diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc index 831e1f5deb..03d9537a69 100644 --- a/media/engine/webrtc_voice_engine.cc +++ b/media/engine/webrtc_voice_engine.cc @@ -397,8 +397,7 @@ void WebRtcVoiceEngine::Init() { options.audio_jitter_buffer_max_packets = 200; options.audio_jitter_buffer_fast_accelerate = false; options.audio_jitter_buffer_min_delay_ms = 0; - bool error = ApplyOptions(options); - RTC_DCHECK(error); + ApplyOptions(options); } initialized_ = true; } @@ -419,7 +418,7 @@ VoiceMediaChannel* WebRtcVoiceEngine::CreateMediaChannel( call); } -bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) { +void WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) { RTC_DCHECK_RUN_ON(&worker_thread_checker_); RTC_LOG(LS_INFO) << "WebRtcVoiceEngine::ApplyOptions: " << options_in.ToString(); @@ -451,7 +450,6 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) { // On iOS, VPIO provides built-in AGC. options.auto_gain_control = false; RTC_LOG(LS_INFO) << "Always disable AGC on iOS. Use built-in instead."; -#elif defined(WEBRTC_ANDROID) #endif #if defined(WEBRTC_IOS) || defined(WEBRTC_ANDROID) @@ -522,35 +520,25 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) { } 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; } webrtc::AudioProcessing* ap = apm(); if (!ap) { - RTC_LOG(LS_INFO) - << "No audio processing module present. No software-provided effects " - "(AEC, NS, AGC, ...) are activated"; - return true; + return; } webrtc::AudioProcessing::Config apm_config = ap->GetConfig(); @@ -581,11 +569,9 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) { 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; } ap->ApplyConfig(apm_config); - return true; } const std::vector& WebRtcVoiceEngine::send_codecs() const { @@ -1499,11 +1485,7 @@ bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) { // on top. This means there is no way to "clear" options such that // they go back to the engine default. options_.SetAll(options); - if (!engine()->ApplyOptions(options_)) { - RTC_LOG(LS_WARNING) - << "Failed to apply engine options during channel SetOptions."; - return false; - } + engine()->ApplyOptions(options_); absl::optional audio_network_adaptor_config = GetAudioNetworkAdaptorConfig(options_); diff --git a/media/engine/webrtc_voice_engine.h b/media/engine/webrtc_voice_engine.h index e55a0a3463..9cb7ec82eb 100644 --- a/media/engine/webrtc_voice_engine.h +++ b/media/engine/webrtc_voice_engine.h @@ -91,7 +91,7 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface { // Every option that is "set" will be applied. Every option not "set" will be // ignored. This allows us to selectively turn on and off different options // easily at any time. - bool ApplyOptions(const AudioOptions& options); + void ApplyOptions(const AudioOptions& options); int CreateVoEChannel();