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spa: aec: Add support for webrtc-audio-processing-2
This commit is contained in:
Arun Raghavan
Wim Taymans
parent
3752535171
commit
71e403bbdb
3 changed files with 65 additions and 20 deletions
28
meson.build
28
meson.build
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@ -431,20 +431,28 @@ cdata.set('HAVE_GSTREAMER_DMA_DRM', gst_dma_drm_found)
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if get_option('echo-cancel-webrtc').disabled()
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webrtc_dep = dependency('', required: false)
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summary({'WebRTC Echo Canceling >= 1.2': webrtc_dep.found()}, bool_yn: true, section: 'Misc dependencies')
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summary({'WebRTC Echo Canceling': webrtc_dep.found()}, bool_yn: false, section: 'Misc dependencies')
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else
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webrtc_dep = dependency('webrtc-audio-processing-1',
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version : ['>= 1.2' ],
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webrtc_dep = dependency('webrtc-audio-processing-2',
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version : ['>= 2.0' ],
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required : false)
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cdata.set('HAVE_WEBRTC1', webrtc_dep.found())
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cdata.set('HAVE_WEBRTC2', webrtc_dep.found())
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if webrtc_dep.found()
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summary({'WebRTC Echo Canceling >= 1.2': webrtc_dep.found()}, bool_yn: true, section: 'Misc dependencies')
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summary({'WebRTC Echo Canceling >= 2.0': webrtc_dep.found()}, bool_yn: true, section: 'Misc dependencies')
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else
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webrtc_dep = dependency('webrtc-audio-processing',
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version : ['>= 0.2', '< 1.0'],
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required : get_option('echo-cancel-webrtc'))
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cdata.set('HAVE_WEBRTC', webrtc_dep.found())
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summary({'WebRTC Echo Canceling < 1.0': webrtc_dep.found()}, bool_yn: true, section: 'Misc dependencies')
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webrtc_dep = dependency('webrtc-audio-processing-1',
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version : ['>= 1.2' ],
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required : false)
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cdata.set('HAVE_WEBRTC1', webrtc_dep.found())
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if webrtc_dep.found()
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summary({'WebRTC Echo Canceling >= 1.2': webrtc_dep.found()}, bool_yn: true, section: 'Misc dependencies')
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else
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webrtc_dep = dependency('webrtc-audio-processing',
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version : ['>= 0.2', '< 1.0'],
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required : get_option('echo-cancel-webrtc'))
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cdata.set('HAVE_WEBRTC', webrtc_dep.found())
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summary({'WebRTC Echo Canceling < 1.0': webrtc_dep.found()}, bool_yn: true, section: 'Misc dependencies')
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endif
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endif
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endif
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@ -28,7 +28,11 @@ struct impl_data {
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struct spa_audio_aec aec;
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struct spa_log *log;
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#if defined(HAVE_WEBRTC) || defined(HAVE_WEBRTC1)
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std::unique_ptr<webrtc::AudioProcessing> apm;
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#elif defined(HAVE_WEBRTC2)
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rtc::scoped_refptr<webrtc::AudioProcessing> apm;
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#endif
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spa_audio_info_raw rec_info;
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spa_audio_info_raw out_info;
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spa_audio_info_raw play_info;
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@ -103,16 +107,17 @@ static int webrtc_init2(void *object, const struct spa_dict *args,
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bool high_pass_filter = webrtc_get_spa_bool(args, "webrtc.high_pass_filter", true);
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bool noise_suppression = webrtc_get_spa_bool(args, "webrtc.noise_suppression", true);
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bool voice_detection = webrtc_get_spa_bool(args, "webrtc.voice_detection", true);
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#ifdef HAVE_WEBRTC
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#if defined(HAVE_WEBRTC)
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bool extended_filter = webrtc_get_spa_bool(args, "webrtc.extended_filter", true);
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bool delay_agnostic = webrtc_get_spa_bool(args, "webrtc.delay_agnostic", true);
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// Disable experimental flags by default
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bool experimental_agc = webrtc_get_spa_bool(args, "webrtc.experimental_agc", false);
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bool experimental_ns = webrtc_get_spa_bool(args, "webrtc.experimental_ns", false);
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bool voice_detection = webrtc_get_spa_bool(args, "webrtc.voice_detection", true);
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bool beamforming = webrtc_get_spa_bool(args, "webrtc.beamforming", false);
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#else
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#elif defined(HAVE_WEBRTC1)
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bool voice_detection = webrtc_get_spa_bool(args, "webrtc.voice_detection", true);
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bool transient_suppression = webrtc_get_spa_bool(args, "webrtc.transient_suppression", true);
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#endif
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// Note: AGC seems to mess up with Agnostic Delay Detection, especially with speech,
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@ -123,7 +128,7 @@ static int webrtc_init2(void *object, const struct spa_dict *args,
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// This filter will modify playback buffer (when calling ProcessReverseStream), but now
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// playback buffer modifications are discarded.
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#ifdef HAVE_WEBRTC
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#if defined(HAVE_WEBRTC)
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webrtc::Config config;
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config.Set<webrtc::ExtendedFilter>(new webrtc::ExtendedFilter(extended_filter));
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config.Set<webrtc::DelayAgnostic>(new webrtc::DelayAgnostic(delay_agnostic));
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@ -167,7 +172,7 @@ static int webrtc_init2(void *object, const struct spa_dict *args,
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config.Set<webrtc::Beamforming>(new webrtc::Beamforming(true, geometry));
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}
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}
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#else
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#elif defined(HAVE_WEBRTC1)
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webrtc::AudioProcessing::Config config;
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config.echo_canceller.enabled = true;
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config.pipeline.multi_channel_capture = rec_info->channels > 1;
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@ -184,20 +189,43 @@ static int webrtc_init2(void *object, const struct spa_dict *args,
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// FIXME: expose pre/postamp gain
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config.transient_suppression.enabled = transient_suppression;
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config.voice_detection.enabled = voice_detection;
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#elif defined(HAVE_WEBRTC2)
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webrtc::AudioProcessing::Config config;
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config.pipeline.multi_channel_capture = rec_info->channels > 1;
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config.pipeline.multi_channel_render = play_info->channels > 1;
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// FIXME: Example code enables both gain controllers, but that seems sus
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config.gain_controller1.enabled = gain_control;
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config.gain_controller1.mode = webrtc::AudioProcessing::Config::GainController1::Mode::kAdaptiveDigital;
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config.gain_controller2.enabled = gain_control;
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config.high_pass_filter.enabled = high_pass_filter;
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config.noise_suppression.enabled = noise_suppression;
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config.noise_suppression.level = webrtc::AudioProcessing::Config::NoiseSuppression::kHigh;
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// FIXME: expose pre/postamp gain
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#endif
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#if defined(HAVE_WEBRTC) || defined(HAVE_WEBRTC1)
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webrtc::ProcessingConfig pconfig = {{
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webrtc::StreamConfig(rec_info->rate, rec_info->channels, false), /* input stream */
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webrtc::StreamConfig(out_info->rate, out_info->channels, false), /* output stream */
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webrtc::StreamConfig(play_info->rate, play_info->channels, false), /* reverse input stream */
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webrtc::StreamConfig(play_info->rate, play_info->channels, false), /* reverse output stream */
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}};
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#elif defined(HAVE_WEBRTC2)
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webrtc::ProcessingConfig pconfig = {{
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webrtc::StreamConfig(rec_info->rate, rec_info->channels), /* input stream */
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webrtc::StreamConfig(out_info->rate, out_info->channels), /* output stream */
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webrtc::StreamConfig(play_info->rate, play_info->channels), /* reverse input stream */
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webrtc::StreamConfig(play_info->rate, play_info->channels), /* reverse output stream */
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}};
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#endif
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#ifdef HAVE_WEBRTC
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#if defined(HAVE_WEBRTC)
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auto apm = std::unique_ptr<webrtc::AudioProcessing>(webrtc::AudioProcessing::Create(config));
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#else
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#elif defined(HAVE_WEBRTC1)
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auto apm = std::unique_ptr<webrtc::AudioProcessing>(webrtc::AudioProcessingBuilder().Create());
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apm->ApplyConfig(config);
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#elif defined(HAVE_WEBRTC2)
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auto apm = webrtc::AudioProcessingBuilder().Create();
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apm->ApplyConfig(config);
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#endif
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@ -250,12 +278,21 @@ static int webrtc_run(void *object, const float *rec[], const float *play[], flo
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auto impl = static_cast<struct impl_data*>(object);
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int res;
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#if defined(HAVE_WEBRTC) || defined(HAVE_WEBRTC1)
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webrtc::StreamConfig play_config =
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webrtc::StreamConfig(impl->play_info.rate, impl->play_info.channels, false);
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webrtc::StreamConfig rec_config =
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webrtc::StreamConfig(impl->rec_info.rate, impl->rec_info.channels, false);
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webrtc::StreamConfig out_config =
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webrtc::StreamConfig(impl->out_info.rate, impl->out_info.channels, false);
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#elif defined(HAVE_WEBRTC2)
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webrtc::StreamConfig play_config =
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webrtc::StreamConfig(impl->play_info.rate, impl->play_info.channels);
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webrtc::StreamConfig rec_config =
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webrtc::StreamConfig(impl->rec_info.rate, impl->rec_info.channels);
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webrtc::StreamConfig out_config =
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webrtc::StreamConfig(impl->out_info.rate, impl->out_info.channels);
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#endif
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unsigned int num_blocks = n_samples * 1000 / impl->play_info.rate / 10;
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if (n_samples * 1000 / impl->play_info.rate % 10 != 0) {
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@ -2,7 +2,7 @@
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directory = webrtc-audio-processing
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url = https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing.git
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push-url = git@gitlab.freedesktop.org:pulseaudio/webrtc-audio-processing.git
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revision = v1.3
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revision = v2.0
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[provide]
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dependency_names = webrtc-audio-coding-1, webrtc-audio-processing-1
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dependency_names = webrtc-audio-processing-2
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