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}
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index 831e1f5..03d9537 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -397,8 +397,7 @@
     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 @@
                                      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 @@
   // 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 @@
   }
 
   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 @@
     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 @@
   // 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 e55a0a3..9cb7ec8 100644
--- a/media/engine/webrtc_voice_engine.h
+++ b/media/engine/webrtc_voice_engine.h
@@ -91,7 +91,7 @@
   // 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();