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Potential deadlock in VideoSendStreamTest::ProducesStats

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VideoSendStream::GetStats() should not be called by
RtpRtcpObserver::OnSendRtcp(), as at this stage that thread will still
hold internal send locks.

Use an event and signal the test thread to call GetStats() instead.

BUG=
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5648 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
sprang@webrtc.org 2014-03-06 10:03:36 +00:00
parent 998cb8fcd0
commit 60ad5fdadf
1 changed files with 34 additions and 7 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

41
webrtc/video/video_send_stream_tests.cc
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@ -1083,12 +1083,38 @@ TEST_F(VideoSendStreamTest, NoPaddingWhenVideoIsMuted) {
}
TEST_F(VideoSendStreamTest, ProducesStats) {
static std::string kCName = "PjQatC14dGfbVwGPUOA9IH7RlsFDbWl4AhXEiDsBizo=";
static const std::string kCName =
"PjQatC14dGfbVwGPUOA9IH7RlsFDbWl4AhXEiDsBizo=";
static const uint32_t kTimeoutMs = 30 * 1000;
class StatsObserver : public test::RtpRtcpObserver {
public:
StatsObserver() : RtpRtcpObserver(30 * 1000), stream_(NULL) {}
StatsObserver()
: RtpRtcpObserver(kTimeoutMs),
stream_(NULL),
event_(EventWrapper::Create()) {}
virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
event_->Set();
return SEND_PACKET;
}
bool WaitForFilledStats() {
Clock* clock = Clock::GetRealTimeClock();
int64_t now = clock->TimeInMilliseconds();
int64_t stop_time = now + kTimeoutMs;
while (now < stop_time) {
int64_t time_left = stop_time - now;
if (time_left > 0 && event_->Wait(time_left) == kEventSignaled &&
CheckStats()) {
return true;
}
now = clock->TimeInMilliseconds();
}
return false;
}
bool CheckStats() {
VideoSendStream::Stats stats = stream_->GetStats();
// Check that all applicable data sources have been used.
if (stats.input_frame_rate > 0 && stats.encode_frame_rate > 0 &&
@ -1101,14 +1127,13 @@ TEST_F(VideoSendStreamTest, ProducesStats) {
config_.rtp.ssrcs.begin(), config_.rtp.ssrcs.end(), ssrc));
// Check for data populated by various sources. RTCP excluded as this
// data is received from remote side. Tested in call tests instead.
StreamStats& entry = stats.substreams[ssrc];
const StreamStats& entry = stats.substreams[ssrc];
if (entry.key_frames > 0u && entry.bitrate_bps > 0 &&
entry.rtp_stats.packets > 0u) {
observation_complete_->Set();
return true;
}
}
return SEND_PACKET;
return false;
}
void SetConfig(const VideoSendStream::Config& config) { config_ = config; }
@ -1117,6 +1142,7 @@ TEST_F(VideoSendStreamTest, ProducesStats) {
VideoSendStream* stream_;
VideoSendStream::Config config_;
scoped_ptr<EventWrapper> event_;
} observer;
Call::Config call_config(observer.SendTransport());
@ -1134,7 +1160,8 @@ TEST_F(VideoSendStreamTest, ProducesStats) {
send_stream_->StartSending();
frame_generator_capturer->Start();
EXPECT_EQ(kEventSignaled, observer.Wait());
EXPECT_TRUE(observer.WaitForFilledStats())
<< "Timed out waiting for filled statistics.";
observer.StopSending();
frame_generator_capturer->Stop();