| /* |
| * 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_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_ |
| #define WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_ |
| |
| #include "common_types.h" |
| #include "typedefs.h" |
| #include "module.h" |
| #include "module_common_types.h" |
| #include "media_file_defines.h" |
| |
| namespace webrtc { |
| class MediaFile : public Module |
| { |
| public: |
| // Factory method. Constructor disabled. id is the identifier for the |
| // MediaFile instance. |
| static MediaFile* CreateMediaFile(const WebRtc_Word32 id); |
| static void DestroyMediaFile(MediaFile* module); |
| |
| // Set the MediaFile instance identifier. |
| virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id) = 0; |
| |
| // Put 10-60ms of audio data from file into the audioBuffer depending on |
| // codec frame size. dataLengthInBytes is both an input and output |
| // parameter. As input parameter it indicates the size of audioBuffer. |
| // As output parameter it indicates the number of bytes written to |
| // audioBuffer. |
| // Note: This API only play mono audio but can be used on file containing |
| // audio with more channels (in which case the audio will be converted to |
| // mono). |
| virtual WebRtc_Word32 PlayoutAudioData( |
| WebRtc_Word8* audioBuffer, |
| WebRtc_UWord32& dataLengthInBytes) = 0; |
| |
| // Put one video frame into videoBuffer. dataLengthInBytes is both an input |
| // and output parameter. As input parameter it indicates the size of |
| // videoBuffer. As output parameter it indicates the number of bytes written |
| // to videoBuffer. |
| virtual WebRtc_Word32 PlayoutAVIVideoData( |
| WebRtc_Word8* videoBuffer, |
| WebRtc_UWord32& dataLengthInBytes) = 0; |
| |
| // Put 10-60ms, depending on codec frame size, of audio data from file into |
| // audioBufferLeft and audioBufferRight. The buffers contain the left and |
| // right channel of played out stereo audio. |
| // dataLengthInBytes is both an input and output parameter. As input |
| // parameter it indicates the size of both audioBufferLeft and |
| // audioBufferRight. As output parameter it indicates the number of bytes |
| // written to both audio buffers. |
| // Note: This API can only be successfully called for WAV files with stereo |
| // audio. |
| virtual WebRtc_Word32 PlayoutStereoData( |
| WebRtc_Word8* audioBufferLeft, |
| WebRtc_Word8* audioBufferRight, |
| WebRtc_UWord32& dataLengthInBytes) = 0; |
| |
| // Open the file specified by fileName (relative path is allowed) for |
| // reading. FileCallback::PlayNotification(..) will be called after |
| // notificationTimeMs of the file has been played if notificationTimeMs is |
| // greater than zero. If loop is true the file will be played until |
| // StopPlaying() is called. When end of file is reached the file is read |
| // from the start. format specifies the type of file fileName refers to. |
| // codecInst specifies the encoding of the audio data. Note that |
| // file formats that contain this information (like WAV files) don't need to |
| // provide a non-NULL codecInst. startPointMs and stopPointMs, unless zero, |
| // specify what part of the file should be read. From startPointMs ms to |
| // stopPointMs ms. |
| // Note: codecInst.channels should be set to 2 for stereo (and 1 for |
| // mono). Stereo audio is only supported for WAV files. |
| virtual WebRtc_Word32 StartPlayingAudioFile( |
| const WebRtc_Word8* fileName, |
| const WebRtc_UWord32 notificationTimeMs = 0, |
| const bool loop = false, |
| const FileFormats format = kFileFormatPcm16kHzFile, |
| const CodecInst* codecInst = NULL, |
| const WebRtc_UWord32 startPointMs = 0, |
| const WebRtc_UWord32 stopPointMs = 0) = 0; |
| |
| // Open the file specified by fileName for reading (relative path is |
| // allowed). If loop is true the file will be played until StopPlaying() is |
| // called. When end of file is reached the file is read from the start. |
| // format specifies the type of file fileName refers to. Only video will be |
| // read if videoOnly is true. |
| virtual WebRtc_Word32 StartPlayingVideoFile(const WebRtc_Word8* fileName, |
| const bool loop, |
| bool videoOnly, |
| const FileFormats format) = 0; |
| |
| // Prepare for playing audio from stream. |
| // FileCallback::PlayNotification(..) will be called after |
| // notificationTimeMs of the file has been played if notificationTimeMs is |
| // greater than zero. format specifies the type of file fileName refers to. |
| // codecInst specifies the encoding of the audio data. Note that |
| // file formats that contain this information (like WAV files) don't need to |
| // provide a non-NULL codecInst. startPointMs and stopPointMs, unless zero, |
| // specify what part of the file should be read. From startPointMs ms to |
| // stopPointMs ms. |
| // Note: codecInst.channels should be set to 2 for stereo (and 1 for |
| // mono). Stereo audio is only supported for WAV files. |
| virtual WebRtc_Word32 StartPlayingAudioStream( |
| InStream& stream, |
| const WebRtc_UWord32 notificationTimeMs = 0, |
| const FileFormats format = kFileFormatPcm16kHzFile, |
| const CodecInst* codecInst = NULL, |
| const WebRtc_UWord32 startPointMs = 0, |
| const WebRtc_UWord32 stopPointMs = 0) = 0; |
| |
| // Stop playing from file or stream. |
| virtual WebRtc_Word32 StopPlaying() = 0; |
| |
| // Return true if playing. |
| virtual bool IsPlaying() = 0; |
| |
| |
| // Set durationMs to the number of ms that has been played from file. |
| virtual WebRtc_Word32 PlayoutPositionMs( |
| WebRtc_UWord32& durationMs) const = 0; |
| |
| // Write one audio frame, i.e. the bufferLength first bytes of audioBuffer, |
| // to file. The audio frame size is determined by the codecInst.pacsize |
| // parameter of the last sucessfull StartRecordingAudioFile(..) call. |
| // Note: bufferLength must be exactly one frame. |
| virtual WebRtc_Word32 IncomingAudioData( |
| const WebRtc_Word8* audioBuffer, |
| const WebRtc_UWord32 bufferLength) = 0; |
| |
| // Write one video frame, i.e. the bufferLength first bytes of videoBuffer, |
| // to file. |
| // Note: videoBuffer can contain encoded data. The codec used must be the |
| // same as what was specified by videoCodecInst for the last successfull |
| // StartRecordingVideoFile(..) call. The videoBuffer must contain exactly |
| // one video frame. |
| virtual WebRtc_Word32 IncomingAVIVideoData( |
| const WebRtc_Word8* videoBuffer, |
| const WebRtc_UWord32 bufferLength) = 0; |
| |
| // Open/creates file specified by fileName for writing (relative path is |
| // allowed). FileCallback::RecordNotification(..) will be called after |
| // notificationTimeMs of audio data has been recorded if |
| // notificationTimeMs is greater than zero. |
| // format specifies the type of file that should be created/opened. |
| // codecInst specifies the encoding of the audio data. maxSizeBytes |
| // specifies the number of bytes allowed to be written to file if it is |
| // greater than zero. |
| // Note: codecInst.channels should be set to 2 for stereo (and 1 for |
| // mono). Stereo is only supported for WAV files. |
| virtual WebRtc_Word32 StartRecordingAudioFile( |
| const WebRtc_Word8* fileName, |
| const FileFormats format, |
| const CodecInst& codecInst, |
| const WebRtc_UWord32 notificationTimeMs = 0, |
| const WebRtc_UWord32 maxSizeBytes = 0) = 0; |
| |
| // Open/create the file specified by fileName for writing audio/video data |
| // (relative path is allowed). format specifies the type of file fileName |
| // should be. codecInst specifies the encoding of the audio data. |
| // videoCodecInst specifies the encoding of the video data. Only video data |
| // will be recorded if videoOnly is true. |
| virtual WebRtc_Word32 StartRecordingVideoFile( |
| const WebRtc_Word8* fileName, |
| const FileFormats format, |
| const CodecInst& codecInst, |
| const VideoCodec& videoCodecInst, |
| bool videoOnly = false) = 0; |
| |
| // Prepare for recording audio to stream. |
| // FileCallback::RecordNotification(..) will be called after |
| // notificationTimeMs of audio data has been recorded if |
| // notificationTimeMs is greater than zero. |
| // format specifies the type of file that stream should correspond to. |
| // codecInst specifies the encoding of the audio data. |
| // Note: codecInst.channels should be set to 2 for stereo (and 1 for |
| // mono). Stereo is only supported for WAV files. |
| virtual WebRtc_Word32 StartRecordingAudioStream( |
| OutStream& stream, |
| const FileFormats format, |
| const CodecInst& codecInst, |
| const WebRtc_UWord32 notificationTimeMs = 0) = 0; |
| |
| // Stop recording to file or stream. |
| virtual WebRtc_Word32 StopRecording() = 0; |
| |
| // Return true if recording. |
| virtual bool IsRecording() = 0; |
| |
| // Set durationMs to the number of ms that has been recorded to file. |
| virtual WebRtc_Word32 RecordDurationMs(WebRtc_UWord32& durationMs) = 0; |
| |
| // Return true if recording or playing is stereo. |
| virtual bool IsStereo() = 0; |
| |
| // Register callback to receive media file related notifications. Disables |
| // callbacks if callback is NULL. |
| virtual WebRtc_Word32 SetModuleFileCallback(FileCallback* callback) = 0; |
| |
| // Set durationMs to the size of the file (in ms) specified by fileName. |
| // format specifies the type of file fileName refers to. freqInHz specifies |
| // the sampling frequency of the file. |
| virtual WebRtc_Word32 FileDurationMs( |
| const WebRtc_Word8* fileName, |
| WebRtc_UWord32& durationMs, |
| const FileFormats format, |
| const WebRtc_UWord32 freqInHz = 16000) = 0; |
| |
| // Update codecInst according to the current audio codec being used for |
| // reading or writing. |
| virtual WebRtc_Word32 codec_info(CodecInst& codecInst) const = 0; |
| |
| // Update videoCodecInst according to the current video codec being used for |
| // reading or writing. |
| virtual WebRtc_Word32 VideoCodecInst(VideoCodec& videoCodecInst) const = 0; |
| |
| protected: |
| MediaFile() {} |
| virtual ~MediaFile() {} |
| }; |
| } // namespace webrtc |
| #endif // WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_ |