New method RtpReceiver::GetLatestTimestamps.
The two timestamps, rtp time and corresponding system time, are always used together, for audio/video sync. The new method reads both timestamps, without releasing a lock in between. Ensures that the caller gets values corresponding to the same packet. Bug: webrtc:7135 Change-Id: I25bdcbe9ad620016bfad39841b339c266efade14 Reviewed-on: https://webrtc-review.googlesource.com/4062 Commit-Queue: Niels Moller <nisse@webrtc.org> Commit-Queue: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20120}
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@ -268,10 +268,9 @@ rtc::Optional<Syncable::Info> AudioReceiveStream::GetInfo() const {
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RTC_DCHECK(rtp_rtcp);
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RTC_DCHECK(rtp_receiver);
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if (!rtp_receiver->Timestamp(&info.latest_received_capture_timestamp)) {
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return rtc::Optional<Syncable::Info>();
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}
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if (!rtp_receiver->LastReceivedTimeMs(&info.latest_receive_time_ms)) {
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if (!rtp_receiver->GetLatestTimestamps(
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&info.latest_received_capture_timestamp,
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&info.latest_receive_time_ms)) {
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return rtc::Optional<Syncable::Info>();
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}
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if (rtp_rtcp->RemoteNTP(&info.capture_time_ntp_secs,
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@ -77,12 +77,11 @@ class RtpReceiver {
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PayloadUnion payload_specific,
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bool in_order) = 0;
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// Gets the last received timestamp. Returns true if a packet has been
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// received, false otherwise.
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virtual bool Timestamp(uint32_t* timestamp) const = 0;
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// Gets the time in milliseconds when the last timestamp was received.
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// Returns true if a packet has been received, false otherwise.
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virtual bool LastReceivedTimeMs(int64_t* receive_time_ms) const = 0;
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// Gets the RTP timestamp and the corresponding monotonic system
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// time for the most recent in-order packet. Returns true on
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// success, false if no packet has been received.
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virtual bool GetLatestTimestamps(uint32_t* timestamp,
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int64_t* receive_time_ms) const = 0;
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// Returns the remote SSRC of the currently received RTP stream.
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virtual uint32_t SSRC() const = 0;
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@ -225,24 +225,16 @@ std::vector<RtpSource> RtpReceiverImpl::GetSources() const {
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return sources;
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}
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bool RtpReceiverImpl::Timestamp(uint32_t* timestamp) const {
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bool RtpReceiverImpl::GetLatestTimestamps(uint32_t* timestamp,
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int64_t* receive_time_ms) const {
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rtc::CritScope lock(&critical_section_rtp_receiver_);
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if (!HaveReceivedFrame())
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if (last_received_frame_time_ms_ < 0)
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return false;
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*timestamp = last_received_timestamp_;
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return true;
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}
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bool RtpReceiverImpl::LastReceivedTimeMs(int64_t* receive_time_ms) const {
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rtc::CritScope lock(&critical_section_rtp_receiver_);
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if (!HaveReceivedFrame())
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return false;
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*receive_time_ms = last_received_frame_time_ms_;
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return true;
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}
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bool RtpReceiverImpl::HaveReceivedFrame() const {
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return last_received_frame_time_ms_ >= 0;
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return true;
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}
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// Implementation note: must not hold critsect when called.
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@ -47,9 +47,8 @@ class RtpReceiverImpl : public RtpReceiver {
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PayloadUnion payload_specific,
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bool in_order) override;
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// Returns the last received timestamp.
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bool Timestamp(uint32_t* timestamp) const override;
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bool LastReceivedTimeMs(int64_t* receive_time_ms) const override;
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bool GetLatestTimestamps(uint32_t* timestamp,
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int64_t* receive_time_ms) const override;
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uint32_t SSRC() const override;
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@ -70,9 +69,6 @@ class RtpReceiverImpl : public RtpReceiver {
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}
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private:
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bool HaveReceivedFrame() const
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RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtp_receiver_);
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void CheckSSRCChanged(const RTPHeader& rtp_header);
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void CheckCSRC(const WebRtcRTPHeader& rtp_header);
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int32_t CheckPayloadChanged(const RTPHeader& rtp_header,
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@ -185,8 +185,11 @@ TEST_F(RtpRtcpAudioTest, Basic) {
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EXPECT_EQ(test_ssrc, rtp_receiver2_->SSRC());
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uint32_t timestamp;
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EXPECT_TRUE(rtp_receiver2_->Timestamp(×tamp));
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int64_t receive_time_ms;
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EXPECT_TRUE(
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rtp_receiver2_->GetLatestTimestamps(×tamp, &receive_time_ms));
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EXPECT_EQ(test_timestamp, timestamp);
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EXPECT_EQ(fake_clock.TimeInMilliseconds(), receive_time_ms);
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}
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TEST_F(RtpRtcpAudioTest, DTMF) {
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@ -264,23 +267,30 @@ TEST_F(RtpRtcpAudioTest, ComfortNoise) {
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uint32_t in_timestamp = 0;
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for (const auto& c : kCngCodecs) {
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uint32_t timestamp;
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int64_t receive_time_ms;
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EXPECT_TRUE(module1->SendOutgoingData(
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webrtc::kAudioFrameSpeech, kPcmuPayloadType, in_timestamp, -1,
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kTestPayload, 4, nullptr, nullptr, nullptr));
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EXPECT_EQ(test_ssrc, rtp_receiver2_->SSRC());
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EXPECT_TRUE(rtp_receiver2_->Timestamp(×tamp));
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EXPECT_TRUE(
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rtp_receiver2_->GetLatestTimestamps(×tamp, &receive_time_ms));
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EXPECT_EQ(test_timestamp + in_timestamp, timestamp);
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EXPECT_EQ(fake_clock.TimeInMilliseconds(), receive_time_ms);
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in_timestamp += 10;
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fake_clock.AdvanceTimeMilliseconds(20);
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EXPECT_TRUE(module1->SendOutgoingData(webrtc::kAudioFrameCN, c.payload_type,
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in_timestamp, -1, kTestPayload, 1,
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nullptr, nullptr, nullptr));
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EXPECT_EQ(test_ssrc, rtp_receiver2_->SSRC());
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EXPECT_TRUE(rtp_receiver2_->Timestamp(×tamp));
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EXPECT_TRUE(
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rtp_receiver2_->GetLatestTimestamps(×tamp, &receive_time_ms));
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EXPECT_EQ(test_timestamp + in_timestamp, timestamp);
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EXPECT_EQ(fake_clock.TimeInMilliseconds(), receive_time_ms);
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in_timestamp += 10;
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fake_clock.AdvanceTimeMilliseconds(20);
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}
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}
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@ -364,9 +364,9 @@ rtc::Optional<Syncable::Info> VideoReceiveStream::GetInfo() const {
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RtpReceiver* rtp_receiver = rtp_video_stream_receiver_.GetRtpReceiver();
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RTC_DCHECK(rtp_receiver);
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if (!rtp_receiver->Timestamp(&info.latest_received_capture_timestamp))
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return rtc::Optional<Syncable::Info>();
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if (!rtp_receiver->LastReceivedTimeMs(&info.latest_receive_time_ms))
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if (!rtp_receiver->GetLatestTimestamps(
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&info.latest_received_capture_timestamp,
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&info.latest_receive_time_ms))
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return rtc::Optional<Syncable::Info>();
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RtpRtcp* rtp_rtcp = rtp_video_stream_receiver_.rtp_rtcp();
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