Fixing video loopback test with encoder factory.

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 370 matching lines @@
   VoiceEngine* voice_engine = audio_state->voice_engine();
   RTC_DCHECK(voice_engine);
   return voice_engine;
 }
 
 // Apply current codec settings to a single voe::Channel used for sending.
 bool AudioSendStream::SetupSendCodec(AudioSendStream* stream,
                                      const Config& new_config) {
   RTC_DCHECK(new_config.send_codec_spec);
   const auto& spec = *new_config.send_codec_spec;
+
+  RTC_DCHECK(new_config.encoder_factory);

[ossu, 2017/05/09 15:45:41]

Might as well move this one just below the check on line 389.

   std::unique_ptr<AudioEncoder> encoder =
       new_config.encoder_factory->MakeAudioEncoder(spec.payload_type,
                                                    spec.format);
 
   if (!encoder) {
     LOG(LS_ERROR) << "Unable to create encoder for " << spec.format;
     return false;
   }
   // If a bitrate has been specified for the codec, use it over the
   // codec's default.
@@ skipping 194 matching lines @@
     if (rtpRtcpModule->RegisterSendPayload(codec) != 0) {
       LOG(LS_ERROR) << "RegisterCngPayloadType() failed to register CN to "
                        "RTP/RTCP module";
     }
   }
 }
 
 
 }  // namespace internal
 }  // namespace webrtc