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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.
*/
// Note: the class cannot be used for reading and writing at the same time.
#ifndef WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_UTILITY_H_
#define WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_UTILITY_H_
#include <stdio.h>
#include "webrtc/common_types.h"
#include "webrtc/modules/media_file/media_file_defines.h"
namespace webrtc {
class InStream;
class OutStream;
class ModuleFileUtility
{
public:
ModuleFileUtility();
~ModuleFileUtility();
// Prepare for playing audio from stream.
// startPointMs and stopPointMs, unless zero, specify what part of the file
// should be read. From startPointMs ms to stopPointMs ms.
int32_t InitWavReading(InStream& stream,
const uint32_t startPointMs = 0,
const uint32_t stopPointMs = 0);
// Put 10-60ms of audio data from stream into the audioBuffer depending on
// codec frame size. dataLengthInBytes indicates the size of audioBuffer.
// The return value is 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).
int32_t ReadWavDataAsMono(InStream& stream, int8_t* audioBuffer,
const size_t dataLengthInBytes);
// 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 indicates the size of both audioBufferLeft and
// audioBufferRight.
// The return value is the number of bytes read for each buffer.
// Note: This API can only be successfully called for WAV files with stereo
// audio.
int32_t ReadWavDataAsStereo(InStream& wav,
int8_t* audioBufferLeft,
int8_t* audioBufferRight,
const size_t bufferLength);
// Prepare for recording audio to stream.
// 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.
int32_t InitWavWriting(OutStream& stream, const CodecInst& codecInst);
// 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.
// The return value is the number of bytes written to audioBuffer.
int32_t WriteWavData(OutStream& stream,
const int8_t* audioBuffer,
const size_t bufferLength);
// Finalizes the WAV header so that it is correct if nothing more will be
// written to stream.
// Note: this API must be called before closing stream to ensure that the
// WAVE header is updated with the file size. Don't call this API
// if more samples are to be written to stream.
int32_t UpdateWavHeader(OutStream& stream);
// Prepare for playing audio from stream.
// startPointMs and stopPointMs, unless zero, specify what part of the file
// should be read. From startPointMs ms to stopPointMs ms.
// freqInHz is the PCM sampling frequency.
// NOTE, allowed frequencies are 8000, 16000 and 32000 (Hz)
int32_t InitPCMReading(InStream& stream,
const uint32_t startPointMs = 0,
const uint32_t stopPointMs = 0,
const uint32_t freqInHz = 16000);
// Put 10-60ms of audio data from stream into the audioBuffer depending on
// codec frame size. dataLengthInBytes indicates the size of audioBuffer.
// The return value is the number of bytes written to audioBuffer.
int32_t ReadPCMData(InStream& stream, int8_t* audioBuffer,
const size_t dataLengthInBytes);
// Prepare for recording audio to stream.
// freqInHz is the PCM sampling frequency.
// NOTE, allowed frequencies are 8000, 16000 and 32000 (Hz)
int32_t InitPCMWriting(OutStream& stream, const uint32_t freqInHz = 16000);
// Write one 10ms audio frame, i.e. the bufferLength first bytes of
// audioBuffer, to file. The audio frame size is determined by the freqInHz
// parameter of the last sucessfull InitPCMWriting(..) call.
// The return value is the number of bytes written to audioBuffer.
int32_t WritePCMData(OutStream& stream,
const int8_t* audioBuffer,
size_t bufferLength);
// Prepare for playing audio from stream.
// startPointMs and stopPointMs, unless zero, specify what part of the file
// should be read. From startPointMs ms to stopPointMs ms.
int32_t InitCompressedReading(InStream& stream,
const uint32_t startPointMs = 0,
const uint32_t stopPointMs = 0);
// Put 10-60ms of audio data from stream into the audioBuffer depending on
// codec frame size. dataLengthInBytes indicates the size of audioBuffer.
// The return value is the number of bytes written to audioBuffer.
int32_t ReadCompressedData(InStream& stream,
int8_t* audioBuffer,
const size_t dataLengthInBytes);
// Prepare for recording audio to stream.
// codecInst specifies the encoding of the audio data.
int32_t InitCompressedWriting(OutStream& stream,
const CodecInst& codecInst);
// 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 InitCompressedWriting(..) call.
// The return value is the number of bytes written to stream.
// Note: bufferLength must be exactly one frame.
int32_t WriteCompressedData(OutStream& stream,
const int8_t* audioBuffer,
const size_t bufferLength);
// Prepare for playing audio from stream.
// codecInst specifies the encoding of the audio data.
int32_t InitPreEncodedReading(InStream& stream,
const CodecInst& codecInst);
// Put 10-60ms of audio data from stream into the audioBuffer depending on
// codec frame size. dataLengthInBytes indicates the size of audioBuffer.
// The return value is the number of bytes written to audioBuffer.
int32_t ReadPreEncodedData(InStream& stream,
int8_t* audioBuffer,
const size_t dataLengthInBytes);
// Prepare for recording audio to stream.
// codecInst specifies the encoding of the audio data.
int32_t InitPreEncodedWriting(OutStream& stream,
const CodecInst& codecInst);
// Write one audio frame, i.e. the bufferLength first bytes of audioBuffer,
// to stream. The audio frame size is determined by the codecInst.pacsize
// parameter of the last sucessfull InitPreEncodedWriting(..) call.
// The return value is the number of bytes written to stream.
// Note: bufferLength must be exactly one frame.
int32_t WritePreEncodedData(OutStream& stream,
const int8_t* inData,
const size_t dataLengthInBytes);
// Set durationMs to the size of the file (in ms) specified by fileName.
// freqInHz specifies the sampling frequency of the file.
int32_t FileDurationMs(const char* fileName,
const FileFormats fileFormat,
const uint32_t freqInHz = 16000);
// Return the number of ms that have been played so far.
uint32_t PlayoutPositionMs();
// Update codecInst according to the current audio codec being used for
// reading or writing.
int32_t codec_info(CodecInst& codecInst);
private:
// Biggest WAV frame supported is 10 ms at 48kHz of 2 channel, 16 bit audio.
static const size_t WAV_MAX_BUFFER_SIZE = 480 * 2 * 2;
int32_t InitWavCodec(uint32_t samplesPerSec,
size_t channels,
uint32_t bitsPerSample,
uint32_t formatTag);
// Parse the WAV header in stream.
int32_t ReadWavHeader(InStream& stream);
// Update the WAV header. freqInHz, bytesPerSample, channels, format,
// lengthInBytes specify characterists of the audio data.
// freqInHz is the sampling frequency. bytesPerSample is the sample size in
// bytes. channels is the number of channels, e.g. 1 is mono and 2 is
// stereo. format is the encode format (e.g. PCMU, PCMA, PCM etc).
// lengthInBytes is the number of bytes the audio samples are using up.
int32_t WriteWavHeader(OutStream& stream,
uint32_t freqInHz,
size_t bytesPerSample,
size_t channels,
uint32_t format,
size_t lengthInBytes);
// Put dataLengthInBytes of audio data from stream into the audioBuffer.
// The return value is the number of bytes written to audioBuffer.
int32_t ReadWavData(InStream& stream, uint8_t* audioBuffer,
size_t dataLengthInBytes);
// Update the current audio codec being used for reading or writing
// according to codecInst.
int32_t set_codec_info(const CodecInst& codecInst);
struct WAVE_FMTINFO_header
{
int16_t formatTag;
int16_t nChannels;
int32_t nSamplesPerSec;
int32_t nAvgBytesPerSec;
int16_t nBlockAlign;
int16_t nBitsPerSample;
};
// Identifiers for preencoded files.
enum MediaFileUtility_CodecType
{
kCodecNoCodec = 0,
kCodecIsac,
kCodecIsacSwb,
kCodecIsacLc,
kCodecL16_8Khz,
kCodecL16_16kHz,
kCodecL16_32Khz,
kCodecPcmu,
kCodecPcma,
kCodecIlbc20Ms,
kCodecIlbc30Ms,
kCodecG722,
kCodecG722_1_32Kbps,
kCodecG722_1_24Kbps,
kCodecG722_1_16Kbps,
kCodecG722_1c_48,
kCodecG722_1c_32,
kCodecG722_1c_24,
kCodecAmr,
kCodecAmrWb,
kCodecG729,
kCodecG729_1,
kCodecG726_40,
kCodecG726_32,
kCodecG726_24,
kCodecG726_16
};
// TODO (hellner): why store multiple formats. Just store either codec_info_
// or _wavFormatObj and supply conversion functions.
WAVE_FMTINFO_header _wavFormatObj;
size_t _dataSize; // Chunk size if reading a WAV file
// Number of bytes to read. I.e. frame size in bytes. May be multiple
// chunks if reading WAV.
size_t _readSizeBytes;
uint32_t _stopPointInMs;
uint32_t _startPointInMs;
uint32_t _playoutPositionMs;
size_t _bytesWritten;
CodecInst codec_info_;
MediaFileUtility_CodecType _codecId;
// The amount of bytes, on average, used for one audio sample.
size_t _bytesPerSample;
size_t _readPos;
// Only reading or writing can be enabled, not both.
bool _reading;
bool _writing;
// Scratch buffer used for turning stereo audio to mono.
uint8_t _tempData[WAV_MAX_BUFFER_SIZE];
};
} // namespace webrtc
#endif // WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_UTILITY_H_