Using AudioOption to enable audio network adaptor.

webrtc/audio/audio_send_stream.cc

 /*
  *  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 308 matching lines @@
 
 // Apply current codec settings to a single voe::Channel used for sending.
 bool AudioSendStream::SetupSendCodec() {
   ScopedVoEInterface<VoEBase> base(voice_engine());
   ScopedVoEInterface<VoECodec> codec(voice_engine());
 
   const int channel = config_.voe_channel_id;
 
   // Disable VAD and FEC unless we know the other side wants them.
   codec->SetVADStatus(channel, false);
   codec->SetFECStatus(channel, false);

[michaelt, 2016/10/20 09:10:04]

shouldn't we move this code to the corresponding settings ?

if (codec->SetFECStatus(channel, send_codec_spec.enable_codec_fec) == -1) {
LOG_RTCERR2(SetFECStatus, channel, send_codec_spec.enable_codec_fec,
base->LastError());
}
	
	

[minyue-webrtc, 2016/10/20 09:22:11]

I agree. This part should be refactored. This is not the purpose of this CL
though.

 
   const auto& send_codec_spec = config_.send_codec_spec;
 
   // Set the codec immediately, since SetVADStatus() depends on whether
   // the current codec is mono or stereo.
   LOG(LS_INFO) << "Send channel " << channel << " selected voice codec "
                << ToString(send_codec_spec.codec_inst)
                << ", bitrate=" << send_codec_spec.codec_inst.rate;
 
   // If codec is already configured, we do not it again.
@@ skipping 40 matching lines @@
       LOG(LS_INFO) << "Attempt to set maximum playback rate to "
                    << send_codec_spec.opus_max_playback_rate
                    << " Hz on channel " << channel;
       if (codec->SetOpusMaxPlaybackRate(
               channel, send_codec_spec.opus_max_playback_rate) == -1) {
         LOG_RTCERR2(SetOpusMaxPlaybackRate, channel,
                     send_codec_spec.opus_max_playback_rate, base->LastError());
         return false;
       }
     }
+
+    if (config_.enable_audio_network_adaptor) {
+      // Audio network adaptor is only allowed for Opus currently.
+      // |SetReceiverFrameLengthRange| needs to be called before
+      // |EnableAudioNetworkAdaptor|.
+      channel_proxy_->SetReceiverFrameLengthRange(send_codec_spec.min_ptime_ms,
+                                                  send_codec_spec.max_ptime_ms);
+      if (!channel_proxy_->EnableAudioNetworkAdaptor(
+              config_.audio_network_adaptor_config)) {
+        LOG(LS_ERROR) << "Audio network adaptor cannot be enabled on SSRC "
+                      << config_.rtp.ssrc;
+        return false;
+      }
+      LOG(LS_INFO) << "Audio network adaptor enabled on SSRC "
+                   << config_.rtp.ssrc;
+    } else {
+      channel_proxy_->DisableAudioNetworkAdaptor();
+    }
   }
 
   // Set the CN payloadtype and the VAD status.
   if (send_codec_spec.cng_payload_type != -1) {
     // The CN payload type for 8000 Hz clockrate is fixed at 13.
     if (send_codec_spec.cng_plfreq != 8000) {
       webrtc::PayloadFrequencies cn_freq;
       switch (send_codec_spec.cng_plfreq) {
         case 16000:
           cn_freq = webrtc::kFreq16000Hz;
@@ skipping 33 matching lines @@
         LOG_RTCERR2(SetVADStatus, channel, true, base->LastError());
         return false;
       }
     }
   }
   return true;
 }
 
 }  // namespace internal
 }  // namespace webrtc