Made AudioReceiveStream a mixer participant.

webrtc/voice_engine/channel.h

Index: webrtc/voice_engine/channel.h
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index 4988e079d1203a2322347b0e32c297650d4c40ff..625cd617f3b80ce8a56c7406299a80d06d069352 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -22,6 +22,7 @@
 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
+#include "webrtc/modules/audio_mixer/audio_mixer_defines.h"
 #include "webrtc/modules/audio_processing/rms_level.h"
 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
@@ -374,6 +375,11 @@ class Channel
       AudioFrame* audioFrame) override;
   int32_t NeededFrequency(int32_t id) const override;
 
+  // From MixerAudioSource
+  MixerAudioSource::AudioFrameWithMuted GetAudioFrameWithMuted(
+      int32_t id,
+      int sample_rate_hz);
+
   // From FileCallback
   void PlayNotification(int32_t id, uint32_t durationMs) override;
   void RecordNotification(int32_t id, uint32_t durationMs) override;
@@ -466,7 +472,7 @@ class Channel
   std::unique_ptr<AudioSinkInterface> audio_sink_;
   AudioLevel _outputAudioLevel;
   bool _externalTransport;
-  AudioFrame _audioFrame;
+  AudioFrame _audioFrame, mix_audio_frame_;

[the sun, 2016/10/03 11:41:26]

One member per line, please.

[aleloi, 2016/10/03 12:57:28]

Done.

   // Downsamples to the codec rate if necessary.
   PushResampler<int16_t> input_resampler_;
   std::unique_ptr<FilePlayer> input_file_player_;