Cast:Transport: Dividing A/V Initialization pipeline

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"
 #include "net/base/big_endian.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,
-    const CastTransportConfig& config,
-    bool is_audio,
-    const scoped_refptr<base::TaskRunner>& transport_task_runner,
+    const scoped_refptr<base::SingleThreadTaskRunner>& transport_task_runner,
     PacedSender* const transport)
-    : config_(),
+    : clock_(clock),
       transport_(transport),
       stats_callback_(),
-      transport_task_runner_(transport_task_runner) {
-  // Store generic cast config and create packetizer config.
-  if (is_audio) {
-    storage_.reset(
-        new PacketStorage(clock, config.audio_rtp_config.history_ms));
-    config_.audio = true;
-    config_.ssrc = config.audio_ssrc;
-    config_.payload_type = config.audio_rtp_config.payload_type;
-    config_.frequency = config.audio_frequency;
-    config_.audio_codec = config.audio_codec;
-  } else {
-    storage_.reset(
-        new PacketStorage(clock, config.audio_rtp_config.history_ms));
-    config_.audio = false;
-    config_.ssrc = config.video_ssrc;
-    config_.payload_type = config.video_rtp_config.payload_type;
-    config_.frequency = kVideoFrequency;
-    config_.video_codec = config.video_codec;
-  }
-  // Randomly set start values.
+      transport_task_runner_(transport_task_runner),
+      weak_factory_(this) {
+  // Randomly set sequence number start value.
   config_.sequence_number = base::RandInt(0, 65535);
-  packetizer_.reset(
-      new RtpPacketizer(transport, storage_.get(), config_));
 }
 
 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_));
+}
+
+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_));
+}
+
 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,
     const base::TimeTicks& recorded_time) {
+  DCHECK(packetizer_);
   packetizer_->IncomingEncodedAudioFrame(audio_frame, recorded_time);
 }
 
 void RtpSender::ResendPackets(
     const MissingFramesAndPacketsMap& missing_frames_and_packets) {
+  DCHECK(storage_);
   // Iterate over all frames in the list.
   for (MissingFramesAndPacketsMap::const_iterator it =
            missing_frames_and_packets.begin();
        it != missing_frames_and_packets.end();
        ++it) {
     PacketList packets_to_resend;
     uint8 frame_id = it->first;
     const PacketIdSet& packets_set = it->second;
     bool success = false;
 
@@ skipping 46 matching lines @@
 
 void RtpSender::SubscribeRtpStatsCallback(
     const CastTransportRtpStatistics& callback) {
   stats_callback_ = callback;
   ScheduleNextStatsReport();
 }
 
 void RtpSender::ScheduleNextStatsReport() {
   transport_task_runner_->PostDelayedTask(
       FROM_HERE,
-      base::Bind(&RtpSender::RtpStatistics, base::AsWeakPtr(this)),
+      base::Bind(&RtpSender::RtpStatistics, weak_factory_.GetWeakPtr()),
       base::TimeDelta::FromMilliseconds(kStatsCallbackIntervalMs));
 }
 
 void RtpSender::RtpStatistics() {
   RtcpSenderInfo sender_info;
   base::TimeTicks time_sent;
   uint32 rtp_timestamp = 0;
   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