Index: media/engine/webrtcvoiceengine.cc |
diff --git a/media/engine/webrtcvoiceengine.cc b/media/engine/webrtcvoiceengine.cc |
index 271ace12125eb61acd5b1c44d8077136f505f64e..8fa5fe2654ab2ab2391809b78155507a595cee02 100644 |
--- a/media/engine/webrtcvoiceengine.cc |
+++ b/media/engine/webrtcvoiceengine.cc |
@@ -290,7 +290,6 @@ void WebRtcVoiceEngine::Init() { |
// 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); |
@@ -628,14 +627,18 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) { |
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; |
@@ -673,11 +676,6 @@ RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const { |
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! |
@@ -1832,7 +1830,7 @@ bool WebRtcVoiceMediaChannel::SetAudioSend(uint32_t ssrc, |
int WebRtcVoiceMediaChannel::CreateVoEChannel() { |
int id = engine()->CreateVoEChannel(); |
if (id == -1) { |
- LOG_RTCERR0(CreateVoEChannel); |
+ LOG(LS_WARNING) << "CreateVoEChannel() failed."; |
return -1; |
} |
@@ -1841,7 +1839,7 @@ int WebRtcVoiceMediaChannel::CreateVoEChannel() { |
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; |