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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@ -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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