WebRTC音视频之噪音消除功能

WebRTC对于iOS和Android的音频处理，是有很大的不同的，WebRTC基本上是使用的iOS自身的音频降噪、环境音处理、人声增益处理，因为WebRTC认为iOS的音频处理已经满足相对理想的效果，所以在算法处理上进行了iOS和Android的区分。

但是我研究发现，如果将WebRTC用于Android的算法，也用于iOS，在部分手机上还是有相当的效果，现在就是将修改的噪音消除的代码奉上，以供参考。

bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {RTC_DCHECK(worker_thread_checker_.IsCurrent());RTC_LOG(LS_INFO) << "WebRtcVoiceEngine::ApplyOptions: "<< options_in.ToString();AudioOptions options = options_in;  // The options are modified below.// Set and adjust echo canceller options.// kEcConference is AEC with high suppression.webrtc::EcModes ec_mode = webrtc::kEcConference;#if defined(WEBRTC_IOS)if (options.ios_force_software_aec_HACK &&*options.ios_force_software_aec_HACK) {// EC may be forced on for a device known to have non-functioning platform// AEC.options.echo_cancellation = true;RTC_LOG(LS_WARNING)<< "Force software AEC on iOS. May conflict with platform AEC.";} else {// On iOS, VPIO provides built-in EC.options.echo_cancellation = false;RTC_LOG(LS_INFO) << "Always disable AEC on iOS. Use built-in instead.";}
#elif defined(WEBRTC_ANDROID)ec_mode = webrtc::kEcAecm;
#endif// Set and adjust noise suppressor options.
#if defined(WEBRTC_IOS)// On iOS, VPIO provides built-in NS.//修改处//options.noise_suppression = false;options.typing_detection = false;options.experimental_ns = false;RTC_LOG(LS_INFO) << "Always disable NS on iOS. Use built-in instead.";
#elif defined(WEBRTC_ANDROID)options.typing_detection = false;options.experimental_ns = false;
#endif// Set and adjust gain control options.
#if defined(WEBRTC_IOS)// On iOS, VPIO provides built-in AGC.//修改处//options.auto_gain_control = false;options.experimental_agc = false;RTC_LOG(LS_INFO) << "Always disable AGC on iOS. Use built-in instead.";
#elif defined(WEBRTC_ANDROID)options.experimental_agc = false;
#endif#if defined(WEBRTC_IOS) || defined(WEBRTC_ANDROID)// Turn off the gain control if specified by the field trial.// The purpose of the field trial is to reduce the amount of resampling// performed inside the audio processing module on mobile platforms by// whenever possible turning off the fixed AGC mode and the high-pass filter.// (https://bugs.chromium.org/p/webrtc/issues/detail?id=6181).if (webrtc::field_trial::IsEnabled("WebRTC-Audio-MinimizeResamplingOnMobile")) {options.auto_gain_control = false;RTC_LOG(LS_INFO) << "Disable AGC according to field trial.";if (!(options.noise_suppression.value_or(false) ||options.echo_cancellation.value_or(false))) {// If possible, turn off the high-pass filter.RTC_LOG(LS_INFO)<< "Disable high-pass filter in response to field trial.";options.highpass_filter = false;}}
#endifif (options.echo_cancellation) {// Check if platform supports built-in EC. Currently only supported on// Android and in combination with Java based audio layer.// TODO(henrika): investigate possibility to support built-in EC also// in combination with Open SL ES audio.const bool built_in_aec = adm()->BuiltInAECIsAvailable();if (built_in_aec) {// Built-in EC exists on this device. Enable/Disable it according to the// echo_cancellation audio option.const bool enable_built_in_aec = *options.echo_cancellation;if (adm()->EnableBuiltInAEC(enable_built_in_aec) == 0 &&enable_built_in_aec) {// Disable internal software EC if built-in EC is enabled,// i.e., replace the software EC with the built-in EC.options.echo_cancellation = false;RTC_LOG(LS_INFO)<< "Disabling EC since built-in EC will be used instead";}}webrtc::apm_helpers::SetEcStatus(apm(), *options.echo_cancellation,ec_mode);}if (options.auto_gain_control) {bool built_in_agc_avaliable = adm()->BuiltInAGCIsAvailable();if (built_in_agc_avaliable) {if (adm()->EnableBuiltInAGC(*options.auto_gain_control) == 0 &&*options.auto_gain_control) {// Disable internal software AGC if built-in AGC is enabled,// i.e., replace the software AGC with the built-in AGC.options.auto_gain_control = false;RTC_LOG(LS_INFO)<< "Disabling AGC since built-in AGC will be used instead";}}}if (options.noise_suppression) {if (adm()->BuiltInNSIsAvailable()) {bool builtin_ns = *options.noise_suppression;if (adm()->EnableBuiltInNS(builtin_ns) == 0 && builtin_ns) {// Disable internal software NS if built-in NS is enabled,// i.e., replace the software NS with the built-in NS.options.noise_suppression = false;RTC_LOG(LS_INFO)<< "Disabling NS since built-in NS will be used instead";}}}if (options.stereo_swapping) {RTC_LOG(LS_INFO) << "Stereo swapping enabled? " << *options.stereo_swapping;audio_state()->SetStereoChannelSwapping(*options.stereo_swapping);}if (options.audio_jitter_buffer_max_packets) {RTC_LOG(LS_INFO) << "NetEq capacity is "<< *options.audio_jitter_buffer_max_packets;audio_jitter_buffer_max_packets_ =std::max(20, *options.audio_jitter_buffer_max_packets);}if (options.audio_jitter_buffer_fast_accelerate) {RTC_LOG(LS_INFO) << "NetEq fast mode? "<< *options.audio_jitter_buffer_fast_accelerate;audio_jitter_buffer_fast_accelerate_ =*options.audio_jitter_buffer_fast_accelerate;}if (options.audio_jitter_buffer_min_delay_ms) {RTC_LOG(LS_INFO) << "NetEq minimum delay is "<< *options.audio_jitter_buffer_min_delay_ms;audio_jitter_buffer_min_delay_ms_ =*options.audio_jitter_buffer_min_delay_ms;}if (options.audio_jitter_buffer_enable_rtx_handling) {RTC_LOG(LS_INFO) << "NetEq handle reordered packets? "<< *options.audio_jitter_buffer_enable_rtx_handling;audio_jitter_buffer_enable_rtx_handling_ =*options.audio_jitter_buffer_enable_rtx_handling;}webrtc::Config config;if (options.experimental_ns) {experimental_ns_ = options.experimental_ns;}if (experimental_ns_) {RTC_LOG(LS_INFO) << "Experimental ns is enabled? " << *experimental_ns_;config.Set<webrtc::ExperimentalNs>(new webrtc::ExperimentalNs(*experimental_ns_));}webrtc::AudioProcessing::Config apm_config = apm()->GetConfig();if (options.auto_gain_control) {const bool enabled = *options.auto_gain_control;apm_config.gain_controller1.enabled = enabled;RTC_LOG(LS_INFO) << "Setting AGC to " << enabled;}if (options.tx_agc_target_dbov) {apm_config.gain_controller1.target_level_dbfs = *options.tx_agc_target_dbov;}if (options.tx_agc_digital_compression_gain) {apm_config.gain_controller1.compression_gain_db =*options.tx_agc_digital_compression_gain;}if (options.tx_agc_limiter) {apm_config.gain_controller1.enable_limiter = *options.tx_agc_limiter;}if (options.highpass_filter) {apm_config.high_pass_filter.enabled = *options.highpass_filter;}if (options.residual_echo_detector) {apm_config.residual_echo_detector.enabled = *options.residual_echo_detector;}if (options.noise_suppression) {const bool enabled = *options.noise_suppression;apm_config.noise_suppression.enabled = enabled;apm_config.noise_suppression.level =webrtc::AudioProcessing::Config::NoiseSuppression::Level::kHigh;RTC_LOG(LS_INFO) << "NS set to " << enabled;}if (options.typing_detection) {RTC_LOG(LS_INFO) << "Typing detection is enabled? "<< *options.typing_detection;apm_config.voice_detection.enabled = *options.typing_detection;}apm()->SetExtraOptions(config);apm()->ApplyConfig(apm_config);return true;
}

使用方式：

NSMutableDictionary *mandatory = [NSMutableDictionary dictionary];if (_closeNoiseSuppression) {[mandatory setValue:@"false" forKey:@"googNoiseSuppression"];[mandatory setValue:@"false" forKey:@"googAutoGainControl"];}NSMutableDictionary *optional = [NSMutableDictionary dictionary];RTCMediaConstraints *constraints = [[RTCMediaConstraints alloc] initWithMandatoryConstraints:mandatory optionalConstraints:optional];_audioSource = [_factory audioSourceWithConstraints:constraints];_audioTrack = [_factory audioTrackWithSource:_audioSource trackId:kSmoothAudioTrackId];

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