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Resolves TSan v2 reports data races in voe_auto_test.

Browse Source 
--- Note that I will add more fixes to this CL ---

BUG=1590

Review URL: https://webrtc-codereview.appspot.com/1286005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3770 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrika@webrtc.org
2013-04-05 14:34:57 +00:00
parent 10eb92039b
commit 19da719a5f
8 changed files with 111 additions and 49 deletions
Download Patch File Download Diff File Expand all files Collapse all files

12
webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
View File

@ -347,6 +347,7 @@ void RTCPSender::SetStartTimestamp(uint32_t start_timestamp) {
void RTCPSender::SetLastRtpTime(uint32_t rtp_timestamp,
int64_t capture_time_ms) {
CriticalSectionScoped lock(_criticalSectionRTCPSender);
last_rtp_timestamp_ = rtp_timestamp;
if (capture_time_ms < 0) {
// We don't currently get a capture time from VoiceEngine.
@ -631,13 +632,18 @@ RTCPSender::BuildSR(WebRtc_UWord8* rtcpbuffer,
if(_audio) {
freqHz = _rtpRtcp.CurrentSendFrequencyHz();
}
// The timestamp of this RTCP packet should be estimated as the timestamp of
// the frame being captured at this moment. We are calculating that
// timestamp as the last frame's timestamp + the time since the last frame
// was captured.
RTPtime = start_timestamp_ + last_rtp_timestamp_ + (
_clock->TimeInMilliseconds() - last_frame_capture_time_ms_) *
(freqHz / 1000);
{
// Needs protection since this method is called on the process thread.
CriticalSectionScoped lock(_criticalSectionRTCPSender);
RTPtime = start_timestamp_ + last_rtp_timestamp_ + (
_clock->TimeInMilliseconds() - last_frame_capture_time_ms_) *
(freqHz / 1000);
}
// Add sender data
// Save for our length field

41
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -309,19 +309,31 @@ WebRtc_Word32 ModuleRtpRtcpImpl::Process() {
}
void ModuleRtpRtcpImpl::ProcessDeadOrAliveTimer() {
if (dead_or_alive_active_) {
const WebRtc_Word64 now = clock_->TimeInMilliseconds();
if (now > dead_or_alive_timeout_ms_ + dead_or_alive_last_timer_) {
// RTCP is alive if we have received a report the last 12 seconds.
dead_or_alive_last_timer_ += dead_or_alive_timeout_ms_;
bool RTCPalive = false;
if (rtcp_receiver_.LastReceived() + 12000 > now) {
RTCPalive = true;
bool RTCPalive = false;
WebRtc_Word64 now = 0;
bool do_callback = false;
// Do operations on members under lock but avoid making the
// ProcessDeadOrAlive() callback under the same lock.
{
CriticalSectionScoped lock(critical_section_module_ptrs_.get());
if (dead_or_alive_active_) {
now = clock_->TimeInMilliseconds();
if (now > dead_or_alive_timeout_ms_ + dead_or_alive_last_timer_) {
// RTCP is alive if we have received a report the last 12 seconds.
dead_or_alive_last_timer_ += dead_or_alive_timeout_ms_;
if (rtcp_receiver_.LastReceived() + 12000 > now)
RTCPalive = true;
do_callback = true;
}
rtp_receiver_->ProcessDeadOrAlive(RTCPalive, now);
}
}
if (do_callback)
rtp_receiver_->ProcessDeadOrAlive(RTCPalive, now);
}
WebRtc_Word32 ModuleRtpRtcpImpl::SetPeriodicDeadOrAliveStatus(
@ -342,10 +354,13 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SetPeriodicDeadOrAliveStatus(
if (sample_time_seconds == 0) {
return -1;
}
dead_or_alive_active_ = enable;
dead_or_alive_timeout_ms_ = sample_time_seconds * 1000;
// Trigger the first after one period.
dead_or_alive_last_timer_ = clock_->TimeInMilliseconds();
{
CriticalSectionScoped lock(critical_section_module_ptrs_.get());
dead_or_alive_active_ = enable;
dead_or_alive_timeout_ms_ = sample_time_seconds * 1000;
// Trigger the first after one period.
dead_or_alive_last_timer_ = clock_->TimeInMilliseconds();
}
return 0;
}