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RW lock access to ssrc maps in VideoCall.

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BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1640004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4202 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org 2013-06-10 13:48:26 +00:00
parent adb51f5709
commit 1819fd711a
2 changed files with 30 additions and 24 deletions
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46
webrtc/video_engine/internal/video_call.cc
View File

@ -29,7 +29,10 @@ namespace internal {
VideoCall::VideoCall(webrtc::VideoEngine* video_engine,
newapi::Transport* send_transport)
: send_transport(send_transport), video_engine_(video_engine) {
: send_transport(send_transport),
receive_lock_(RWLockWrapper::CreateRWLock()),
send_lock_(RWLockWrapper::CreateRWLock()),
video_engine_(video_engine) {
assert(video_engine != NULL);
assert(send_transport != NULL);
@ -66,19 +69,18 @@ VideoSendStream::Config VideoCall::GetDefaultSendConfig() {
}
newapi::VideoSendStream* VideoCall::CreateSendStream(
const newapi::VideoSendStream::Config& send_stream_config) {
assert(send_stream_config.rtp.ssrcs.size() > 0);
assert(send_stream_config.codec.numberOfSimulcastStreams == 0 ||
send_stream_config.codec.numberOfSimulcastStreams ==
send_stream_config.rtp.ssrcs.size());
const newapi::VideoSendStream::Config& config) {
assert(config.rtp.ssrcs.size() > 0);
assert(config.codec.numberOfSimulcastStreams == 0 ||
config.codec.numberOfSimulcastStreams == config.rtp.ssrcs.size());
VideoSendStream* send_stream =
new VideoSendStream(send_transport, video_engine_, send_stream_config);
for (size_t i = 0; i < send_stream_config.rtp.ssrcs.size(); ++i) {
uint32_t ssrc = send_stream_config.rtp.ssrcs[i];
// SSRC must be previously unused!
assert(send_ssrcs_[ssrc] == NULL &&
receive_ssrcs_.find(ssrc) == receive_ssrcs_.end());
send_ssrcs_[ssrc] = send_stream;
new VideoSendStream(send_transport, video_engine_, config);
WriteLockScoped write_lock(*send_lock_);
for (size_t i = 0; i < config.rtp.ssrcs.size(); ++i) {
assert(send_ssrcs_.find(config.rtp.ssrcs[i]) == send_ssrcs_.end());
send_ssrcs_[config.rtp.ssrcs[i]] = send_stream;
}
return send_stream;
}
@ -100,14 +102,13 @@ VideoReceiveStream::Config VideoCall::GetDefaultReceiveConfig() {
}
newapi::VideoReceiveStream* VideoCall::CreateReceiveStream(
const newapi::VideoReceiveStream::Config& receive_stream_config) {
assert(receive_ssrcs_[receive_stream_config.rtp.ssrc] == NULL);
const newapi::VideoReceiveStream::Config& config) {
VideoReceiveStream* receive_stream = new VideoReceiveStream(
video_engine_, receive_stream_config, send_transport);
receive_ssrcs_[receive_stream_config.rtp.ssrc] = receive_stream;
video_engine_, config, send_transport);
WriteLockScoped write_lock(*receive_lock_);
assert(receive_ssrcs_.find(config.rtp.ssrc) == receive_ssrcs_.end());
receive_ssrcs_[config.rtp.ssrc] = receive_stream;
return receive_stream;
}
@ -135,6 +136,7 @@ bool VideoCall::DeliverRtcp(ModuleRTPUtility::RTPHeaderParser* rtp_parser,
// TODO(pbos): Figure out what channel needs it actually.
// Do NOT broadcast! Also make sure it's a valid packet.
bool rtcp_delivered = false;
ReadLockScoped read_lock(*receive_lock_);
for (std::map<uint32_t, newapi::VideoReceiveStream*>::iterator it =
receive_ssrcs_.begin();
it != receive_ssrcs_.end(); ++it) {
@ -156,14 +158,14 @@ bool VideoCall::DeliverRtp(ModuleRTPUtility::RTPHeaderParser* rtp_parser,
return false;
}
uint32_t ssrc = rtp_header.ssrc;
if (receive_ssrcs_.find(ssrc) == receive_ssrcs_.end()) {
ReadLockScoped read_lock(*receive_lock_);
if (receive_ssrcs_.find(rtp_header.ssrc) == receive_ssrcs_.end()) {
// TODO(pbos): Log some warning, SSRC without receiver.
return false;
}
VideoReceiveStream* receiver =
static_cast<VideoReceiveStream*>(receive_ssrcs_[ssrc]);
static_cast<VideoReceiveStream*>(receive_ssrcs_[rtp_header.ssrc]);
return receiver->DeliverRtp(packet, length);
}

8
webrtc/video_engine/internal/video_call.h
View File

@ -15,6 +15,8 @@
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/video_engine/internal/video_receive_stream.h"
#include "webrtc/video_engine/internal/video_send_stream.h"
#include "webrtc/video_engine/new_include/video_engine.h"
@ -41,7 +43,7 @@ class VideoCall : public newapi::VideoCall, public newapi::PacketReceiver {
virtual newapi::VideoSendStream::Config GetDefaultSendConfig() OVERRIDE;
virtual newapi::VideoSendStream* CreateSendStream(
const newapi::VideoSendStream::Config& send_stream_config) OVERRIDE;
const newapi::VideoSendStream::Config& config) OVERRIDE;
virtual newapi::SendStreamState* DestroySendStream(
newapi::VideoSendStream* send_stream) OVERRIDE;
@ -49,7 +51,7 @@ class VideoCall : public newapi::VideoCall, public newapi::PacketReceiver {
virtual newapi::VideoReceiveStream::Config GetDefaultReceiveConfig() OVERRIDE;
virtual newapi::VideoReceiveStream* CreateReceiveStream(
const newapi::VideoReceiveStream::Config& receive_stream_config) OVERRIDE;
const newapi::VideoReceiveStream::Config& config) OVERRIDE;
virtual void DestroyReceiveStream(newapi::VideoReceiveStream* receive_stream)
OVERRIDE;
@ -68,7 +70,9 @@ class VideoCall : public newapi::VideoCall, public newapi::PacketReceiver {
newapi::Transport* send_transport;
std::map<uint32_t, newapi::VideoReceiveStream*> receive_ssrcs_;
scoped_ptr<RWLockWrapper> receive_lock_;
std::map<uint32_t, newapi::VideoSendStream*> send_ssrcs_;
scoped_ptr<RWLockWrapper> send_lock_;
webrtc::VideoEngine* video_engine_;
ViERTP_RTCP* rtp_rtcp_;