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TSan v2 reports data races in WebRTCAudioDeviceTest.FullDuplexAudioWithAGC

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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
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12
webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
View File

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

15
webrtc/voice_engine/transmit_mixer.cc
View File

@@ -56,7 +56,18 @@ TransmitMixer::OnPeriodicProcess()
} }
#endif #endif
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) if (_saturationWarning)
_saturationWarning = false;
}
if (saturationWarning)
{ {
CriticalSectionScoped cs(&_callbackCritSect); CriticalSectionScoped cs(&_callbackCritSect);
if (_voiceEngineObserverPtr) if (_voiceEngineObserverPtr)
@@ -66,7 +77,6 @@ TransmitMixer::OnPeriodicProcess()
" CallbackOnError(VE_SATURATION_WARNING)"); " CallbackOnError(VE_SATURATION_WARNING)");
_voiceEngineObserverPtr->CallbackOnError(-1, VE_SATURATION_WARNING); _voiceEngineObserverPtr->CallbackOnError(-1, VE_SATURATION_WARNING);
} }
_saturationWarning = false;
} }
} }
@@ -454,6 +464,7 @@ TransmitMixer::EncodeAndSend()
WebRtc_UWord32 TransmitMixer::CaptureLevel() const WebRtc_UWord32 TransmitMixer::CaptureLevel() const
{ {
CriticalSectionScoped cs(&_critSect);
return _captureLevel; return _captureLevel;
} }
@@ -1311,6 +1322,8 @@ void TransmitMixer::ProcessAudio(int delay_ms, int clock_drift,
LOG(LS_ERROR) << "ProcessStream() error: " << err; LOG(LS_ERROR) << "ProcessStream() error: " << err;
} }
CriticalSectionScoped cs(&_critSect);
// Store new capture level. Only updated when analog AGC is enabled. // Store new capture level. Only updated when analog AGC is enabled.
_captureLevel = agc->stream_analog_level(); _captureLevel = agc->stream_analog_level();