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TSan v2 reports data races in WebRTCAudioDeviceTest.FullDuplexAudioWithAGC

Browse Source 
BUG=226044
TEST=content_unittests in Chrome with TSan v2 enabled

Review URL: https://webrtc-codereview.appspot.com/1201010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3760 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrika@webrtc.org 2013-04-04 08:39:09 +00:00
parent 0c45957e3a
commit bb8ada686e
2 changed files with 33 additions and 12 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

28
webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
View File

@ -2053,8 +2053,11 @@ WebRtc_Word32 AudioCodingModuleImpl::IncomingPacket(
return 0;
}
} else {
if (track_neteq_buffer_)
num_bytes_accumulated_ += payload_length;
{
CriticalSectionScoped lock(acm_crit_sect_);
if (track_neteq_buffer_)
num_bytes_accumulated_ += payload_length;
}
return neteq_.RecIn(incoming_payload, payload_length, rtp_header);
}
}
@ -2136,9 +2139,12 @@ WebRtc_Word32 AudioCodingModuleImpl::SetMinimumPlayoutDelay(
"Delay must be in the range of 0-10000 milliseconds.");
return -1;
}
// Don't let the extra delay modified while accumulating buffers in NetEq.
if (track_neteq_buffer_ && first_payload_received_)
return 0;
{
CriticalSectionScoped lock(acm_crit_sect_);
// Don't let the extra delay modified while accumulating buffers in NetEq.
if (track_neteq_buffer_ && first_payload_received_)
return 0;
}
return neteq_.SetExtraDelay(time_ms);
}
@ -2665,12 +2671,14 @@ WebRtc_Word32 AudioCodingModuleImpl::PlayoutTimestamp(
WebRtc_UWord32* timestamp) {
WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
"PlayoutTimestamp()");
if (track_neteq_buffer_) {
*timestamp = playout_ts_;
return 0;
} else {
return neteq_.PlayoutTimestamp(*timestamp);
{
CriticalSectionScoped lock(acm_crit_sect_);
if (track_neteq_buffer_) {
*timestamp = playout_ts_;
return 0;
}
}
return neteq_.PlayoutTimestamp(*timestamp);
}
bool AudioCodingModuleImpl::HaveValidEncoder(const char* caller_name) const {

17
webrtc/voice_engine/transmit_mixer.cc
View File

@ -56,7 +56,18 @@ TransmitMixer::OnPeriodicProcess()
}
#endif
if (_saturationWarning)
bool saturationWarning = false;
{
// Modify |_saturationWarning| under lock to avoid conflict with write op
// in ProcessAudio and also ensure that we don't hold the lock during the
// callback.
CriticalSectionScoped cs(&_critSect);
saturationWarning = _saturationWarning;
if (_saturationWarning)
_saturationWarning = false;
}
if (saturationWarning)
{
CriticalSectionScoped cs(&_callbackCritSect);
if (_voiceEngineObserverPtr)
@ -66,7 +77,6 @@ TransmitMixer::OnPeriodicProcess()
" CallbackOnError(VE_SATURATION_WARNING)");
_voiceEngineObserverPtr->CallbackOnError(-1, VE_SATURATION_WARNING);
}
_saturationWarning = false;
}
}
@ -454,6 +464,7 @@ TransmitMixer::EncodeAndSend()
WebRtc_UWord32 TransmitMixer::CaptureLevel() const
{
CriticalSectionScoped cs(&_critSect);
return _captureLevel;
}
@ -1311,6 +1322,8 @@ void TransmitMixer::ProcessAudio(int delay_ms, int clock_drift,
LOG(LS_ERROR) << "ProcessStream() error: " << err;
}
CriticalSectionScoped cs(&_critSect);
// Store new capture level. Only updated when analog AGC is enabled.
_captureLevel = agc->stream_analog_level();