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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 |
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22 | 22 |
23 class FileCallback; | 23 class FileCallback; |
24 | 24 |
25 class FilePlayer { | 25 class FilePlayer { |
26 public: | 26 public: |
27 // The largest decoded frame size in samples (60ms with 32kHz sample rate). | 27 // The largest decoded frame size in samples (60ms with 32kHz sample rate). |
28 enum { MAX_AUDIO_BUFFER_IN_SAMPLES = 60 * 32 }; | 28 enum { MAX_AUDIO_BUFFER_IN_SAMPLES = 60 * 32 }; |
29 enum { MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES * 2 }; | 29 enum { MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES * 2 }; |
30 | 30 |
31 // Note: will return NULL for unsupported formats. | 31 // Note: will return NULL for unsupported formats. |
| 32 static std::unique_ptr<FilePlayer> CreateFilePlayer( |
| 33 const uint32_t instanceID, |
| 34 const FileFormats fileFormat); |
| 35 |
| 36 // Deprecated. Call CreateFilePlayer instead. |
32 static std::unique_ptr<FilePlayer> NewFilePlayer( | 37 static std::unique_ptr<FilePlayer> NewFilePlayer( |
33 const uint32_t instanceID, | 38 const uint32_t instanceID, |
34 const FileFormats fileFormat); | 39 const FileFormats fileFormat) { |
| 40 return CreateFilePlayer(instanceID, fileFormat); |
| 41 } |
35 | 42 |
36 virtual ~FilePlayer() = default; | 43 virtual ~FilePlayer() = default; |
37 | 44 |
38 // Read 10 ms of audio at |frequencyInHz| to |outBuffer|. |lengthInSamples| | 45 // Read 10 ms of audio at |frequencyInHz| to |outBuffer|. |lengthInSamples| |
39 // will be set to the number of samples read (not the number of samples per | 46 // will be set to the number of samples read (not the number of samples per |
40 // channel). | 47 // channel). |
41 virtual int Get10msAudioFromFile(int16_t* outBuffer, | 48 virtual int Get10msAudioFromFile(int16_t* outBuffer, |
42 size_t* lengthInSamples, | 49 size_t* lengthInSamples, |
43 int frequencyInHz) = 0; | 50 int frequencyInHz) = 0; |
44 | 51 |
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72 // Set audioCodec to the currently used audio codec. | 79 // Set audioCodec to the currently used audio codec. |
73 virtual int32_t AudioCodec(CodecInst* audioCodec) const = 0; | 80 virtual int32_t AudioCodec(CodecInst* audioCodec) const = 0; |
74 | 81 |
75 virtual int32_t Frequency() const = 0; | 82 virtual int32_t Frequency() const = 0; |
76 | 83 |
77 // Note: scaleFactor is in the range [0.0 - 2.0] | 84 // Note: scaleFactor is in the range [0.0 - 2.0] |
78 virtual int32_t SetAudioScaling(float scaleFactor) = 0; | 85 virtual int32_t SetAudioScaling(float scaleFactor) = 0; |
79 }; | 86 }; |
80 } // namespace webrtc | 87 } // namespace webrtc |
81 #endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ | 88 #endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ |
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