Support external audio mixer in webrtc 2.

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 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 259 matching lines @@
 
   // If OPUS, change what we send according to the "stereo" codec
   // parameter, and not the "channels" parameter.  We set
   // voe_codec.channels to 2 if "stereo=1" and 1 otherwise.  If
   // the bitrate is not specified, i.e. is <= zero, we set it to the
   // appropriate default value for mono or stereo Opus.
   voe_codec->channels = IsCodecFeatureEnabled(codec, kCodecParamStereo) ? 2 : 1;
   voe_codec->rate = GetOpusBitrate(codec, *max_playback_rate);
 }
 
-webrtc::AudioState::Config MakeAudioStateConfig(VoEWrapper* voe_wrapper) {
+webrtc::AudioState::Config MakeAudioStateConfig(
+    VoEWrapper* voe_wrapper,
+    rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer) {
   webrtc::AudioState::Config config;
   config.voice_engine = voe_wrapper->engine();
-  config.audio_mixer = webrtc::AudioMixerImpl::Create();
+  if (audio_mixer) {
+    config.audio_mixer = audio_mixer;
+  } else {
+    config.audio_mixer = webrtc::AudioMixerImpl::Create();
+  }
   return config;
 }
 
 class WebRtcVoiceCodecs final {
  public:
   // TODO(solenberg): Do this filtering once off-line, add a simple AudioCodec
   // list and add a test which verifies VoE supports the listed codecs.
   static std::vector<AudioCodec> SupportedSendCodecs() {
     std::vector<AudioCodec> result;
     // Iterate first over our preferred codecs list, so that the results are
@@ skipping 239 matching lines @@
 
 }  // namespace {
 
 bool WebRtcVoiceEngine::ToCodecInst(const AudioCodec& in,
                                     webrtc::CodecInst* out) {
   return WebRtcVoiceCodecs::ToCodecInst(in, out);
 }
 
 WebRtcVoiceEngine::WebRtcVoiceEngine(
     webrtc::AudioDeviceModule* adm,
-    const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory)
-    : WebRtcVoiceEngine(adm, decoder_factory, new VoEWrapper()) {
-  audio_state_ = webrtc::AudioState::Create(MakeAudioStateConfig(voe()));
+    const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
+    rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer)
+    : WebRtcVoiceEngine(adm, decoder_factory, audio_mixer, new VoEWrapper()) {
+  audio_state_ =
+      webrtc::AudioState::Create(MakeAudioStateConfig(voe(), audio_mixer));
 }
 
 WebRtcVoiceEngine::WebRtcVoiceEngine(
     webrtc::AudioDeviceModule* adm,
     const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
+    rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
     VoEWrapper* voe_wrapper)
     : adm_(adm), decoder_factory_(decoder_factory), voe_wrapper_(voe_wrapper) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "WebRtcVoiceEngine::WebRtcVoiceEngine";
   RTC_DCHECK(voe_wrapper);
   RTC_DCHECK(decoder_factory);
 
   signal_thread_checker_.DetachFromThread();
 
   // Load our audio codec list.
@@ skipping 2092 matching lines @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   const auto it = send_streams_.find(ssrc);
   if (it != send_streams_.end()) {
     return it->second->channel();
   }
   return -1;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE