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..4cf79a5822e4ec31f07ac47c98e118999d90b86b 100644
--- a/webrtc/audio/audio_send_stream.h
+++ b/webrtc/audio/audio_send_stream.h
@@ -15,6 +15,7 @@
 
 #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"
@@ -61,6 +62,11 @@ class AudioSendStream final : public webrtc::AudioSendStream,
  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_;