Support for remote audio into tracks

webrtc/voice_engine/channel.cc

 /*
  *  Copyright (c) 2012 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 542 matching lines @@
     ChannelState::State state = channel_state_.Get();
 
     if (state.rx_apm_is_enabled) {
       int err = rx_audioproc_->ProcessStream(audioFrame);
       if (err) {
         LOG(LS_ERROR) << "ProcessStream() error: " << err;
         assert(false);
       }
     }
 
+    // External media.
+    // Pass the audio buffers to the external media callback, if set, before
+    // applying scaling/panning, as that applies to the mix operation.

[the sun, 2015/12/10 12:36:32]

Does this break any of our tests? It might break 3rd party code using the
processing API and relying on scaling etc being applied.

If you pick option b) in the previous comment you'd keep the external media code
in its original place for now, and add a new hook for AudioRenderer::Sink in
this place.

+    // External recipients of the audio (e.g. via AudioTrack), will do their own
+    // mixing.
+    if (_outputExternalMedia)
+    {
+        CriticalSectionScoped cs(&_callbackCritSect);
+        const bool isStereo = (audioFrame->num_channels_ == 2);
+        if (_outputExternalMediaCallbackPtr)
+        {
+            // TODO(tommi, hlundin): There's no timestamp information that
+            // accompanies the audio data.  Fix pwetty pwease.
+            _outputExternalMediaCallbackPtr->Process(
+                _channelId,
+                kPlaybackPerChannel,
+                reinterpret_cast<int16_t*>(audioFrame->data_),
+                audioFrame->samples_per_channel_,
+                audioFrame->sample_rate_hz_,
+                isStereo);
+        }
+    }
+
     float output_gain = 1.0f;
     float left_pan =  1.0f;
     float right_pan =  1.0f;
     {
       CriticalSectionScoped cs(&volume_settings_critsect_);
       output_gain = _outputGain;
       left_pan = _panLeft;
       right_pan= _panRight;
     }
 
@@ skipping 21 matching lines @@
         // stage)
         AudioFrameOperations::Scale(left_pan, right_pan, *audioFrame);
     }
 
     // Mix decoded PCM output with file if file mixing is enabled
     if (state.output_file_playing)
     {
         MixAudioWithFile(*audioFrame, audioFrame->sample_rate_hz_);
     }
 
-    // External media
-    if (_outputExternalMedia)
-    {
-        CriticalSectionScoped cs(&_callbackCritSect);
-        const bool isStereo = (audioFrame->num_channels_ == 2);
-        if (_outputExternalMediaCallbackPtr)
-        {
-            _outputExternalMediaCallbackPtr->Process(
-                _channelId,
-                kPlaybackPerChannel,
-                (int16_t*)audioFrame->data_,
-                audioFrame->samples_per_channel_,
-                audioFrame->sample_rate_hz_,
-                isStereo);
-        }
-    }
-
     // Record playout if enabled
     {
         CriticalSectionScoped cs(&_fileCritSect);
 
         if (_outputFileRecording && _outputFileRecorderPtr)
         {
             _outputFileRecorderPtr->RecordAudioToFile(*audioFrame);
         }
     }
 
@@ skipping 3455 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt)
       != 0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc