// // libtgvoip is free and unencumbered public domain software. // For more information, see http://unlicense.org or the UNLICENSE file // you should have received with this source code distribution. // #ifndef TGVOIP_NO_DSP #include "webrtc_dsp/modules/audio_processing/include/audio_processing.h" #include "webrtc_dsp/api/audio/audio_frame.h" #endif #include "controller/audio/EchoCanceller.h" #include "audio/AudioOutput.h" #include "audio/AudioInput.h" #include "tools/logging.h" #include "VoIPServerConfig.h" #include #include #include using namespace tgvoip; EchoCanceller::EchoCanceller(bool enableAEC, bool enableNS, bool enableAGC) { #ifndef TGVOIP_NO_DSP this->enableAEC = enableAEC; this->enableAGC = enableAGC; this->enableNS = enableNS; isOn = true; webrtc::Config extraConfig; #ifdef TGVOIP_USE_DESKTOP_DSP extraConfig.Set(new webrtc::DelayAgnostic(true)); #endif apm = webrtc::AudioProcessingBuilder().Create(extraConfig); webrtc::AudioProcessing::Config config; config.echo_canceller.enabled = enableAEC; #ifndef TGVOIP_USE_DESKTOP_DSP config.echo_canceller.mobile_mode = true; #else config.echo_canceller.mobile_mode = false; #endif config.high_pass_filter.enabled = enableAEC; config.gain_controller2.enabled = enableAGC; apm->ApplyConfig(config); webrtc::NoiseSuppression::Level nsLevel; #ifdef __APPLE__ switch (ServerConfig::GetSharedInstance()->GetInt("webrtc_ns_level_vpio", 0)) { #else switch (ServerConfig::GetSharedInstance()->GetInt("webrtc_ns_level", 2)) { #endif case 0: nsLevel = webrtc::NoiseSuppression::Level::kLow; break; case 1: nsLevel = webrtc::NoiseSuppression::Level::kModerate; break; case 3: nsLevel = webrtc::NoiseSuppression::Level::kVeryHigh; break; case 2: default: nsLevel = webrtc::NoiseSuppression::Level::kHigh; break; } apm->noise_suppression()->set_level(nsLevel); apm->noise_suppression()->Enable(enableNS); if (enableAGC) { apm->gain_control()->set_mode(webrtc::GainControl::Mode::kAdaptiveDigital); apm->gain_control()->set_target_level_dbfs(ServerConfig::GetSharedInstance()->GetInt("webrtc_agc_target_level", 9)); apm->gain_control()->enable_limiter(ServerConfig::GetSharedInstance()->GetBoolean("webrtc_agc_enable_limiter", true)); apm->gain_control()->set_compression_gain_db(ServerConfig::GetSharedInstance()->GetInt("webrtc_agc_compression_gain", 20)); } apm->voice_detection()->set_likelihood(webrtc::VoiceDetection::Likelihood::kVeryLowLikelihood); audioFrame = new webrtc::AudioFrame(); audioFrame->samples_per_channel_ = 480; audioFrame->sample_rate_hz_ = 48000; audioFrame->num_channels_ = 1; farendQueue = new BlockingQueue(11); running = true; bufferFarendThread = new Thread(std::bind(&EchoCanceller::RunBufferFarendThread, this)); bufferFarendThread->SetName("VoipECBufferFarEnd"); bufferFarendThread->Start(); #else this->enableAEC = this->enableAGC = enableAGC = this->enableNS = enableNS = false; isOn = true; #endif } EchoCanceller::~EchoCanceller() { #ifndef TGVOIP_NO_DSP delete apm; delete audioFrame; farendQueue->Put(Buffer()); bufferFarendThread->Join(); delete bufferFarendThread; #endif } void EchoCanceller::Start() { } void EchoCanceller::Stop() { } void EchoCanceller::SpeakerOutCallback(unsigned char *data, size_t len) { if (len != 960 * 2 || !enableAEC || !isOn) return; #ifndef TGVOIP_NO_DSP try { Buffer buf = farendBufferPool.Get(); buf.CopyFrom(data, 0, 960 * 2); farendQueue->Put(std::move(buf)); } catch (std::bad_alloc &x) { LOGW("Echo canceller can't keep up with real time"); } #endif } #ifndef TGVOIP_NO_DSP void EchoCanceller::RunBufferFarendThread() { webrtc::AudioFrame frame; frame.num_channels_ = 1; frame.sample_rate_hz_ = 48000; frame.samples_per_channel_ = 480; while (running) { Buffer buf = farendQueue->GetBlocking(); if (buf.IsEmpty()) { LOGI("Echo canceller buffer farend thread exiting"); return; } int16_t *samplesIn = (int16_t *)*buf; memcpy(frame.mutable_data(), samplesIn, 480 * 2); apm->ProcessReverseStream(&frame); memcpy(frame.mutable_data(), samplesIn + 480, 480 * 2); apm->ProcessReverseStream(&frame); didBufferFarend = true; } } #endif void EchoCanceller::Enable(bool enabled) { isOn = enabled; } void EchoCanceller::ProcessInput(int16_t *inOut, size_t numSamples, bool &hasVoice) { #ifndef TGVOIP_NO_DSP if (!isOn || (!enableAEC && !enableAGC && !enableNS)) { return; } int delay = audio::AudioInput::GetEstimatedDelay() + audio::AudioOutput::GetEstimatedDelay(); assert(numSamples == 960); memcpy(audioFrame->mutable_data(), inOut, 480 * 2); if (enableAEC) apm->set_stream_delay_ms(delay); apm->ProcessStream(audioFrame); if (enableVAD) hasVoice = apm->voice_detection()->stream_has_voice(); memcpy(inOut, audioFrame->data(), 480 * 2); memcpy(audioFrame->mutable_data(), inOut + 480, 480 * 2); if (enableAEC) apm->set_stream_delay_ms(delay); apm->ProcessStream(audioFrame); if (enableVAD) { hasVoice = hasVoice || apm->voice_detection()->stream_has_voice(); } memcpy(inOut + 480, audioFrame->data(), 480 * 2); #endif } void EchoCanceller::SetAECStrength(int strength) { #ifndef TGVOIP_NO_DSP /*if(aec){ #ifndef TGVOIP_USE_DESKTOP_DSP AecmConfig cfg; cfg.cngMode=AecmFalse; cfg.echoMode=(int16_t) strength; WebRtcAecm_set_config(aec, cfg); #endif }*/ #endif } void EchoCanceller::SetVoiceDetectionEnabled(bool enabled) { enableVAD = enabled; #ifndef TGVOIP_NO_DSP apm->voice_detection()->Enable(enabled); #endif } using namespace tgvoip::effects; AudioEffect::~AudioEffect() { } void AudioEffect::SetPassThrough(bool passThrough) { this->passThrough = passThrough; } Volume::Volume() { } Volume::~Volume() { } void Volume::Process(int16_t *inOut, size_t numSamples) { if (level == 1.0f || passThrough) { return; } for (size_t i = 0; i < numSamples; i++) { float sample = (float)inOut[i] * multiplier; if (sample > 32767.0f) inOut[i] = INT16_MAX; else if (sample < -32768.0f) inOut[i] = INT16_MIN; else inOut[i] = (int16_t)sample; } } void Volume::SetLevel(float level) { this->level = level; float db; if (level < 1.0f) db = -50.0f * (1.0f - level); else if (level > 1.0f && level <= 2.0f) db = 10.0f * (level - 1.0f); else db = 0.0f; multiplier = expf(db / 20.0f * logf(10.0f)); } float Volume::GetLevel() { return level; }