Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream.

webrtc/media/engine/webrtcvoiceengine.h

 /*
  *  Copyright (c) 2004 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.
  */
 
@@ skipping 19 matching lines @@
 #include "webrtc/media/engine/webrtcvoe.h"
 #include "webrtc/pc/channel.h"
 
 namespace cricket {
 
 class AudioDeviceModule;
 class AudioSource;
 class VoEWrapper;
 class WebRtcVoiceMediaChannel;
 
-struct SendCodecSpec {
-  SendCodecSpec() {
-    webrtc::CodecInst empty_inst = {0};
-    codec_inst = empty_inst;
-    codec_inst.pltype = -1;
-  }
-  bool operator==(const SendCodecSpec& rhs) const;
-  bool operator!=(const SendCodecSpec& rhs) const;
-
-  bool nack_enabled = false;
-  bool transport_cc_enabled = false;
-  bool enable_codec_fec = false;
-  bool enable_opus_dtx = false;
-  int opus_max_playback_rate = 0;
-  int red_payload_type = -1;
-  int cng_payload_type = -1;
-  int cng_plfreq = -1;
-  webrtc::CodecInst codec_inst;
-};
-
 // WebRtcVoiceEngine is a class to be used with CompositeMediaEngine.
 // It uses the WebRtc VoiceEngine library for audio handling.
 class WebRtcVoiceEngine final : public webrtc::TraceCallback  {
   friend class WebRtcVoiceMediaChannel;
  public:
   // Exposed for the WVoE/MC unit test.
   static bool ToCodecInst(const AudioCodec& in, webrtc::CodecInst* out);
 
   WebRtcVoiceEngine(
       webrtc::AudioDeviceModule* adm,
@@ skipping 160 matching lines @@
     return VoiceMediaChannel::SendRtcp(&packet, rtc::PacketOptions());
   }
 
   int GetReceiveChannelId(uint32_t ssrc) const;
   int GetSendChannelId(uint32_t ssrc) const;
 
  private:
   bool SetOptions(const AudioOptions& options);
   bool SetRecvCodecs(const std::vector<AudioCodec>& codecs);
   bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
-  bool SetSendCodecs(int channel, const webrtc::RtpParameters& rtp_parameters);
-  bool SetSendCodec(int channel, const webrtc::CodecInst& send_codec);
   bool SetLocalSource(uint32_t ssrc, AudioSource* source);
   bool MuteStream(uint32_t ssrc, bool mute);
 
   WebRtcVoiceEngine* engine() { return engine_; }
   int GetLastEngineError() { return engine()->GetLastEngineError(); }
   int GetOutputLevel(int channel);
   void ChangePlayout(bool playout);
   int CreateVoEChannel();
   bool DeleteVoEChannel(int channel);
   bool IsDefaultRecvStream(uint32_t ssrc) {
     return default_recv_ssrc_ == static_cast<int64_t>(ssrc);
   }
   bool SetMaxSendBitrate(int bps);
-  bool SetChannelSendParameters(int channel,
-                                const webrtc::RtpParameters& parameters);
-  bool SetMaxSendBitrate(int channel, int bps);
-  bool HasSendCodec() const {
-    return send_codec_spec_.codec_inst.pltype != -1;
-  }
   bool ValidateRtpParameters(const webrtc::RtpParameters& parameters);
   void SetupRecording();
 
   rtc::ThreadChecker worker_thread_checker_;
 
   WebRtcVoiceEngine* const engine_ = nullptr;
   std::vector<AudioCodec> send_codecs_;
   std::vector<AudioCodec> recv_codecs_;
   int max_send_bitrate_bps_ = 0;

[minyue-webrtc, 2016/10/14 13:32:56]

add back because SetSendBitrate (on all streams) can happen before
AddSendStream, which is captured by one of the unittests.

   AudioOptions options_;
   rtc::Optional<int> dtmf_payload_type_;
   bool desired_playout_ = false;
   bool recv_transport_cc_enabled_ = false;
   bool recv_nack_enabled_ = false;
   bool playout_ = false;
   bool send_ = false;
   webrtc::Call* const call_ = nullptr;
 
   // SSRC of unsignalled receive stream, or -1 if there isn't one.
   int64_t default_recv_ssrc_ = -1;
   // Volume for unsignalled stream, which may be set before the stream exists.
   double default_recv_volume_ = 1.0;
   // Sink for unsignalled stream, which may be set before the stream exists.
   std::unique_ptr<webrtc::AudioSinkInterface> default_sink_;
   // Default SSRC to use for RTCP receiver reports in case of no signaled
   // send streams. See: https://code.google.com/p/webrtc/issues/detail?id=4740
   // and https://code.google.com/p/chromium/issues/detail?id=547661
   uint32_t receiver_reports_ssrc_ = 0xFA17FA17u;
 
   class WebRtcAudioSendStream;
   std::map<uint32_t, WebRtcAudioSendStream*> send_streams_;
   std::vector<webrtc::RtpExtension> send_rtp_extensions_;
 
   class WebRtcAudioReceiveStream;
   std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_;
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 
-  SendCodecSpec send_codec_spec_;
+  webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
 };
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_