| Index: content/renderer/media/webrtc/webrtc_audio_sink_adapter.h
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| diff --git a/content/renderer/media/webrtc/webrtc_audio_sink_adapter.h b/content/renderer/media/webrtc/webrtc_audio_sink_adapter.h
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| new file mode 100644
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| index 0000000000000000000000000000000000000000..fc67c42c8f521d94c8c6546b2d610218a7e96795
|
| --- /dev/null
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| +++ b/content/renderer/media/webrtc/webrtc_audio_sink_adapter.h
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| @@ -0,0 +1,51 @@
|
| +// Copyright 2014 The Chromium Authors. All rights reserved.
|
| +// Use of this source code is governed by a BSD-style license that can be
|
| +// found in the LICENSE file.
|
| +
|
| +#ifndef CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_AUDIO_SINK_ADAPTER_H_
|
| +#define CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_AUDIO_SINK_ADAPTER_H_
|
| +
|
| +#include <stdint.h>
|
| +
|
| +#include <memory>
|
| +
|
| +#include "base/macros.h"
|
| +#include "content/public/renderer/media_stream_audio_sink.h"
|
| +#include "media/base/audio_parameters.h"
|
| +
|
| +namespace webrtc {
|
| +class AudioTrackSinkInterface;
|
| +} // namespace webrtc
|
| +
|
| +namespace content {
|
| +
|
| +// Adapter to the webrtc::AudioTrackSinkInterface of the audio track.
|
| +// This class is used in between the MediaStreamAudioSink and
|
| +// webrtc::AudioTrackSinkInterface. It gets data callback via the
|
| +// MediaStreamAudioSink::OnData() interface and pass the data to
|
| +// webrtc::AudioTrackSinkInterface.
|
| +class WebRtcAudioSinkAdapter : public MediaStreamAudioSink {
|
| + public:
|
| + explicit WebRtcAudioSinkAdapter(
|
| + webrtc::AudioTrackSinkInterface* sink);
|
| + ~WebRtcAudioSinkAdapter() override;
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| +
|
| + bool IsEqual(const webrtc::AudioTrackSinkInterface* other) const;
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| +
|
| + private:
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| + // MediaStreamAudioSink implementation.
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| + void OnData(const media::AudioBus& audio_bus,
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| + base::TimeTicks estimated_capture_time) override;
|
| + void OnSetFormat(const media::AudioParameters& params) override;
|
| +
|
| + webrtc::AudioTrackSinkInterface* const sink_;
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| +
|
| + media::AudioParameters params_;
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| + std::unique_ptr<int16_t[]> interleaved_data_;
|
| +
|
| + DISALLOW_COPY_AND_ASSIGN(WebRtcAudioSinkAdapter);
|
| +};
|
| +
|
| +} // namespace content
|
| +
|
| +#endif // CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_AUDIO_SINK_ADAPTER_H_
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|
|