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| 1 /* | |
| 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | |
| 3 * | |
| 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 | |
| 6 * tree. An additional intellectual property rights grant can be found | |
| 7 * in the file PATENTS. All contributing project authors may | |
| 8 * be found in the AUTHORS file in the root of the source tree. | |
| 9 */ | |
| 10 | |
| 11 #ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ | |
| 12 #define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ | |
| 13 | |
| 14 #include <memory> | |
| 15 | |
| 16 #include "webrtc/common_types.h" | |
| 17 #include "webrtc/engine_configurations.h" | |
| 18 #include "webrtc/modules/include/module_common_types.h" | |
| 19 #include "webrtc/typedefs.h" | |
| 20 | |
| 21 namespace webrtc { | |
| 22 | |
| 23 class FileCallback; | |
| 24 | |
| 25 class FilePlayer { | |
| 26 public: | |
| 27 // The largest decoded frame size in samples (60ms with 32kHz sample rate). | |
| 28 enum { MAX_AUDIO_BUFFER_IN_SAMPLES = 60 * 32 }; | |
| 29 enum { MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES * 2 }; | |
| 30 | |
| 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 virtual ~FilePlayer() = default; | |
| 37 | |
| 38 // 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 | |
| 40 // channel). | |
| 41 virtual int Get10msAudioFromFile(int16_t* outBuffer, | |
| 42 size_t* lengthInSamples, | |
| 43 int frequencyInHz) = 0; | |
| 44 | |
| 45 // Register callback for receiving file playing notifications. | |
| 46 virtual int32_t RegisterModuleFileCallback(FileCallback* callback) = 0; | |
| 47 | |
| 48 // API for playing audio from fileName to channel. | |
| 49 // Note: codecInst is used for pre-encoded files. | |
| 50 virtual int32_t StartPlayingFile(const char* fileName, | |
| 51 bool loop, | |
| 52 uint32_t startPosition, | |
| 53 float volumeScaling, | |
| 54 uint32_t notification, | |
| 55 uint32_t stopPosition, | |
| 56 const CodecInst* codecInst) = 0; | |
| 57 | |
| 58 // Note: codecInst is used for pre-encoded files. | |
| 59 virtual int32_t StartPlayingFile(InStream* sourceStream, | |
| 60 uint32_t startPosition, | |
| 61 float volumeScaling, | |
| 62 uint32_t notification, | |
| 63 uint32_t stopPosition, | |
| 64 const CodecInst* codecInst) = 0; | |
| 65 | |
| 66 virtual int32_t StopPlayingFile() = 0; | |
| 67 | |
| 68 virtual bool IsPlayingFile() const = 0; | |
| 69 | |
| 70 virtual int32_t GetPlayoutPosition(uint32_t* durationMs) = 0; | |
| 71 | |
| 72 // Set audioCodec to the currently used audio codec. | |
| 73 virtual int32_t AudioCodec(CodecInst* audioCodec) const = 0; | |
| 74 | |
| 75 virtual int32_t Frequency() const = 0; | |
| 76 | |
| 77 // Note: scaleFactor is in the range [0.0 - 2.0] | |
| 78 virtual int32_t SetAudioScaling(float scaleFactor) = 0; | |
| 79 }; | |
| 80 } // namespace webrtc | |
| 81 #endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ | |
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Chromium Code Reviews
Issue 2321473004:
Move coder, file_player, and file_recorder to webrtc/voice_engine (Closed)
