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Tuning some webrtc parameters

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KangJing Huang (Chaserhkj) 2021-10-19 14:16:23 -04:00 • committed by Kangjing "Chaser" Huang
parent 2bfc03f43c
commit 22aad6eaff
1 changed files with 14 additions and 6 deletions
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20
src/modules/module-echo-cancel/aec-webrtc.cpp
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@ -46,6 +46,11 @@ static void *webrtc_create(const struct pw_properties *args, const spa_audio_inf
webrtc::ProcessingConfig pconfig;
webrtc::Config config;
// TODO: wire up args to control these
config.Set<webrtc::ExtendedFilter>(new webrtc::ExtendedFilter(true));
// Enable Agnostic Delay Detection, would greatly improve performance for larger latencies
config.Set<webrtc::DelayAgnostic>(new webrtc::DelayAgnostic(true));
apm = webrtc::AudioProcessing::Create(config);
pconfig = {{
@ -64,13 +69,15 @@ static void *webrtc_create(const struct pw_properties *args, const spa_audio_inf
apm->high_pass_filter()->Enable(true);
apm->echo_cancellation()->enable_drift_compensation(false);
apm->echo_cancellation()->Enable(true);
apm->echo_cancellation()->set_suppression_level(webrtc::EchoCancellation::kHighSuppression);
apm->noise_suppression()->set_level(webrtc::NoiseSuppression::kHigh);
apm->noise_suppression()->Enable(true);
apm->gain_control()->set_analog_level_limits(0, 255);
// FIXME: can we hook up AGC?
apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveDigital);
apm->gain_control()->Enable(true);
apm->voice_detection()->Enable(true);
// FIXME: wire up args to AGC
// Note: AGC seems to mess up with Agnostic Delay Detection, especially with speech,
// result in very poor performance, disable by default
// apm->gain_control()->set_analog_level_limits(0, 255);
// apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveDigital);
// apm->gain_control()->Enable(true);
impl = (struct impl *)calloc(1, sizeof(struct impl));
impl->info = *info;
@ -127,7 +134,8 @@ static int webrtc_run(void *ec, const float *rec[], const float *play[], float *
pw_log_error("Processing reverse stream failed");
}
impl->apm->set_stream_delay_ms(0);
// Extra delay introduced by multiple frames
impl->apm->set_stream_delay_ms((num_blocks - 1) * 10);
if (impl->apm->ProcessStream(impl->rec_buffer, config, config, impl->out_buffer) !=
webrtc::AudioProcessing::kNoError) {