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Backed out changeset daf5df0306b2 (bug 985714)
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@ -40,7 +40,6 @@
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#include "webrtc/voice_engine/include/voe_volume_control.h"
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#include "webrtc/voice_engine/include/voe_external_media.h"
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#include "webrtc/voice_engine/include/voe_audio_processing.h"
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#include "webrtc/voice_engine/include/voe_call_report.h"
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// Video Engine
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#include "webrtc/video_engine/include/vie_base.h"
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@ -262,7 +261,6 @@ public:
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, mChannel(-1)
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, mInitDone(false)
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, mStarted(false)
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, mSamples(0)
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, mEchoOn(false), mAgcOn(false), mNoiseOn(false)
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, mEchoCancel(webrtc::kEcDefault)
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, mAGC(webrtc::kAgcDefault)
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@ -319,7 +317,6 @@ private:
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ScopedCustomReleasePtr<webrtc::VoEExternalMedia> mVoERender;
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ScopedCustomReleasePtr<webrtc::VoENetwork> mVoENetwork;
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ScopedCustomReleasePtr<webrtc::VoEAudioProcessing> mVoEProcessing;
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ScopedCustomReleasePtr<webrtc::VoECallReport> mVoECallReport;
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// mMonitor protects mSources[] access/changes, and transitions of mState
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// from kStarted to kStopped (which are combined with EndTrack()).
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@ -332,7 +329,6 @@ private:
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TrackID mTrackID;
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bool mInitDone;
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bool mStarted;
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int mSamples; // int to avoid conversions when comparing/etc to samplingFreq & length
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nsString mDeviceName;
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nsString mDeviceUUID;
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@ -221,13 +221,8 @@ MediaEngineWebRTCAudioSource::Config(bool aEchoOn, uint32_t aEcho,
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if (update_echo &&
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0 != (error = mVoEProcessing->SetEcStatus(mEchoOn, (webrtc::EcModes) aEcho))) {
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LOG(("%s Error setting Echo Status: %d ",__FUNCTION__, error));
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// Overhead of capturing all the time is very low (<0.1% of an audio only call)
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if (mEchoOn) {
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if (0 != (error = mVoEProcessing->SetEcMetricsStatus(true))) {
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LOG(("%s Error setting Echo Metrics: %d ",__FUNCTION__, error));
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}
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}
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}
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if (update_agc &&
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0 != (error = mVoEProcessing->SetAgcStatus(mAgcOn, (webrtc::AgcModes) aAGC))) {
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LOG(("%s Error setting AGC Status: %d ",__FUNCTION__, error));
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@ -410,11 +405,6 @@ MediaEngineWebRTCAudioSource::Init()
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return;
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}
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mVoECallReport = webrtc::VoECallReport::GetInterface(mVoiceEngine);
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if (!mVoECallReport) {
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return;
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}
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mChannel = mVoEBase->CreateChannel();
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if (mChannel < 0) {
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return;
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@ -537,24 +527,6 @@ MediaEngineWebRTCAudioSource::Process(int channel,
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free(buffer);
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}
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#ifdef PR_LOGGING
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mSamples += length;
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if (mSamples > samplingFreq) {
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mSamples %= samplingFreq; // just in case mSamples >> samplingFreq
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if (PR_LOG_TEST(GetMediaManagerLog(), PR_LOG_DEBUG)) {
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webrtc::EchoStatistics echo;
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mVoECallReport->GetEchoMetricSummary(echo);
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#define DUMP_STATVAL(x) (x).min, (x).max, (x).average
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LOG(("Echo: ERL: %d/%d/%d, ERLE: %d/%d/%d, RERL: %d/%d/%d, NLP: %d/%d/%d",
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DUMP_STATVAL(echo.erl),
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DUMP_STATVAL(echo.erle),
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DUMP_STATVAL(echo.rerl),
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DUMP_STATVAL(echo.a_nlp)));
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}
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}
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#endif
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MonitorAutoLock lock(mMonitor);
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if (mState != kStarted)
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return;
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