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Side by Side Diff: webrtc/audio_sink.h

Issue 1737593002: Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Added a few more missing headers Created 4 years, 10 months ago
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1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
11 #ifndef WEBRTC_AUDIO_AUDIO_SINK_H_ 11 #ifndef WEBRTC_AUDIO_SINK_H_
12 #define WEBRTC_AUDIO_AUDIO_SINK_H_ 12 #define WEBRTC_AUDIO_SINK_H_
13 13
14 #if defined(WEBRTC_POSIX) && !defined(__STDC_FORMAT_MACROS) 14 #if defined(WEBRTC_POSIX) && !defined(__STDC_FORMAT_MACROS)
15 // Avoid conflict with format_macros.h. 15 // Avoid conflict with format_macros.h.
16 #define __STDC_FORMAT_MACROS 16 #define __STDC_FORMAT_MACROS
17 #endif 17 #endif
18 18
19 #include <inttypes.h> 19 #include <inttypes.h>
20 #include <stddef.h> 20 #include <stddef.h>
21 21
22 namespace webrtc { 22 namespace webrtc {
(...skipping 20 matching lines...) Expand all
43 int sample_rate; // Sample rate in Hz. 43 int sample_rate; // Sample rate in Hz.
44 size_t channels; // Number of channels in the audio data. 44 size_t channels; // Number of channels in the audio data.
45 uint32_t timestamp; // The RTP timestamp of the first sample. 45 uint32_t timestamp; // The RTP timestamp of the first sample.
46 }; 46 };
47 47
48 virtual void OnData(const Data& audio) = 0; 48 virtual void OnData(const Data& audio) = 0;
49 }; 49 };
50 50
51 } // namespace webrtc 51 } // namespace webrtc
52 52
53 #endif // WEBRTC_AUDIO_AUDIO_SINK_H_ 53 #endif // WEBRTC_AUDIO_SINK_H_
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