webrtc::StreamConfig out_config = webrtc::StreamConfig out_config =
webrtc::StreamConfig(impl->out_info.rate, impl->out_info.channels, false); webrtc::StreamConfig(impl->out_info.rate, impl->out_info.channels, false);
#elif defined(HAVE_WEBRTC2)
webrtc::StreamConfig play_config =
webrtc::StreamConfig(impl->play_info.rate, impl->play_info.channels);
webrtc::StreamConfig rec_config =
webrtc::StreamConfig(impl->rec_info.rate, impl->rec_info.channels);
webrtc::StreamConfig out_config =
webrtc::StreamConfig(impl->out_info.rate, impl->out_info.channels);
#endif
unsigned int num_blocks = n_samples * 1000 / impl->play_info.rate / 10; unsigned int num_blocks = n_samples * 1000 / impl->play_info.rate / 10;
if (n_samples * 1000 / impl->play_info.rate % 10 != 0) { if (n_samples * 1000 / impl->play_info.rate % 10 != 0) {

4
subprojects/webrtc-audio-processing.wrap
View file

@ -2,7 +2,7 @@
directory = webrtc-audio-processing directory = webrtc-audio-processing
url = https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing.git url = https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing.git
push-url = git@gitlab.freedesktop.org:pulseaudio/webrtc-audio-processing.git push-url = git@gitlab.freedesktop.org:pulseaudio/webrtc-audio-processing.git
revision = v1.3 revision = v2.0
[provide] [provide]
dependency_names = webrtc-audio-coding-1, webrtc-audio-processing-1 dependency_names = webrtc-audio-processing-2