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Fix tsan failures in channel.cc regarding to the volume settings.

Browse Source 
BUG=2461
TEST=try bots
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2377004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4992 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
wu@webrtc.org 2013-10-17 18:28:55 +00:00
parent b22049b295
commit 6342066974
3 changed files with 25 additions and 53 deletions
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47
tools/valgrind-webrtc/tsan/suppressions.txt
View File

@ -47,26 +47,6 @@
fun:webrtc::ThreadPosix::Run
fun:webrtc::StartThread
}
{
bug_2461_1
ThreadSanitizer:Race
fun:webrtc::voe::Channel::SetChannelOutputVolumeScaling
fun:webrtc::VoEVolumeControlImpl::SetChannelOutputVolumeScaling
fun:cricket::WebRtcVoiceMediaChannel::SetOutputScaling
fun:cricket::VoiceChannel::SetOutputScaling_w
fun:cricket::VoiceChannel::OnMessage
...
}
{
bug_2461_2
ThreadSanitizer:Race
fun:webrtc::voe::Channel::SetOutputVolumePan
fun:webrtc::VoEVolumeControlImpl::SetOutputVolumePan
fun:cricket::WebRtcVoiceMediaChannel::SetOutputScaling
fun:cricket::VoiceChannel::SetOutputScaling_w
fun:cricket::VoiceChannel::OnMessage
...
}
{
bug_2497
ThreadSanitizer:Race
@ -146,16 +126,6 @@
fun:webrtc::ThreadPosix::Run
fun:StartThread
}
{
bug_1205_12
ThreadSanitizer:Race
fun:webrtc::voe::Channel::GetAudioFrame
fun:webrtc::AudioConferenceMixerImpl::UpdateToMix
fun:webrtc::AudioConferenceMixerImpl::Process
fun:webrtc::voe::OutputMixer::MixActiveChannels
fun:webrtc::VoEBaseImpl::NeedMorePlayData
...
}
{
bug_1205_22
ThreadSanitizer:Race
@ -164,23 +134,6 @@
fun:webrtc::ThreadPosix::Run
fun:StartThread
}
{
bug_1205_25
ThreadSanitizer:Race
fun:webrtc::AudioCodingModuleImpl::PlayoutData10Ms
fun:webrtc::voe::Channel::GetAudioFrame
fun:webrtc::AudioConferenceMixerImpl::UpdateToMix
fun:webrtc::AudioConferenceMixerImpl::Process
fun:webrtc::voe::OutputMixer::MixActiveChannels
fun:webrtc::VoEBaseImpl::NeedMorePlayData
fun:FakeAudioCaptureModule::ReceiveFrameP
fun:FakeAudioCaptureModule::ProcessFrameP
fun:FakeAudioCaptureModule::OnMessage
fun:talk_base::MessageQueue::Dispatch
fun:talk_base::Thread::ProcessMessages
fun:talk_base::Thread::Run
fun:talk_base::Thread::PreRun
}
{
bug_1205_26
ThreadSanitizer:Race

30
webrtc/voice_engine/channel.cc
View File

@ -703,16 +703,26 @@ int32_t Channel::GetAudioFrame(int32_t id, AudioFrame& audioFrame)
ApmProcessRx(audioFrame);
}
// Output volume scaling
if (_outputGain < 0.99f || _outputGain > 1.01f)
float output_gain = 1.0f;
float left_pan = 1.0f;
float right_pan = 1.0f;
{
AudioFrameOperations::ScaleWithSat(_outputGain, audioFrame);
CriticalSectionScoped cs(&volume_settings_critsect_);
output_gain = _outputGain;
left_pan = _panLeft;
right_pan= _panRight;
}
// Output volume scaling
if (output_gain < 0.99f || output_gain > 1.01f)
{
AudioFrameOperations::ScaleWithSat(output_gain, audioFrame);
}
// Scale left and/or right channel(s) if stereo and master balance is
// active
if (_panLeft != 1.0f || _panRight != 1.0f)
if (left_pan != 1.0f || right_pan != 1.0f)
{
if (audioFrame.num_channels_ == 1)
{
@ -725,7 +735,7 @@ int32_t Channel::GetAudioFrame(int32_t id, AudioFrame& audioFrame)
// Do the panning operation (the audio frame contains stereo at this
// stage)
AudioFrameOperations::Scale(_panLeft, _panRight, audioFrame);
AudioFrameOperations::Scale(left_pan, right_pan, audioFrame);
}
// Mix decoded PCM output with file if file mixing is enabled
@ -905,6 +915,7 @@ Channel::Channel(int32_t channelId,
const Config& config) :
_fileCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
_callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
volume_settings_critsect_(*CriticalSectionWrapper::CreateCriticalSection()),
_instanceId(instanceId),
_channelId(channelId),
rtp_header_parser_(RtpHeaderParser::Create()),
@ -1103,6 +1114,7 @@ Channel::~Channel()
delete [] _decryptionRTCPBufferPtr;
delete &_callbackCritSect;
delete &_fileCritSect;
delete &volume_settings_critsect_;
}
int32_t
@ -2957,6 +2969,7 @@ Channel::GetSpeechOutputLevelFullRange(uint32_t& level) const
int
Channel::SetMute(bool enable)
{
CriticalSectionScoped cs(&volume_settings_critsect_);
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetMute(enable=%d)", enable);
_mute = enable;
@ -2966,12 +2979,14 @@ Channel::SetMute(bool enable)
bool
Channel::Mute() const
{
CriticalSectionScoped cs(&volume_settings_critsect_);
return _mute;
}
int
Channel::SetOutputVolumePan(float left, float right)
{
CriticalSectionScoped cs(&volume_settings_critsect_);
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetOutputVolumePan()");
_panLeft = left;
@ -2982,6 +2997,7 @@ Channel::SetOutputVolumePan(float left, float right)
int
Channel::GetOutputVolumePan(float& left, float& right) const
{
CriticalSectionScoped cs(&volume_settings_critsect_);
left = _panLeft;
right = _panRight;
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
@ -2993,6 +3009,7 @@ Channel::GetOutputVolumePan(float& left, float& right) const
int
Channel::SetChannelOutputVolumeScaling(float scaling)
{
CriticalSectionScoped cs(&volume_settings_critsect_);
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetChannelOutputVolumeScaling()");
_outputGain = scaling;
@ -3002,6 +3019,7 @@ Channel::SetChannelOutputVolumeScaling(float scaling)
int
Channel::GetChannelOutputVolumeScaling(float& scaling) const
{
CriticalSectionScoped cs(&volume_settings_critsect_);
scaling = _outputGain;
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
VoEId(_instanceId,_channelId),
@ -4397,7 +4415,7 @@ Channel::PrepareEncodeAndSend(int mixingFrequency)
MixOrReplaceAudioWithFile(mixingFrequency);
}
if (_mute)
if (Mute())
{
AudioFrameOperations::Mute(_audioFrame);
}

1
webrtc/voice_engine/channel.h
View File

@ -447,6 +447,7 @@ private:
CriticalSectionWrapper& _fileCritSect;
CriticalSectionWrapper& _callbackCritSect;
CriticalSectionWrapper& volume_settings_critsect_;
uint32_t _instanceId;
int32_t _channelId;