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.
  */
 
 #include "webrtc/voice_engine/channel.h"
 
 #include <algorithm>
+#include <utility>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/format_macros.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/common.h"
 #include "webrtc/config.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
@@ skipping 529 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);
       }
     }
 
+    {
+      // Pass the audio buffers to an optional sink callback, before applying
+      // scaling/panning, as that applies to the mix operation.
+      // External recipients of the audio (e.g. via AudioTrack), will do their
+      // own mixing/dynamic processing.
+      CriticalSectionScoped cs(&_callbackCritSect);
+      if (audio_sink_) {
+        AudioSink::Data data(reinterpret_cast<uint8_t*>(&audioFrame->data_[0]),
+                             audioFrame->samples_per_channel_,
+                             audioFrame->sample_rate_hz_,
+                             audioFrame->num_channels_, audioFrame->timestamp_);
+        audio_sink_->OnData(data);
+      }
+    }
+
     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
+    // 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);
+          _outputExternalMediaCallbackPtr->Process(
+              _channelId, kPlaybackPerChannel,
+              reinterpret_cast<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 537 matching lines @@
 }
 
 int32_t
 Channel::UpdateLocalTimeStamp()
 {
 
     _timeStamp += static_cast<uint32_t>(_audioFrame.samples_per_channel_);
     return 0;
 }
 
+void Channel::SetSink(rtc::scoped_ptr<AudioSink> sink) {
+  CriticalSectionScoped cs(&_callbackCritSect);
+  audio_sink_ = std::move(sink);
+}
+
 int32_t
 Channel::StartPlayout()
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::StartPlayout()");
     if (channel_state_.Get().playing)
     {
         return 0;
     }
 
@@ skipping 2901 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