Refactor: simplify interface of SenderRtcpSession and CastTransport.

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 {
 
+namespace {
+
+class AudioSenderRtcpClient : public RtpSenderRtcpClient {

[miu, 2016/04/15 23:14:39]

This is duplicate code (same as in VideoSender).  So, can you just write it once
for FrameSender?  See comment below...

[xjz, 2016/04/20 01:09:03]

Done.

+ public:
+  explicit AudioSenderRtcpClient(base::WeakPtr<AudioSender> audio_sender)
+      : audio_sender_(audio_sender) {}
+
+  void OnCastMessageReceived(const RtcpCastMessage& cast_message) override {
+    if (audio_sender_)
+      audio_sender_->OnReceivedCastFeedback(cast_message);
+  }
+
+  void OnRttReceived(base::TimeDelta round_trip_time) override {
+    if (audio_sender_)
+      audio_sender_->OnMeasuredRoundTripTime(round_trip_time);
+  }
+
+  void OnPliReceived() override {
+    if (audio_sender_)
+      audio_sender_->OnReceivedPli();
+  }
+
+ private:
+  base::WeakPtr<AudioSender> audio_sender_;
+
+  DISALLOW_COPY_AND_ASSIGN(AudioSenderRtcpClient);
+};
+
+}  // namespace
+
 AudioSender::AudioSender(scoped_refptr<CastEnvironment> cast_environment,
                          const AudioSenderConfig& audio_config,
                          const StatusChangeCallback& status_change_cb,
                          CastTransport* const transport_sender)
     : FrameSender(cast_environment,
                   true,
                   transport_sender,
                   audio_config.frequency,
                   audio_config.ssrc,
                   0,  // |max_frame_rate_| is set after encoder initialization.
@@ skipping 32 matching lines @@
       audio_config.frequency / audio_encoder_->GetSamplesPerFrame();
 
   media::cast::CastTransportRtpConfig transport_config;
   transport_config.ssrc = audio_config.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::Bind(&AudioSender::OnReceivedCastFeedback,
-                                   weak_factory_.GetWeakPtr()),
-      base::Bind(&AudioSender::OnMeasuredRoundTripTime,
-                 weak_factory_.GetWeakPtr()),
-      base::Bind(&AudioSender::OnReceivedPli, weak_factory_.GetWeakPtr()));
+      transport_config,
+      base::WrapUnique(new AudioSenderRtcpClient(weak_factory_.GetWeakPtr())));
 }
 
 AudioSender::~AudioSender() {}
 
-void AudioSender::InsertAudio(scoped_ptr<AudioBus> audio_bus,
+void AudioSender::InsertAudio(std::unique_ptr<AudioBus> audio_bus,
                               const base::TimeTicks& recorded_time) {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
 
   if (!audio_encoder_) {
     NOTREACHED();
     return;
   }
 
   const base::TimeDelta next_frame_duration =
       RtpTimeDelta::FromTicks(audio_bus->frames()).ToTimeDelta(rtp_timebase());
@@ skipping 24 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