| Index: media/cast/audio_receiver/audio_receiver.cc
|
| diff --git a/media/cast/audio_receiver/audio_receiver.cc b/media/cast/audio_receiver/audio_receiver.cc
|
| index c3dc3937b9acf176364fcc8aa5dbe3a16461da3a..c38947be684e9f417eb256f11e39fdfd48ba66ba 100644
|
| --- a/media/cast/audio_receiver/audio_receiver.cc
|
| +++ b/media/cast/audio_receiver/audio_receiver.cc
|
| @@ -107,7 +107,7 @@ AudioReceiver::AudioReceiver(scoped_refptr<CastEnvironment> cast_environment,
|
| new LocalRtpReceiverStatistics(rtp_receiver_.get()));
|
| base::TimeDelta rtcp_interval_delta =
|
| base::TimeDelta::FromMilliseconds(audio_config.rtcp_interval);
|
| - rtcp_.reset(new Rtcp(cast_environment->Clock(),
|
| + rtcp_.reset(new Rtcp(cast_environment,
|
| NULL,
|
| packet_sender,
|
| NULL,
|
| @@ -127,6 +127,10 @@ AudioReceiver::~AudioReceiver() {}
|
| void AudioReceiver::IncomingParsedRtpPacket(const uint8* payload_data,
|
| size_t payload_size,
|
| const RtpCastHeader& rtp_header) {
|
| + cast_environment_->Logging()->InsertPacketEvent(kPacketReceived,
|
| + rtp_header.webrtc.header.timestamp, rtp_header.frame_id,
|
| + rtp_header.packet_id, rtp_header.max_packet_id, payload_size);
|
| +
|
| // TODO(pwestin): update this as video to refresh over time.
|
| if (time_first_incoming_packet_.is_null()) {
|
| first_incoming_rtp_timestamp_ = rtp_header.webrtc.header.timestamp;
|
| @@ -183,6 +187,10 @@ void AudioReceiver::DecodeAudioFrameThread(
|
| base::TimeTicks now = cast_environment_->Clock()->NowTicks();
|
| base::TimeTicks playout_time;
|
| playout_time = GetPlayoutTime(now, rtp_timestamp);
|
| + base::TimeDelta diff = playout_time - now;
|
| +
|
| + cast_environment_->Logging()->InsertFrameEvent(kAudioPlayoutDelay,
|
| + rtp_timestamp, diff.InMilliseconds());
|
|
|
| // Frame is ready - Send back to the main thread.
|
| cast_environment_->PostTask(CastEnvironment::MAIN, FROM_HERE,
|
|
|