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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 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_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ | 11 #ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ |
12 #define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ | 12 #define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ |
13 | 13 |
14 #include <memory> | 14 #include <memory> |
15 | 15 |
16 #include "webrtc/common_types.h" | 16 #include "webrtc/common_types.h" |
17 #include "webrtc/modules/include/module_common_types.h" | 17 #include "webrtc/modules/include/module_common_types.h" |
18 #include "webrtc/typedefs.h" | 18 #include "webrtc/typedefs.h" |
19 #include "webrtc/voice_engine_configurations.h" | |
20 | 19 |
21 namespace webrtc { | 20 namespace webrtc { |
22 | 21 |
23 class FileCallback; | 22 class FileCallback; |
24 | 23 |
25 class FilePlayer { | 24 class FilePlayer { |
26 public: | 25 public: |
27 // The largest decoded frame size in samples (60ms with 32kHz sample rate). | 26 // The largest decoded frame size in samples (60ms with 32kHz sample rate). |
28 enum { MAX_AUDIO_BUFFER_IN_SAMPLES = 60 * 32 }; | 27 enum { MAX_AUDIO_BUFFER_IN_SAMPLES = 60 * 32 }; |
29 enum { MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES * 2 }; | 28 enum { MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES * 2 }; |
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72 // Set audioCodec to the currently used audio codec. | 71 // Set audioCodec to the currently used audio codec. |
73 virtual int32_t AudioCodec(CodecInst* audioCodec) const = 0; | 72 virtual int32_t AudioCodec(CodecInst* audioCodec) const = 0; |
74 | 73 |
75 virtual int32_t Frequency() const = 0; | 74 virtual int32_t Frequency() const = 0; |
76 | 75 |
77 // Note: scaleFactor is in the range [0.0 - 2.0] | 76 // Note: scaleFactor is in the range [0.0 - 2.0] |
78 virtual int32_t SetAudioScaling(float scaleFactor) = 0; | 77 virtual int32_t SetAudioScaling(float scaleFactor) = 0; |
79 }; | 78 }; |
80 } // namespace webrtc | 79 } // namespace webrtc |
81 #endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ | 80 #endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ |
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