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Added critical section to prevent race.

Browse Source 
Review URL: https://webrtc-codereview.appspot.com/595004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2289 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pwestin@webrtc.org
2012-05-24 11:28:31 +00:00
parent 5c3a400fae
commit 2d1fc9bf17
2 changed files with 18 additions and 0 deletions
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17
src/video_engine/vie_channel.cc
View File

@@ -46,6 +46,7 @@ ViEChannel::ViEChannel(WebRtc_Word32 channel_id,
number_of_cores_(number_of_cores),
num_socket_threads_(kViESocketThreads),
callback_cs_(CriticalSectionWrapper::CreateCriticalSection()),
rtp_rtcp_cs_(CriticalSectionWrapper::CreateCriticalSection()),
default_rtp_rtcp_(default_rtp_rtcp),
rtp_rtcp_(NULL),
#ifndef WEBRTC_EXTERNAL_TRANSPORT
@@ -220,9 +221,12 @@ WebRtc_Word32 ViEChannel::SetSendCodec(const VideoCodec& video_codec,
restart_rtp = true;
rtp_rtcp_->SetSendingStatus(false);
}
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
if (video_codec.numberOfSimulcastStreams > 0) {
// Set correct bitrate to base layer.
// Create our simulcast RTP modules.
for (int i = simulcast_rtp_rtcp_.size();
i < video_codec.numberOfSimulcastStreams - 1;
i++) {
@@ -475,6 +479,7 @@ WebRtc_Word32 ViEChannel::SetRTCPMode(const RTCPMethod rtcp_mode) {
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
"%s: %d", __FUNCTION__, rtcp_mode);
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
it != simulcast_rtp_rtcp_.end();
it++) {
@@ -540,6 +545,8 @@ WebRtc_Word32 ViEChannel::ProcessNACKRequest(const bool enable) {
vcm_.RegisterPacketRequestCallback(this);
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
it != simulcast_rtp_rtcp_.end();
it++) {
@@ -547,6 +554,7 @@ WebRtc_Word32 ViEChannel::ProcessNACKRequest(const bool enable) {
rtp_rtcp->SetStorePacketsStatus(true, kNackHistorySize);
}
} else {
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
it != simulcast_rtp_rtcp_.end();
it++) {
@@ -590,6 +598,7 @@ WebRtc_Word32 ViEChannel::ProcessFECRequest(
enable);
return -1;
}
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
it != simulcast_rtp_rtcp_.end();
it++) {
@@ -665,6 +674,7 @@ WebRtc_Word32 ViEChannel::SetSSRC(const WebRtc_UWord32 SSRC,
if (simulcast_idx == 0) {
return rtp_rtcp_->SetSSRC(SSRC);
}
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
if (simulcast_idx > simulcast_rtp_rtcp_.size()) {
return -1;
}
@@ -862,6 +872,8 @@ WebRtc_Word32 ViEChannel::GetSendRtcpStatistics(WebRtc_UWord16& fraction_lost,
// TODO(pwestin) how do we do this for simulcast ? average for all
// except cumulative_lost that is the sum ?
// CriticalSectionScoped cs(rtp_rtcp_cs_.get());
// for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
// it != simulcast_rtp_rtcp_.end();
// it++) {
@@ -955,6 +967,7 @@ WebRtc_Word32 ViEChannel::GetRtpStatistics(
"%s: Could not get counters", __FUNCTION__);
return -1;
}
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
it != simulcast_rtp_rtcp_.end();
it++) {
@@ -979,6 +992,7 @@ void ViEChannel::GetBandwidthUsage(WebRtc_UWord32& total_bitrate_sent,
&video_bitrate_sent,
&fec_bitrate_sent,
&nackBitrateSent);
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
it != simulcast_rtp_rtcp_.end(); it++) {
WebRtc_UWord32 stream_rate = 0;
@@ -1305,6 +1319,7 @@ WebRtc_Word32 ViEChannel::StartSend() {
"%s: Could not start sending RTP", __FUNCTION__);
return -1;
}
CriticalSectionScoped cs_rtp(rtp_rtcp_cs_.get());
for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
it != simulcast_rtp_rtcp_.end();
it++) {
@@ -1319,6 +1334,7 @@ WebRtc_Word32 ViEChannel::StopSend() {
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
__FUNCTION__);
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
rtp_rtcp_->SetSendingMediaStatus(false);
for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
it != simulcast_rtp_rtcp_.end();
@@ -1791,6 +1807,7 @@ WebRtc_Word32 ViEChannel::SetMTU(WebRtc_UWord16 mtu) {
// Logging done.
return -1;
}
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
it != simulcast_rtp_rtcp_.end();
it++) {

1
src/video_engine/vie_channel.h
View File

@@ -348,6 +348,7 @@ class ViEChannel
// Used for all registered callbacks except rendering.
scoped_ptr<CriticalSectionWrapper> callback_cs_;
scoped_ptr<CriticalSectionWrapper> rtp_rtcp_cs_;
RtpRtcp* default_rtp_rtcp_;