Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream.

webrtc/audio/audio_send_stream.h

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #ifndef WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
 #define WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
 
 #include <memory>
 
 #include "webrtc/api/call/audio_send_stream.h"
 #include "webrtc/api/call/audio_state.h"
+#include "webrtc/api/rtpparameters.h"

[the sun, 2016/10/12 15:15:17]

why?

[minyue-webrtc, 2016/10/12 18:59:21]

No need now. In patch set 1, some moved function kept their signatures and need
RtpParameters

 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/call/bitrate_allocator.h"
 
 namespace webrtc {
 class CongestionController;
 class VoiceEngine;
 class RtcEventLog;
 
 namespace voe {
@@ skipping 26 matching lines @@
   // Implements BitrateAllocatorObserver.
   uint32_t OnBitrateUpdated(uint32_t bitrate_bps,
                             uint8_t fraction_loss,
                             int64_t rtt) override;
 
   const webrtc::AudioSendStream::Config& config() const;
 
  private:
   VoiceEngine* voice_engine() const;
 
+  bool SetSendCodecs();
+  bool SetSendCodec(const webrtc::CodecInst& send_codec);
+  bool ApplyMaxSendBitrate();
+  bool HasSendCodec() const;
+
   rtc::ThreadChecker thread_checker_;
   rtc::TaskQueue* worker_queue_;
   const webrtc::AudioSendStream::Config config_;
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
   std::unique_ptr<voe::ChannelProxy> channel_proxy_;
 
   BitrateAllocator* const bitrate_allocator_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
 };
 }  // namespace internal
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_