cast: Wire upp logging to be sent over RTCP between receiver and sender.

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 "crypto/encryptor.h"
@@ skipping 138 matching lines @@
   ScheduleNextRtcpReport();
   ScheduleNextCastMessage();
 }
 
 void AudioReceiver::IncomingParsedRtpPacket(const uint8* payload_data,
                                             size_t payload_size,
                                             const RtpCastHeader& rtp_header) {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
   base::TimeTicks now = cast_environment_->Clock()->NowTicks();
 
-  cast_environment_->Logging()->InsertPacketEvent(now, kPacketReceived,
+  cast_environment_->Logging()->InsertPacketEvent(now, kAudioPacketReceived,
       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.
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
   if (time_first_incoming_packet_.is_null()) {
     InitializeTimers();
     first_incoming_rtp_timestamp_ = rtp_header.webrtc.header.timestamp;
     time_first_incoming_packet_ = now;
   }
@@ skipping 213 matching lines @@
   bool rtcp_packet = Rtcp::IsRtcpPacket(packet, length);
   if (!rtcp_packet) {
     rtp_receiver_->ReceivedPacket(packet, length);
   } else {
     rtcp_->IncomingRtcpPacket(packet, length);
   }
   cast_environment_->PostTask(CastEnvironment::MAIN, FROM_HERE, callback);
 }
 
 void AudioReceiver::CastFeedback(const RtcpCastMessage& cast_message) {
-  // TODO(pwestin): add logging.
-  rtcp_->SendRtcpFromRtpReceiver(&cast_message, NULL);
+  RtcpReceiverLogMessage receiver_log;
+  AudioRtcpRawMap audio_logs =
+      cast_environment_->Logging()->GetAudioRtcpRawData();
+
+  while (!audio_logs.empty()) {
+    AudioRtcpRawMap::iterator it = audio_logs.begin();
+    uint32 rtp_timestamp = it->first;
+    std::pair<AudioRtcpRawMap::iterator, AudioRtcpRawMap::iterator>
+        frame_range = audio_logs.equal_range(rtp_timestamp);
+
+    RtcpReceiverFrameLogMessage frame_log(rtp_timestamp);
+
+    AudioRtcpRawMap::const_iterator event_it = frame_range.first;
+    for (; event_it != frame_range.second; ++event_it) {
+      RtcpReceiverEventLogMessage event_log_message;
+      event_log_message.type = event_it->second.type;
+      event_log_message.event_timestamp = event_it->second.timestamp;
+      event_log_message.delay_delta = event_it->second.delay_delta;
+      event_log_message.packet_id = event_it->second.packet_id;
+      frame_log.event_log_messages_.push_back(event_log_message);
+    }
+    receiver_log.push_back(frame_log);
+    audio_logs.erase(rtp_timestamp);
+  }
+
+  base::TimeTicks now = cast_environment_->Clock()->NowTicks();
+  cast_environment_->Logging()->InsertGenericEvent(now, kAudioAckSent,
+      cast_message.ack_frame_id_);
+
+  rtcp_->SendRtcpFromRtpReceiver(&cast_message, &receiver_log);
 }
 
 base::TimeTicks AudioReceiver::GetPlayoutTime(base::TimeTicks now,
                                               uint32 rtp_timestamp) {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
   // Senders time in ms when this frame was recorded.
   // Note: the senders clock and our local clock might not be synced.
   base::TimeTicks rtp_timestamp_in_ticks;
   base::TimeTicks playout_time;
   if (time_offset_ == base::TimeDelta()) {
@@ skipping 98 matching lines @@
   }
   if (audio_decoder_) {
     // Will only send a message if it is time.
     audio_decoder_->SendCastMessage();
   }
   ScheduleNextCastMessage();
 }
 
 }  // namespace cast
 }  // namespace media