Using AudioOption to enable audio network adaptor.

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.
  */
 
@@ skipping 93 matching lines @@
   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;
 
+  virtual bool EnableAudioNetworkAdaptor(const std::string& config_string) = 0;

[the sun, 2016/10/06 08:03:46]

One of the major ideas with the new API is to have as few state-mutating
functions as possible, and instead move all of these types of settings to the
Config struct, so that they are only ever initialized in the constructor. When
settings change (because e.g. an Offer-Answer exchange), streams are instead
*recreated*.

That being said, IIUC you're also adding the information to
AudioCodingModule::Config. Since there is an ACM per VoE Channel (and a VoE
Channel per send stream), does that mean you do not need any of these API calls
at all, but that it is enough to propagate the information that path?

[minyue-webrtc, 2016/10/06 08:16:53]

Actually, this is audio network adaptor is a codec feature. It belongs to
DTX/FEC etc. But differently, it is set by AudioOption instead of SDP. I see
that SetOption only touches call_config_, and therefore, I tried to reuse
call_config_ (call_config_.acm_config in particular), but this is not necessary.

What makes it cumbersome is that I need to call to
Channel::EnableAudioNetworkAdaptor in SetSendCodec, which makes it similar to
many VoECodec functions. As long as SetSendCodec is called outside Ctor, I
cannot find a good way to avoid adding these APIs.

+
+  virtual void DisableAudioNetworkAdaptor() = 0;
+
+  virtual void SetReceiverFrameLengthRange(int min_frame_length_ms,
+                                           int max_frame_length_ms) = 0;
+
  protected:
   virtual ~AudioSendStream() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_