webrtc/modules/media_file/interface/media_file.h

237 lines
11 KiB
C++

/*
* 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);
// Writes the version of the MediaFile to version. remainingBufferInBytes
// is both an input parameter and an output parameter. It indicates the size
// of version less messages in it. position is both an input parameter and
// an output parameter. It indicates the position of the NULL termination
// in the version string.
static WebRtc_Word32 GetVersion(WebRtc_Word8* version,
WebRtc_UWord32& remainingBufferInBytes,
WebRtc_UWord32& position);
// 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_