Implement 'abs-send-time' extension in VideoSendStream.
BUG=2229 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2184010 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4727 4adac7df-926f-26a2-2b94-8c16560cd09d
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@ -112,6 +112,9 @@ VideoSendStream::VideoSendStream(newapi::Transport* transport,
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if (extension == "toffset") {
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if (rtp_rtcp_->SetSendTimestampOffsetStatus(channel_, true, id) != 0)
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abort();
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} else if (extension == "abs-send-time") {
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if (rtp_rtcp_->SetSendAbsoluteSendTimeStatus(channel_, true, id) != 0)
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abort();
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} else {
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abort(); // Unsupported extension.
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}
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@ -138,6 +138,37 @@ TEST_F(VideoSendStreamTest, SupportsCName) {
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RunSendTest(call.get(), send_config, &observer);
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}
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TEST_F(VideoSendStreamTest, SupportsAbsoluteSendTime) {
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static const uint8_t kAbsSendTimeExtensionId = 13;
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class AbsoluteSendTimeObserver : public SendTransportObserver {
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public:
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AbsoluteSendTimeObserver() : SendTransportObserver(30 * 1000) {
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EXPECT_TRUE(rtp_header_parser_->RegisterRtpHeaderExtension(
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kRtpExtensionAbsoluteSendTime, kAbsSendTimeExtensionId));
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}
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virtual bool SendRTP(const uint8_t* packet, size_t length) OVERRIDE {
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RTPHeader header;
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EXPECT_TRUE(
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rtp_header_parser_->Parse(packet, static_cast<int>(length), &header));
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if (header.extension.absoluteSendTime > 0)
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send_test_complete_->Set();
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return true;
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}
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} observer;
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Call::Config call_config(&observer);
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scoped_ptr<Call> call(Call::Create(call_config));
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VideoSendStream::Config send_config = GetSendTestConfig(call.get());
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send_config.rtp.extensions.push_back(
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RtpExtension("abs-send-time", kAbsSendTimeExtensionId));
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RunSendTest(call.get(), send_config, &observer);
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}
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TEST_F(VideoSendStreamTest, SupportsTransmissionTimeOffset) {
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static const uint8_t kTOffsetExtensionId = 13;
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class DelayedEncoder : public test::FakeEncoder {
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