7ffeab525c
This is a reland of https://codereview.webrtc.org/1737593002/ minus the added missing headers in webrtc/{BUILD.gn,webrtc.gyp} and webrtc/common.gyp that breaks GN in Chromium since it's using the --check flag (which we should support). BUG=webrtc:4243, webrtc:5589 TESTED=Tried generating GN files with --check in a Chromium checkout with this patch applied, successfully. TBR=pbos@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1740873003 . Cr-Commit-Position: refs/heads/master@{#11794}
54 lines
1.5 KiB
C++
54 lines
1.5 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_AUDIO_SINK_H_
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#define WEBRTC_AUDIO_SINK_H_
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#if defined(WEBRTC_POSIX) && !defined(__STDC_FORMAT_MACROS)
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// Avoid conflict with format_macros.h.
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#define __STDC_FORMAT_MACROS
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#endif
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#include <inttypes.h>
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#include <stddef.h>
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namespace webrtc {
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// Represents a simple push audio sink.
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class AudioSinkInterface {
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public:
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virtual ~AudioSinkInterface() {}
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struct Data {
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Data(int16_t* data,
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size_t samples_per_channel,
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int sample_rate,
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size_t channels,
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uint32_t timestamp)
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: data(data),
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samples_per_channel(samples_per_channel),
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sample_rate(sample_rate),
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channels(channels),
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timestamp(timestamp) {}
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int16_t* data; // The actual 16bit audio data.
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size_t samples_per_channel; // Number of frames in the buffer.
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int sample_rate; // Sample rate in Hz.
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size_t channels; // Number of channels in the audio data.
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uint32_t timestamp; // The RTP timestamp of the first sample.
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};
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virtual void OnData(const Data& audio) = 0;
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};
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} // namespace webrtc
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#endif // WEBRTC_AUDIO_SINK_H_
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