| /* |
| * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ |
| #define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ |
| |
| #include "webrtc/common_types.h" |
| #include "webrtc/engine_configurations.h" |
| #include "webrtc/modules/include/module_common_types.h" |
| #include "webrtc/typedefs.h" |
| |
| namespace webrtc { |
| class FileCallback; |
| |
| class FilePlayer |
| { |
| public: |
| // The largest decoded frame size in samples (60ms with 32kHz sample rate). |
| enum {MAX_AUDIO_BUFFER_IN_SAMPLES = 60*32}; |
| enum {MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES*2}; |
| |
| // Note: will return NULL for unsupported formats. |
| static FilePlayer* CreateFilePlayer(const uint32_t instanceID, |
| const FileFormats fileFormat); |
| |
| static void DestroyFilePlayer(FilePlayer* player); |
| |
| // Read 10 ms of audio at |frequencyInHz| to |outBuffer|. |lengthInSamples| |
| // will be set to the number of samples read (not the number of samples per |
| // channel). |
| virtual int Get10msAudioFromFile( |
| int16_t* outBuffer, |
| size_t& lengthInSamples, |
| int frequencyInHz) = 0; |
| |
| // Register callback for receiving file playing notifications. |
| virtual int32_t RegisterModuleFileCallback( |
| FileCallback* callback) = 0; |
| |
| // API for playing audio from fileName to channel. |
| // Note: codecInst is used for pre-encoded files. |
| virtual int32_t StartPlayingFile( |
| const char* fileName, |
| bool loop, |
| uint32_t startPosition, |
| float volumeScaling, |
| uint32_t notification, |
| uint32_t stopPosition = 0, |
| const CodecInst* codecInst = NULL) = 0; |
| |
| // Note: codecInst is used for pre-encoded files. |
| virtual int32_t StartPlayingFile( |
| InStream& sourceStream, |
| uint32_t startPosition, |
| float volumeScaling, |
| uint32_t notification, |
| uint32_t stopPosition = 0, |
| const CodecInst* codecInst = NULL) = 0; |
| |
| virtual int32_t StopPlayingFile() = 0; |
| |
| virtual bool IsPlayingFile() const = 0; |
| |
| virtual int32_t GetPlayoutPosition(uint32_t& durationMs) = 0; |
| |
| // Set audioCodec to the currently used audio codec. |
| virtual int32_t AudioCodec(CodecInst& audioCodec) const = 0; |
| |
| virtual int32_t Frequency() const = 0; |
| |
| // Note: scaleFactor is in the range [0.0 - 2.0] |
| virtual int32_t SetAudioScaling(float scaleFactor) = 0; |
| |
| protected: |
| virtual ~FilePlayer() {} |
| |
| }; |
| } // namespace webrtc |
| #endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ |