Refactoring: CastTransport InitializeAudio/InitializeVideo.

media/cast/sender/audio_sender.cc

 // Copyright 2014 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/sender/audio_sender.h"
 
 #include <utility>
 
 #include "base/bind.h"
 #include "base/logging.h"
-#include "base/memory/ptr_util.h"
 #include "base/message_loop/message_loop.h"
 #include "media/cast/common/rtp_time.h"
 #include "media/cast/net/cast_transport_config.h"
 #include "media/cast/sender/audio_encoder.h"
 
 namespace media {
 namespace cast {
 
 AudioSender::AudioSender(scoped_refptr<CastEnvironment> cast_environment,
                          const FrameSenderConfig& audio_config,
                          const StatusChangeCallback& status_change_cb,
                          CastTransport* const transport_sender)
     : FrameSender(cast_environment,
-                  true,
                   transport_sender,
-                  audio_config.rtp_timebase,
-                  audio_config.sender_ssrc,
-                  0,  // |max_frame_rate_| is set after encoder initialization.
-                  audio_config.min_playout_delay,
-                  audio_config.max_playout_delay,
-                  audio_config.animated_playout_delay,
+                  audio_config,
                   NewFixedCongestionControl(audio_config.max_bitrate)),
       samples_in_encoder_(0),
       weak_factory_(this) {
   if (!audio_config.use_external_encoder) {
     audio_encoder_.reset(new AudioEncoder(
         cast_environment, audio_config.channels, audio_config.rtp_timebase,
         audio_config.max_bitrate, audio_config.codec,
         base::Bind(&AudioSender::OnEncodedAudioFrame,
                    weak_factory_.GetWeakPtr(), audio_config.max_bitrate)));
   }
 
   // AudioEncoder provides no operational status changes during normal use.
   // Post a task now with its initialization result status to allow the client
   // to start sending frames.
   cast_environment_->PostTask(
       CastEnvironment::MAIN,
       FROM_HERE,
       base::Bind(status_change_cb,
                  audio_encoder_ ? audio_encoder_->InitializationResult() :
                      STATUS_INVALID_CONFIGURATION));
 
   // The number of samples per encoded audio frame depends on the codec and its
   // initialization parameters. Now that we have an encoder, we can calculate
   // the maximum frame rate.
   max_frame_rate_ =
       audio_config.rtp_timebase / audio_encoder_->GetSamplesPerFrame();
-
-  media::cast::CastTransportRtpConfig transport_config;
-  transport_config.ssrc = audio_config.sender_ssrc;
-  transport_config.feedback_ssrc = audio_config.receiver_ssrc;
-  transport_config.rtp_payload_type = audio_config.rtp_payload_type;
-  transport_config.aes_key = audio_config.aes_key;
-  transport_config.aes_iv_mask = audio_config.aes_iv_mask;
-
-  transport_sender->InitializeAudio(
-      transport_config, base::WrapUnique(new FrameSender::RtcpClient(
-                            weak_factory_.GetWeakPtr())));
 }
 
 AudioSender::~AudioSender() {}
 
 void AudioSender::InsertAudio(std::unique_ptr<AudioBus> audio_bus,
                               const base::TimeTicks& recorded_time) {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
 
   if (!audio_encoder_) {
     NOTREACHED();
@@ skipping 29 matching lines @@
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
 
   samples_in_encoder_ -= audio_encoder_->GetSamplesPerFrame() + samples_skipped;
   DCHECK_GE(samples_in_encoder_, 0);
 
   SendEncodedFrame(encoder_bitrate, std::move(encoded_frame));
 }
 
 }  // namespace cast
 }  // namespace media