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Unified Diff: webrtc/audio/audio_receive_stream.h

Issue 1505253004: Support for remote audio into tracks (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Add thread check, Remove bits_per_sample and use int16_t. Created 5 years ago
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Index: webrtc/audio/audio_receive_stream.h
diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h
index 08e65cd0843507dd6575b923408ded6214066e24..48e674b2978d516cf7a8dcac2603d30264d7954f 100644
--- a/webrtc/audio/audio_receive_stream.h
+++ b/webrtc/audio/audio_receive_stream.h
@@ -11,6 +11,7 @@
#ifndef WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
#define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
+#include "webrtc/audio/audio_sink.h"
#include "webrtc/audio_receive_stream.h"
#include "webrtc/audio_state.h"
#include "webrtc/base/thread_checker.h"
@@ -24,6 +25,7 @@ class ChannelProxy;
} // namespace voe
namespace internal {
+
class AudioReceiveStream final : public webrtc::AudioReceiveStream {
public:
AudioReceiveStream(RemoteBitrateEstimator* remote_bitrate_estimator,
@@ -43,6 +45,8 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream {
// webrtc::AudioReceiveStream implementation.
webrtc::AudioReceiveStream::Stats GetStats() const override;
+ void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) override;
+
const webrtc::AudioReceiveStream::Config& config() const;
private:

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