[Cast] Cleanup: Remove TransportXXXXXSender, an unnecessary layer of indirection.

media/cast/audio_sender/audio_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/audio_sender/audio_sender.h"
 
 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/message_loop/message_loop.h"
 #include "media/cast/audio_sender/audio_encoder.h"
@@ skipping 17 matching lines @@
             transport_sender_,
             NULL,  // paced sender.
             NULL,
             audio_config.rtcp_mode,
             base::TimeDelta::FromMilliseconds(audio_config.rtcp_interval),
             audio_config.rtp_config.ssrc,
             audio_config.incoming_feedback_ssrc,
             audio_config.rtcp_c_name,
             true),
       num_aggressive_rtcp_reports_sent_(0),
-      cast_initialization_cb_(STATUS_AUDIO_UNINITIALIZED),
+      cast_initialization_status_(STATUS_AUDIO_UNINITIALIZED),
       weak_factory_(this) {
   rtcp_.SetCastReceiverEventHistorySize(kReceiverRtcpEventHistorySize);
   if (!audio_config.use_external_encoder) {
     audio_encoder_.reset(
         new AudioEncoder(cast_environment,
                          audio_config,
                          base::Bind(&AudioSender::SendEncodedAudioFrame,
                                     weak_factory_.GetWeakPtr())));
-    cast_initialization_cb_ = audio_encoder_->InitializationResult();
+    cast_initialization_status_ = audio_encoder_->InitializationResult();
+  } else {
+    NOTREACHED();  // No support for external audio encoding.
+    cast_initialization_status_ = STATUS_AUDIO_INITIALIZED;
   }
 
   media::cast::transport::CastTransportAudioConfig transport_config;
   transport_config.codec = audio_config.codec;
   transport_config.rtp.config = audio_config.rtp_config;
   transport_config.frequency = audio_config.frequency;
   transport_config.channels = audio_config.channels;
   transport_config.rtp.max_outstanding_frames =
       audio_config.rtp_config.max_delay_ms / 100 + 1;
   transport_sender_->InitializeAudio(transport_config);
 
   memset(frame_id_to_rtp_timestamp_, 0, sizeof(frame_id_to_rtp_timestamp_));
 }
 
 AudioSender::~AudioSender() {}
 
 void AudioSender::InsertAudio(scoped_ptr<AudioBus> audio_bus,
                               const base::TimeTicks& recorded_time) {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
+  if (cast_initialization_status_ != STATUS_AUDIO_INITIALIZED) {
+    NOTREACHED();
+    return;
+  }
   DCHECK(audio_encoder_.get()) << "Invalid internal state";
   audio_encoder_->InsertAudio(audio_bus.Pass(), recorded_time);
 }
 
 void AudioSender::SendEncodedAudioFrame(
     scoped_ptr<transport::EncodedFrame> audio_frame) {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
+
   DCHECK(!audio_frame->reference_time.is_null());
   rtp_timestamp_helper_.StoreLatestTime(audio_frame->reference_time,
                                         audio_frame->rtp_timestamp);
 
   // At the start of the session, it's important to send reports before each
   // frame so that the receiver can properly compute playout times.  The reason
   // more than one report is sent is because transmission is not guaranteed,
   // only best effort, so we send enough that one should almost certainly get
   // through.
   if (num_aggressive_rtcp_reports_sent_ < kNumAggressiveReportsSentAtStart) {
@@ skipping 75 matching lines @@
   uint32 acked_frame_id = static_cast<uint32>(cast_feedback.ack_frame_id_);
   VLOG(2) << "Received audio ACK: " << acked_frame_id;
   cast_environment_->Logging()->InsertFrameEvent(
         cast_environment_->Clock()->NowTicks(),
         FRAME_ACK_RECEIVED, AUDIO_EVENT,
         frame_id_to_rtp_timestamp_[acked_frame_id & 0xff], acked_frame_id);
 }
 
 }  // namespace cast
 }  // namespace media