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Diffstat (limited to 'webrtc/modules/media_file/media_file.h')
| -rw-r--r-- | webrtc/modules/media_file/media_file.h | 180 |
1 files changed, 180 insertions, 0 deletions
diff --git a/webrtc/modules/media_file/media_file.h b/webrtc/modules/media_file/media_file.h new file mode 100644 index 0000000000..f6924d6bb0 --- /dev/null +++ b/webrtc/modules/media_file/media_file.h @@ -0,0 +1,180 @@ +/* + * 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_ |