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/*
* 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_