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WebRTC voice engine: Remove duplicate and confusing logs

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The line "No audio processing module present [...]" has mislead people several times (see linked bug for one example) and does not add any information that cannot already relatively easily be inferred from platform configuration.

Other removed lines are duplicate (already logged via AudioOptions::ToString()) or never runs (ApplyOptions always returns true + empty #elif).

Bug: b/238780321#comment34
Change-Id: Ie0fbd6675ec963c1180a7f614ec74bba5e850777
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270483
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37697}
This commit is contained in:
Sam Zackrisson 2022-08-05 10:38:02 +02:00 committed by WebRTC LUCI CQ
parent e94b175c42
commit da6291e85d
2 changed files with 5 additions and 23 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

26
media/engine/webrtc_voice_engine.cc
View File

@ -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<AudioCodec>& 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<std::string> audio_network_adaptor_config =
GetAudioNetworkAdaptorConfig(options_);

2
media/engine/webrtc_voice_engine.h
View File

@ -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();