Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream.

webrtc/api/call/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_API_CALL_AUDIO_SEND_STREAM_H_
 #define WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_
 
 #include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/config.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 #include "webrtc/transport.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
+// TODO(minyue): This is copied from cricket::SendCodecSpec. Find better place
+// for it.
+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;
+};
+
 // WORK IN PROGRESS
 // This class is under development and is not yet intended for for use outside
 // of WebRtc/Libjingle. Please use the VoiceEngine API instead.
 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
 
 class AudioSendStream {
  public:
   struct Stats {
     // TODO(solenberg): Harmonize naming and defaults with receive stream stats.
     uint32_t local_ssrc = 0;
@@ skipping 52 matching lines @@
     // passed to Call::CreateAudioSendStream().
     // TODO(solenberg): Implement, once we configure codecs through the new API.
     // std::unique_ptr<AudioEncoder> encoder;
     int cng_payload_type = -1;  // pt, or -1 to disable Comfort Noise Generator.
 
     // Bitrate limits used for variable audio bitrate streams. Set both to -1 to
     // disable audio bitrate adaptation.
     // Note: This is still an experimental feature and not ready for real usage.
     int min_bitrate_kbps = -1;
     int max_bitrate_kbps = -1;
+
+    int max_send_bitrate_bps = 0;
+    SendCodecSpec send_codec_spec;
   };
 
   // Starts stream activity.
   // When a stream is active, it can receive, process and deliver packets.
   virtual void Start() = 0;
   // Stops stream activity.
   // When a stream is stopped, it can't receive, process or deliver packets.
   virtual void Stop() = 0;
 
   // TODO(solenberg): Make payload_type a config property instead.
   virtual bool SendTelephoneEvent(int payload_type, int event,
                                   int duration_ms) = 0;
 
   virtual void SetMuted(bool muted) = 0;
 
   virtual Stats GetStats() const = 0;
 
  protected:
   virtual ~AudioSendStream() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_