Cast: Log sender side packet events.

media/cast/transport/rtp_sender/rtp_sender.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/transport/rtp_sender/rtp_sender.h"
 
 #include "base/logging.h"
 #include "base/rand_util.h"
 #include "media/cast/transport/cast_transport_defines.h"
 #include "media/cast/transport/pacing/paced_sender.h"
 
 namespace media {
 namespace cast {
 namespace transport {
 
 // Schedule the RTP statistics callback every 33mS. As this interval affects the
 // time offset of the render and playout times, we want it in the same ball park
 // as the frame rate.
 static const int kStatsCallbackIntervalMs = 33;
 
 RtpSender::RtpSender(
     base::TickClock* clock,
+    LoggingImpl* logging,
     const scoped_refptr<base::SingleThreadTaskRunner>& transport_task_runner,
     PacedSender* const transport)
     : clock_(clock),
+      logging_(logging),
       transport_(transport),
       stats_callback_(),
       transport_task_runner_(transport_task_runner),
       weak_factory_(this) {
   // Randomly set sequence number start value.
   config_.sequence_number = base::RandInt(0, 65535);
 }
 
 RtpSender::~RtpSender() {}
 
 void RtpSender::InitializeAudio(const CastTransportAudioConfig& config) {
   storage_.reset(new PacketStorage(clock_, config.base.rtp_config.history_ms));
   config_.audio = true;
   config_.ssrc = config.base.ssrc;
   config_.payload_type = config.base.rtp_config.payload_type;
   config_.frequency = config.frequency;
   config_.audio_codec = config.codec;
-  packetizer_.reset(new RtpPacketizer(transport_, storage_.get(), config_));
+  packetizer_.reset(
+      new RtpPacketizer(transport_, storage_.get(), config_, clock_, logging_));
 }
 
 void RtpSender::InitializeVideo(const CastTransportVideoConfig& config) {
   storage_.reset(new PacketStorage(clock_, config.base.rtp_config.history_ms));
   config_.audio = false;
   config_.ssrc = config.base.ssrc;
   config_.payload_type = config.base.rtp_config.payload_type;
   config_.frequency = kVideoFrequency;
   config_.video_codec = config.codec;
-  packetizer_.reset(new RtpPacketizer(transport_, storage_.get(), config_));
+  packetizer_.reset(
+      new RtpPacketizer(transport_, storage_.get(), config_, clock_, logging_));
 }
 
 void RtpSender::IncomingEncodedVideoFrame(const EncodedVideoFrame* video_frame,
                                           const base::TimeTicks& capture_time) {
   DCHECK(packetizer_);
   packetizer_->IncomingEncodedVideoFrame(video_frame, capture_time);
 }
 
 void RtpSender::IncomingEncodedAudioFrame(
     const EncodedAudioFrame* audio_frame,
@@ skipping 82 matching lines @@
   packetizer_->LastSentTimestamp(&time_sent, &rtp_timestamp);
   sender_info.send_packet_count = packetizer_->send_packets_count();
   sender_info.send_octet_count = packetizer_->send_octet_count();
   stats_callback_.Run(sender_info, time_sent, rtp_timestamp);
   ScheduleNextStatsReport();
 }
 
 }  // namespace transport
 }  //  namespace cast
 }  // namespace media