Clean out unused methods from VoiceEngine and VoEBase.

media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 272 matching lines @@
   recv_codecs_ = CollectCodecs(decoder_factory_->GetSupportedDecoders());
   for (const AudioCodec& codec : recv_codecs_) {
     LOG(LS_INFO) << ToString(codec);
   }
 
   channel_config_.enable_voice_pacing = true;
 
   // Temporarily turn logging level up for the Init() call.
   webrtc::Trace::SetTraceCallback(this);
   webrtc::Trace::set_level_filter(kElevatedTraceFilter);
-  LOG(LS_INFO) << webrtc::VoiceEngine::GetVersionString();
   RTC_CHECK_EQ(0,
                voe_wrapper_->base()->Init(adm_.get(), apm(), decoder_factory_));
   webrtc::Trace::set_level_filter(kDefaultTraceFilter);
 
   // No ADM supplied? Get the default one from VoE.
   if (!adm_) {
     adm_ = voe_wrapper_->base()->audio_device_module();
   }
   RTC_DCHECK(adm_);
 
@@ skipping 317 matching lines @@
     apm_config.residual_echo_detector.enabled = *options.residual_echo_detector;
   }
 
   apm()->SetExtraOptions(config);
   apm()->ApplyConfig(apm_config);
 
   if (options.recording_sample_rate) {
     LOG(LS_INFO) << "Recording sample rate is "
                  << *options.recording_sample_rate;
     if (adm()->SetRecordingSampleRate(*options.recording_sample_rate)) {
-      LOG_RTCERR1(SetRecordingSampleRate, *options.recording_sample_rate);
+      LOG(LS_WARNING) << "SetRecordingSampleRate("
+                      << *options.recording_sample_rate << ") failed, err="
+                      << adm()->LastError();
     }
   }
 
   if (options.playout_sample_rate) {
     LOG(LS_INFO) << "Playout sample rate is " << *options.playout_sample_rate;
     if (adm()->SetPlayoutSampleRate(*options.playout_sample_rate)) {
-      LOG_RTCERR1(SetPlayoutSampleRate, *options.playout_sample_rate);
+      LOG(LS_WARNING) << "SetPlayoutSampleRate("
+                      << *options.playout_sample_rate << ") failed, err="
+                      << adm()->LastError();
     }
   }
   return true;
 }
 
 // TODO(solenberg): Remove, once AudioMonitor is gone.
 int WebRtcVoiceEngine::GetInputLevel() {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   int8_t level = transmit_mixer()->AudioLevel();
   RTC_DCHECK_LE(0, level);
@@ skipping 17 matching lines @@
       webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri,
                            webrtc::RtpExtension::kAudioLevelDefaultId));
   if (webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe")) {
     capabilities.header_extensions.push_back(webrtc::RtpExtension(
         webrtc::RtpExtension::kTransportSequenceNumberUri,
         webrtc::RtpExtension::kTransportSequenceNumberDefaultId));
   }
   return capabilities;
 }
 
-int WebRtcVoiceEngine::GetLastEngineError() {
-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  return voe_wrapper_->error();
-}
-
 void WebRtcVoiceEngine::Print(webrtc::TraceLevel level, const char* trace,
                               int length) {
   // Note: This callback can happen on any thread!
   rtc::LoggingSeverity sev = rtc::LS_VERBOSE;
   if (level == webrtc::kTraceError || level == webrtc::kTraceCritical)
     sev = rtc::LS_ERROR;
   else if (level == webrtc::kTraceWarning)
     sev = rtc::LS_WARNING;
   else if (level == webrtc::kTraceStateInfo || level == webrtc::kTraceInfo)
     sev = rtc::LS_INFO;
@@ skipping 1134 matching lines @@
   }
   if (enable && options) {
     return SetOptions(*options);
   }
   return true;
 }
 
 int WebRtcVoiceMediaChannel::CreateVoEChannel() {
   int id = engine()->CreateVoEChannel();
   if (id == -1) {
-    LOG_RTCERR0(CreateVoEChannel);
+    LOG(LS_WARNING) << "CreateVoEChannel() failed.";
     return -1;
   }
 
   return id;
 }
 
 bool WebRtcVoiceMediaChannel::DeleteVoEChannel(int channel) {
   if (engine()->voe()->base()->DeleteChannel(channel) == -1) {
-    LOG_RTCERR1(DeleteChannel, channel);
+    LOG(LS_WARNING) << "DeleteChannel(" << channel << ") failed.";
     return false;
   }
   return true;
 }
 
 bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) {
   TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::AddSendStream");
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "AddSendStream: " << sp.ToString();
 
@@ skipping 522 matching lines @@
                       ssrc);
   if (it != unsignaled_recv_ssrcs_.end()) {
     unsignaled_recv_ssrcs_.erase(it);
     return true;
   }
   return false;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE