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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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28
webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
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@@ -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 {