/* * 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_MEDIA_FILE_MEDIA_FILE_H_ #define WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_H_ #include "webrtc/common_types.h" #include "webrtc/modules/include/module.h" #include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/media_file/media_file_defines.h" #include "webrtc/typedefs.h" namespace webrtc { class MediaFile : public Module { public: // Factory method. Constructor disabled. id is the identifier for the // MediaFile instance. static MediaFile* CreateMediaFile(const int32_t id); static void DestroyMediaFile(MediaFile* module); // 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 int32_t PlayoutAudioData( int8_t* audioBuffer, size_t& 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 int32_t PlayoutStereoData( int8_t* audioBufferLeft, int8_t* audioBufferRight, size_t& 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 int32_t StartPlayingAudioFile( const char* fileName, const uint32_t notificationTimeMs = 0, const bool loop = false, const FileFormats format = kFileFormatPcm16kHzFile, const CodecInst* codecInst = NULL, const uint32_t startPointMs = 0, const uint32_t stopPointMs = 0) = 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 int32_t StartPlayingAudioStream( InStream& stream, const uint32_t notificationTimeMs = 0, const FileFormats format = kFileFormatPcm16kHzFile, const CodecInst* codecInst = NULL, const uint32_t startPointMs = 0, const uint32_t stopPointMs = 0) = 0; // Stop playing from file or stream. virtual int32_t StopPlaying() = 0; // Return true if playing. virtual bool IsPlaying() = 0; // Set durationMs to the number of ms that has been played from file. virtual int32_t PlayoutPositionMs( uint32_t& 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 int32_t IncomingAudioData( const int8_t* audioBuffer, const size_t 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 int32_t StartRecordingAudioFile( const char* fileName, const FileFormats format, const CodecInst& codecInst, const uint32_t notificationTimeMs = 0, const uint32_t maxSizeBytes = 0) = 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 int32_t StartRecordingAudioStream( OutStream& stream, const FileFormats format, const CodecInst& codecInst, const uint32_t notificationTimeMs = 0) = 0; // Stop recording to file or stream. virtual int32_t StopRecording() = 0; // Return true if recording. virtual bool IsRecording() = 0; // Set durationMs to the number of ms that has been recorded to file. virtual int32_t RecordDurationMs(uint32_t& 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 int32_t 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 int32_t FileDurationMs( const char* fileName, uint32_t& durationMs, const FileFormats format, const uint32_t freqInHz = 16000) = 0; // Update codecInst according to the current audio codec being used for // reading or writing. virtual int32_t codec_info(CodecInst& codecInst) const = 0; protected: MediaFile() {} virtual ~MediaFile() {} }; } // namespace webrtc #endif // WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_H_