Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream.

webrtc/audio/audio_send_stream.h

Index: webrtc/audio/audio_send_stream.h
diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h
index 2e0b7aed584620aaddcdf6f8c92b3926389b73b9..c0fc65c90cff60feb33b8a9e2c4897d33b479836 100644
--- a/webrtc/audio/audio_send_stream.h
+++ b/webrtc/audio/audio_send_stream.h
@@ -61,6 +61,12 @@ class AudioSendStream final : public webrtc::AudioSendStream,
  private:
   VoiceEngine* voice_engine() const;
 
+  bool SetSendCodecs();
+  bool SetSendCodec(const webrtc::CodecInst& send_codec);
+  bool ApplyMaxSendBitrate();
+  bool IsCodecMultiRate(const webrtc::CodecInst& codec) const;
+  int MaxBitrateBps(const webrtc::CodecInst& codec) const;
+
   rtc::ThreadChecker thread_checker_;
   rtc::TaskQueue* worker_queue_;
   const webrtc::AudioSendStream::Config config_;