| /* |
| * Copyright (c) 2011 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_INTERFACE_FILE_PLAYER_H_ |
| #define WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_ |
| |
| #include "common_types.h" |
| #include "engine_configurations.h" |
| #include "module_common_types.h" |
| #include "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 video file formats (e.g. AVI) if the flag |
| // WEBRTC_MODULE_UTILITY_VIDEO is not defined. |
| static FilePlayer* CreateFilePlayer(const WebRtc_UWord32 instanceID, |
| const FileFormats fileFormat); |
| |
| static void DestroyFilePlayer(FilePlayer* player); |
| |
| virtual WebRtc_Word32 Get10msAudioFromFile( |
| WebRtc_Word16* decodedDataBuffer, |
| WebRtc_UWord32& decodedDataLengthInSamples, |
| const WebRtc_UWord32 frequencyInHz) = 0; |
| |
| // Register callback for receiving file playing notifications. |
| virtual WebRtc_Word32 RegisterModuleFileCallback( |
| FileCallback* callback) = 0; |
| |
| // API for playing audio from fileName to channel. |
| // Note: codecInst is used for pre-encoded files. |
| virtual WebRtc_Word32 StartPlayingFile( |
| const WebRtc_Word8* fileName, |
| bool loop, |
| WebRtc_UWord32 startPosition, |
| float volumeScaling, |
| WebRtc_UWord32 notification, |
| WebRtc_UWord32 stopPosition = 0, |
| const CodecInst* codecInst = NULL) = 0; |
| |
| // Note: codecInst is used for pre-encoded files. |
| virtual WebRtc_Word32 StartPlayingFile( |
| InStream& sourceStream, |
| WebRtc_UWord32 startPosition, |
| float volumeScaling, |
| WebRtc_UWord32 notification, |
| WebRtc_UWord32 stopPosition = 0, |
| const CodecInst* codecInst = NULL) = 0; |
| |
| virtual WebRtc_Word32 StopPlayingFile() = 0; |
| |
| virtual bool IsPlayingFile() const = 0; |
| |
| virtual WebRtc_Word32 GetPlayoutPosition(WebRtc_UWord32& durationMs) = 0; |
| |
| // Set audioCodec to the currently used audio codec. |
| virtual WebRtc_Word32 AudioCodec(CodecInst& audioCodec) const = 0; |
| |
| virtual WebRtc_Word32 Frequency() const = 0; |
| |
| // Note: scaleFactor is in the range [0.0 - 2.0] |
| virtual WebRtc_Word32 SetAudioScaling(float scaleFactor) = 0; |
| |
| // Return the time in ms until next video frame should be pulled (by |
| // calling GetVideoFromFile(..)). |
| // Note: this API reads one video frame from file. This means that it should |
| // be called exactly once per GetVideoFromFile(..) API call. |
| virtual WebRtc_Word32 TimeUntilNextVideoFrame() { return -1;} |
| |
| virtual WebRtc_Word32 StartPlayingVideoFile( |
| const WebRtc_Word8* /*fileName*/, |
| bool /*loop*/, |
| bool /*videoOnly*/) { return -1;} |
| |
| virtual WebRtc_Word32 video_codec_info(VideoCodec& /*videoCodec*/) const |
| {return -1;} |
| |
| virtual WebRtc_Word32 GetVideoFromFile(VideoFrame& /*videoFrame*/) |
| { return -1;} |
| |
| // Same as GetVideoFromFile(). videoFrame will have the resolution specified |
| // by the width outWidth and height outHeight in pixels. |
| virtual WebRtc_Word32 GetVideoFromFile(VideoFrame& /*videoFrame*/, |
| const WebRtc_UWord32 /*outWidth*/, |
| const WebRtc_UWord32 /*outHeight*/) |
| {return -1;} |
| protected: |
| virtual ~FilePlayer() {} |
| |
| }; |
| } // namespace webrtc |
| #endif // WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_ |