From da6291e85ddccb81b2b78685e2bb2946eacce63c Mon Sep 17 00:00:00 2001
From: Sam Zackrisson <saza@webrtc.org>
Date: Fri, 5 Aug 2022 10:38:02 +0200
Subject: [PATCH] WebRTC voice engine: Remove duplicate and confusing logs

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}
---
 media/engine/webrtc_voice_engine.cc | 26 ++++----------------------
 media/engine/webrtc_voice_engine.h  |  2 +-
 2 files changed, 5 insertions(+), 23 deletions(-)

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<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_);
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();