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 369 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_.audio_network_adaptor_config) {

[the sun, 2016/10/25 09:25:46]

Is the invariant that max/min_ptime_ms must be set when the config is specified?
DCHECK that here then.

What if max < min, etc? I can't make out what happens from the convoluted
structure below this level.

[minyue-webrtc, 2016/10/27 14:33:10]

They are parsed from SDP. This is checked down in Audio Network Adaptor which
DCHECKs that the initial sending frame length is covered in the range. It would
be good to check it in SDP parser too. But I found that pMinTime is not really
parsed there. see
https://cs.chromium.org/chromium/src/third_party/webrtc/api/webrtcsdp.cc?q=kC...

The consequence is that we will use default value and the maxptime, which is
parsed, needs to be meaningful. Therefore I added a DCHECK at the place where
the eventual values are determined.

BTW, maybe pmintime should be parsed (in a separate CL).

+      // 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