Incorporating logging into Cast

media/cast/audio_receiver/audio_receiver.cc

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "media/cast/audio_receiver/audio_receiver.h"
 
 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/message_loop/message_loop.h"
 #include "media/cast/audio_receiver/audio_decoder.h"
@@ skipping 89 matching lines @@
     audio_decoder_ = new AudioDecoder(audio_config);
   }
   rtp_receiver_.reset(new RtpReceiver(cast_environment->Clock(),
                                       &audio_config,
                                       NULL,
                                       incoming_payload_callback_.get()));
   rtp_audio_receiver_statistics_.reset(
       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,
                        rtp_audio_receiver_statistics_.get(),
                        audio_config.rtcp_mode,
                        rtcp_interval_delta,
                        false,
                        audio_config.feedback_ssrc,
                        audio_config.rtcp_c_name));
   rtcp_->SetRemoteSSRC(audio_config.incoming_ssrc);
   ScheduleNextRtcpReport();
   ScheduleNextCastMessage();
 }
 
 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;
     time_first_incoming_packet_ =  cast_environment_->Clock()->NowTicks();
   }
 
   if (audio_decoder_) {
     DCHECK(!audio_buffer_) << "Invalid internal state";
     audio_decoder_->IncomingParsedRtpPacket(payload_data, payload_size,
                                             rtp_header);
@@ skipping 36 matching lines @@
   uint32 rtp_timestamp = 0;
   if (!audio_decoder_->GetRawAudioFrame(number_of_10ms_blocks,
                                         desired_frequency,
                                         audio_frame.get(),
                                         &rtp_timestamp)) {
     return;
   }
   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,
       base::Bind(callback,
       base::Passed(&audio_frame), playout_time));
 }
 
 void AudioReceiver::PlayoutTimeout() {
   DCHECK(audio_buffer_) << "Invalid function call in this configuration";
   if (queued_encoded_callbacks_.empty()) {
@@ skipping 153 matching lines @@
 }
 
 void AudioReceiver::SendNextCastMessage() {
   DCHECK(audio_buffer_) << "Invalid function call in this configuration";
   audio_buffer_->SendCastMessage();  // Will only send a message if it is time.
   ScheduleNextCastMessage();
 }
 
 }  // namespace cast
 }  // namespace media