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Test multiple send/receive streams in Call.

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Removes renderer in VideoReceiveStream as it wasn't properly
deregistered before. Makes sure that send/receive streams are properly
wired so that receive streams receive the expected stream.

BUG=2423
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2326004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4891 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org 2013-10-01 11:33:24 +00:00
parent e546f02c84
commit b74b96f487
2 changed files with 88 additions and 1 deletions
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2
webrtc/video_engine/internal/video_receive_stream.cc
View File

@ -86,11 +86,11 @@ VideoReceiveStream::VideoReceiveStream(webrtc::VideoEngine* video_engine,
}
VideoReceiveStream::~VideoReceiveStream() {
render_->RemoveRenderer(channel_);
for (size_t i = 0; i < config_.external_decoders.size(); ++i) {
external_codec_->DeRegisterExternalReceiveCodec(
channel_, config_.external_decoders[i].payload_type);
}
network_->DeregisterSendTransport(channel_);
video_engine_base_->Release();

87
webrtc/video_engine/test/call_tests.cc
View File

@ -418,4 +418,91 @@ TEST_F(CallTest, SurvivesIncomingRtpPacketsToDestroyedReceiveStream) {
send_transport.StopSending();
receive_transport.StopSending();
}
// Test sets up a Call multiple senders with different resolutions and SSRCs.
// Another is set up to receive all three of these with different renderers.
// Each renderer verifies that it receives the expected resolution, and as soon
// as every renderer has received a frame, the test finishes.
TEST_F(CallTest, SendsAndReceivesMultipleStreams) {
static const size_t kNumStreams = 3;
class VideoOutputObserver : public VideoRenderer {
public:
VideoOutputObserver(int width, int height)
: width_(width), height_(height), done_(EventWrapper::Create()) {}
virtual void RenderFrame(const I420VideoFrame& video_frame,
int time_to_render_ms) OVERRIDE {
EXPECT_EQ(width_, video_frame.width());
EXPECT_EQ(height_, video_frame.height());
done_->Set();
}
void Wait() { done_->Wait(30 * 1000); }
private:
int width_;
int height_;
scoped_ptr<EventWrapper> done_;
};
struct {
uint32_t ssrc;
int width;
int height;
} codec_settings[kNumStreams] = {{1, 640, 480}, {2, 320, 240}, {3, 240, 160}};
test::DirectTransport sender_transport, receiver_transport;
scoped_ptr<Call> sender_call(Call::Create(Call::Config(&sender_transport)));
scoped_ptr<Call> receiver_call(
Call::Create(Call::Config(&receiver_transport)));
sender_transport.SetReceiver(receiver_call->Receiver());
receiver_transport.SetReceiver(sender_call->Receiver());
VideoSendStream* send_streams[kNumStreams];
VideoReceiveStream* receive_streams[kNumStreams];
VideoOutputObserver* observers[kNumStreams];
test::FrameGeneratorCapturer* frame_generators[kNumStreams];
for (size_t i = 0; i < kNumStreams; ++i) {
uint32_t ssrc = codec_settings[i].ssrc;
int width = codec_settings[i].width;
int height = codec_settings[i].height;
observers[i] = new VideoOutputObserver(width, height);
VideoReceiveStream::Config receive_config =
receiver_call->GetDefaultReceiveConfig();
receive_config.renderer = observers[i];
receive_config.rtp.ssrc = ssrc;
receive_streams[i] = receiver_call->CreateReceiveStream(receive_config);
receive_streams[i]->StartReceive();
VideoSendStream::Config send_config = sender_call->GetDefaultSendConfig();
send_config.rtp.ssrcs.push_back(ssrc);
send_config.codec.width = width;
send_config.codec.height = height;
send_streams[i] = sender_call->CreateSendStream(send_config);
send_streams[i]->StartSend();
frame_generators[i] = test::FrameGeneratorCapturer::Create(
send_streams[i]->Input(), width, height, 30, Clock::GetRealTimeClock());
frame_generators[i]->Start();
}
for (size_t i = 0; i < kNumStreams; ++i) {
observers[i]->Wait();
}
for (size_t i = 0; i < kNumStreams; ++i) {
frame_generators[i]->Stop();
delete frame_generators[i];
sender_call->DestroySendStream(send_streams[i]);
receiver_call->DestroyReceiveStream(receive_streams[i]);
delete observers[i];
}
sender_transport.StopSending();
receiver_transport.StopSending();
}
} // namespace webrtc