Add RTP timestamp to RTCRtpReceiver::RTCRtpContributingSource

RTP timestamp was recently added to RTCRtpContributingSource in
the WebRTC specification. This CL reflects that change.

Bug: chromium:965994
Change-Id: Iba76a2b90547e7ca70bb04dd556914c7f36ea3fb
Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/1625106
Commit-Queue: Johannes Kron <kron@chromium.org>
Reviewed-by: Henrik Boström <hbos@chromium.org>
Reviewed-by: Florent Castelli <orphis@chromium.org>
Cr-Commit-Position: refs/heads/master@{#663005}

content/renderer/media/webrtc/rtc_rtp_source.cc

@@ -49,4 +49,8 @@ base::Optional<double> RTCRtpSource::AudioLevel() const {
   return std::pow(10.0, -(double)rfc_level / 20.0);
 }
 
+uint32_t RTCRtpSource::RtpTimestamp() const {
+  return source_.rtp_timestamp();
+}
+
 }  // namespace content

content/renderer/media/webrtc/rtc_rtp_source.h

@@ -22,6 +22,7 @@ class CONTENT_EXPORT RTCRtpSource : public blink::WebRTCRtpSource {
   double TimestampMs() const override;
   uint32_t Source() const override;
   base::Optional<double> AudioLevel() const override;
+  uint32_t RtpTimestamp() const override;
 
  private:
   const webrtc::RtpSource source_;

third_party/blink/public/platform/web_rtc_rtp_source.h

@@ -26,6 +26,7 @@ class BLINK_PLATFORM_EXPORT WebRTCRtpSource {
   virtual double TimestampMs() const = 0;
   virtual uint32_t Source() const = 0;
   virtual base::Optional<double> AudioLevel() const = 0;
+  virtual uint32_t RtpTimestamp() const = 0;
 };
 
 }  // namespace blink

third_party/blink/renderer/modules/peerconnection/rtc_rtp_contributing_source.idl

@@ -7,4 +7,5 @@ dictionary RTCRtpContributingSource {
     required DOMHighResTimeStamp timestamp;
     required unsigned long       source;
              double              audioLevel;
+    required unsigned long       rtpTimestamp;
 };

third_party/blink/renderer/modules/peerconnection/rtc_rtp_receiver.cc

@@ -79,6 +79,7 @@ RTCRtpReceiver::getSynchronizationSources() {
     synchronization_source->setSource(web_source->Source());
     if (web_source->AudioLevel())
       synchronization_source->setAudioLevel(*web_source->AudioLevel());
+    synchronization_source->setRtpTimestamp(web_source->RtpTimestamp());
     synchronization_sources.push_back(synchronization_source);
   }
   return synchronization_sources;
@@ -97,6 +98,7 @@ RTCRtpReceiver::getContributingSources() {
     contributing_source->setSource(web_source->Source());
     if (web_source->AudioLevel())
       contributing_source->setAudioLevel(*web_source->AudioLevel());
+    contributing_source->setRtpTimestamp(web_source->RtpTimestamp());
     contributing_sources.push_back(contributing_source);
   }
   return contributing_sources;

third_party/blink/web_tests/external/wpt/webrtc/RTCRtpReceiver-getSynchronizationSources.https-expected.txt

 This is a testharness.js-based test.
 PASS [audio] getSynchronizationSources() eventually returns a non-empty list
 PASS [audio] RTCRtpSynchronizationSource.timestamp is a number
+PASS [audio] RTCRtpSynchronizationSource.rtpTimestamp is a number [0, 2^32-1]
 PASS [audio] getSynchronizationSources() does not contain SSRCs older than 10 seconds
 FAIL [audio] RTCRtpSynchronizationSource.timestamp is comparable to performance.timeOrigin + performance.now() assert_true: expected true got false
 PASS [audio] RTCRtpSynchronizationSource.source is a number
 PASS [video] getSynchronizationSources() eventually returns a non-empty list
 PASS [video] RTCRtpSynchronizationSource.timestamp is a number
+PASS [video] RTCRtpSynchronizationSource.rtpTimestamp is a number [0, 2^32-1]
 PASS [video] getSynchronizationSources() does not contain SSRCs older than 10 seconds
 FAIL [video] RTCRtpSynchronizationSource.timestamp is comparable to performance.timeOrigin + performance.now() assert_true: expected true got false
 PASS [video] RTCRtpSynchronizationSource.source is a number

third_party/blink/web_tests/external/wpt/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html

@@ -41,6 +41,15 @@ for (const kind of ['audio', 'video']) {
     assert_true(ssrc.timestamp >= startTime);
   }, '[' + kind + '] RTCRtpSynchronizationSource.timestamp is a number');
 
+  promise_test(async t => {
+    const receiver = await initiateSingleTrackCallAndReturnReceiver(t, kind);
+    const [ssrc] = await listenForSSRCs(t, receiver);
+    assert_equals(typeof ssrc.rtpTimestamp, 'number');
+    assert_greater_than_equal(ssrc.rtpTimestamp, 0);
+    assert_less_than_equal(ssrc.rtpTimestamp, 0xffffffff);
+  }, '[' + kind + '] RTCRtpSynchronizationSource.rtpTimestamp is a number ' +
+     '[0, 2^32-1]');
+
   promise_test(async t => {
     const receiver = await initiateSingleTrackCallAndReturnReceiver(t, kind);
     // Wait for packets to start flowing.