trunk/src/modules/media_file/source/media_file_utility.h - vendor/opensource/webrtc - Git at Google

 /*
  *  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.
  */

 // Note: the class cannot be used for reading and writing at the same time.
 #ifndef WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_UTILITY_H_
 #define WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_UTILITY_H_

 #include <stdio.h>

 #include "common_types.h"
 #include "media_file_defines.h"

 namespace webrtc {
 class AviFile;
 class InStream;
 class OutStream;

 class ModuleFileUtility
 {
 public:

     ModuleFileUtility(const WebRtc_Word32 id);
     ~ModuleFileUtility();

 #ifdef WEBRTC_MODULE_UTILITY_VIDEO
     // 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.
     // Only video will be read if videoOnly is true.
     WebRtc_Word32 InitAviReading(const WebRtc_Word8* fileName, bool videoOnly,
                                  bool loop);

     // Put 10-60ms of audio data from file into the outBuffer depending on
     // codec frame size. bufferLengthInBytes indicates the size of outBuffer.
     // 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 coverted to
     // mono).
     WebRtc_Word32 ReadAviAudioData(WebRtc_Word8* outBuffer,
                                    const WebRtc_UWord32 bufferLengthInBytes);

     // Put one video frame into outBuffer. bufferLengthInBytes indicates the
     // size of outBuffer.
     // The return value is the number of bytes written to videoBuffer.
     WebRtc_Word32 ReadAviVideoData(WebRtc_Word8* videoBuffer,
                                    const WebRtc_UWord32 bufferLengthInBytes);

     // Open/create the file specified by fileName for writing audio/video data
     // (relative path is allowed). 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.
     WebRtc_Word32 InitAviWriting(const WebRtc_Word8* filename,
                                  const CodecInst& codecInst,
                                  const VideoCodec& videoCodecInst,
                                  const bool videoOnly);

     // Write one audio frame, i.e. the bufferLengthinBytes first bytes of
     // audioBuffer, to file. The audio frame size is determined by the
     // codecInst.pacsize parameter of the last sucessfull
     // InitAviWriting(..) call.
     // Note: bufferLength must be exactly one frame.
     WebRtc_Word32 WriteAviAudioData(const WebRtc_Word8* audioBuffer,
                                     WebRtc_UWord32 bufferLengthInBytes);


     // 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
     // InitAviWriting(..) call. The videoBuffer must contain exactly
     // one video frame.
     WebRtc_Word32 WriteAviVideoData(const WebRtc_Word8* videoBuffer,
                                     WebRtc_UWord32 bufferLengthInBytes);

     // Stop recording to file or stream.
     WebRtc_Word32 CloseAviFile();

     WebRtc_Word32 VideoCodecInst(VideoCodec& codecInst);
 #endif // #ifdef WEBRTC_MODULE_UTILITY_VIDEO

     // 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.
     WebRtc_Word32 InitWavReading(InStream& stream,
                                  const WebRtc_UWord32 startPointMs = 0,
                                  const WebRtc_UWord32 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).
     WebRtc_Word32 ReadWavDataAsMono(InStream& stream, WebRtc_Word8* audioBuffer,
                                     const WebRtc_UWord32 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.
     WebRtc_Word32 ReadWavDataAsStereo(InStream& wav,
                                       WebRtc_Word8* audioBufferLeft,
                                       WebRtc_Word8* audioBufferRight,
                                       const WebRtc_UWord32 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.
     WebRtc_Word32 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.
     WebRtc_Word32 WriteWavData(OutStream& stream,
                                const WebRtc_Word8* audioBuffer,
                                const WebRtc_UWord32 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.
     WebRtc_Word32 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)
     WebRtc_Word32 InitPCMReading(InStream& stream,
                                  const WebRtc_UWord32 startPointMs = 0,
                                  const WebRtc_UWord32 stopPointMs = 0,
                                  const WebRtc_UWord32 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.
     WebRtc_Word32 ReadPCMData(InStream& stream, WebRtc_Word8* audioBuffer,
                               const WebRtc_UWord32 dataLengthInBytes);

     // Prepare for recording audio to stream.
     // freqInHz is the PCM sampling frequency.
     // NOTE, allowed frequencies are 8000, 16000 and 32000 (Hz)
     WebRtc_Word32 InitPCMWriting(OutStream& stream,
                                  const WebRtc_UWord32 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.
     WebRtc_Word32 WritePCMData(OutStream& stream,
                                const WebRtc_Word8* audioBuffer,
                                WebRtc_UWord32 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.
     WebRtc_Word32 InitCompressedReading(InStream& stream,
                                         const WebRtc_UWord32 startPointMs = 0,
                                         const WebRtc_UWord32 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.
     WebRtc_Word32 ReadCompressedData(InStream& stream,
                                      WebRtc_Word8* audioBuffer,
                                      const WebRtc_UWord32 dataLengthInBytes);

     // Prepare for recording audio to stream.
     // codecInst specifies the encoding of the audio data.
     WebRtc_Word32 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.
     WebRtc_Word32 WriteCompressedData(OutStream& stream,
                                       const WebRtc_Word8* audioBuffer,
                                       const WebRtc_UWord32 bufferLength);

     // Prepare for playing audio from stream.
     // codecInst specifies the encoding of the audio data.
     WebRtc_Word32 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.
     WebRtc_Word32 ReadPreEncodedData(InStream& stream,
                                      WebRtc_Word8* audioBuffer,
                                      const WebRtc_UWord32 dataLengthInBytes);

     // Prepare for recording audio to stream.
     // codecInst specifies the encoding of the audio data.
     WebRtc_Word32 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.
     WebRtc_Word32 WritePreEncodedData(OutStream& stream,
                                       const WebRtc_Word8* inData,
                                       const WebRtc_UWord32 dataLengthInBytes);

     // Set durationMs to the size of the file (in ms) specified by fileName.
     // freqInHz specifies the sampling frequency of the file.
     WebRtc_Word32 FileDurationMs(const WebRtc_Word8* fileName,
                                  const FileFormats fileFormat,
                                  const WebRtc_UWord32 freqInHz = 16000);

     // Return the number of ms that have been played so far.
     WebRtc_UWord32 PlayoutPositionMs();

     // Update codecInst according to the current audio codec being used for
     // reading or writing.
     WebRtc_Word32 codec_info(CodecInst& codecInst);

 private:
     // Biggest WAV frame supported is 10 ms at 48kHz of 2 channel, 16 bit audio.
     enum{WAV_MAX_BUFFER_SIZE = 480*2*2};


     WebRtc_Word32 InitWavCodec(WebRtc_UWord32 samplesPerSec,
                                WebRtc_UWord32 channels,
                                WebRtc_UWord32 bitsPerSample,
                                WebRtc_UWord32 formatTag);

     // Parse the WAV header in stream.
     WebRtc_Word32 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.
     WebRtc_Word32 WriteWavHeader(OutStream& stream,
                                  const WebRtc_UWord32 freqInHz,
                                  const WebRtc_UWord32 bytesPerSample,
                                  const WebRtc_UWord32 channels,
                                  const WebRtc_UWord32 format,
                                  const WebRtc_UWord32 lengthInBytes);

     // Put dataLengthInBytes of audio data from stream into the audioBuffer.
     // The return value is the number of bytes written to audioBuffer.
     WebRtc_Word32 ReadWavData(InStream& stream, WebRtc_UWord8* audioBuffer,
                               const WebRtc_UWord32 dataLengthInBytes);

     // Update the current audio codec being used for reading or writing
     // according to codecInst.
     WebRtc_Word32 set_codec_info(const CodecInst& codecInst);

     struct WAVE_FMTINFO_header
     {
         WebRtc_Word16 formatTag;
         WebRtc_Word16 nChannels;
         WebRtc_Word32 nSamplesPerSec;
         WebRtc_Word32 nAvgBytesPerSec;
         WebRtc_Word16 nBlockAlign;
         WebRtc_Word16 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,
         kCodecSpeex8Khz,
         kCodecSpeex16Khz
     };

     // TODO (hellner): why store multiple formats. Just store either codec_info_
     //                 or _wavFormatObj and supply conversion functions.
     WAVE_FMTINFO_header _wavFormatObj;
     WebRtc_Word32 _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.
     WebRtc_Word32 _readSizeBytes;

     WebRtc_Word32 _id;

     WebRtc_UWord32 _stopPointInMs;
     WebRtc_UWord32 _startPointInMs;
     WebRtc_UWord32 _playoutPositionMs;
     WebRtc_UWord32 _bytesWritten;

     CodecInst codec_info_;
     MediaFileUtility_CodecType _codecId;

     // The amount of bytes, on average, used for one audio sample.
     WebRtc_Word32  _bytesPerSample;
     WebRtc_Word32  _readPos;

     // Only reading or writing can be enabled, not both.
     bool _reading;
     bool _writing;

     // Scratch buffer used for turning stereo audio to mono.
     WebRtc_UWord8 _tempData[WAV_MAX_BUFFER_SIZE];

 #ifdef WEBRTC_MODULE_UTILITY_VIDEO
     AviFile* _aviAudioInFile;
     AviFile* _aviVideoInFile;
     AviFile* _aviOutFile;
     VideoCodec _videoCodec;
 #endif
 };
 } // namespace webrtc
 #endif // WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_UTILITY_H_
	/*
	* 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.
	*/

	// Note: the class cannot be used for reading and writing at the same time.
	#ifndef WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_UTILITY_H_
	#define WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_UTILITY_H_

	#include <stdio.h>

	#include "common_types.h"
	#include "media_file_defines.h"

	namespace webrtc {
	class AviFile;
	class InStream;
	class OutStream;

	class ModuleFileUtility
	{
	public:

	ModuleFileUtility(const WebRtc_Word32 id);
	~ModuleFileUtility();

	#ifdef WEBRTC_MODULE_UTILITY_VIDEO
	// 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.
	// Only video will be read if videoOnly is true.
	WebRtc_Word32 InitAviReading(const WebRtc_Word8* fileName, bool videoOnly,
	bool loop);

	// Put 10-60ms of audio data from file into the outBuffer depending on
	// codec frame size. bufferLengthInBytes indicates the size of outBuffer.
	// 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 coverted to
	// mono).
	WebRtc_Word32 ReadAviAudioData(WebRtc_Word8* outBuffer,
	const WebRtc_UWord32 bufferLengthInBytes);

	// Put one video frame into outBuffer. bufferLengthInBytes indicates the
	// size of outBuffer.
	// The return value is the number of bytes written to videoBuffer.
	WebRtc_Word32 ReadAviVideoData(WebRtc_Word8* videoBuffer,
	const WebRtc_UWord32 bufferLengthInBytes);

	// Open/create the file specified by fileName for writing audio/video data
	// (relative path is allowed). 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.
	WebRtc_Word32 InitAviWriting(const WebRtc_Word8* filename,
	const CodecInst& codecInst,
	const VideoCodec& videoCodecInst,
	const bool videoOnly);

	// Write one audio frame, i.e. the bufferLengthinBytes first bytes of
	// audioBuffer, to file. The audio frame size is determined by the
	// codecInst.pacsize parameter of the last sucessfull
	// InitAviWriting(..) call.
	// Note: bufferLength must be exactly one frame.
	WebRtc_Word32 WriteAviAudioData(const WebRtc_Word8* audioBuffer,
	WebRtc_UWord32 bufferLengthInBytes);


	// 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
	// InitAviWriting(..) call. The videoBuffer must contain exactly
	// one video frame.
	WebRtc_Word32 WriteAviVideoData(const WebRtc_Word8* videoBuffer,
	WebRtc_UWord32 bufferLengthInBytes);

	// Stop recording to file or stream.
	WebRtc_Word32 CloseAviFile();

	WebRtc_Word32 VideoCodecInst(VideoCodec& codecInst);
	#endif // #ifdef WEBRTC_MODULE_UTILITY_VIDEO

	// 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.
	WebRtc_Word32 InitWavReading(InStream& stream,
	const WebRtc_UWord32 startPointMs = 0,
	const WebRtc_UWord32 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).
	WebRtc_Word32 ReadWavDataAsMono(InStream& stream, WebRtc_Word8* audioBuffer,
	const WebRtc_UWord32 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.
	WebRtc_Word32 ReadWavDataAsStereo(InStream& wav,
	WebRtc_Word8* audioBufferLeft,
	WebRtc_Word8* audioBufferRight,
	const WebRtc_UWord32 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.
	WebRtc_Word32 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.
	WebRtc_Word32 WriteWavData(OutStream& stream,
	const WebRtc_Word8* audioBuffer,
	const WebRtc_UWord32 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.
	WebRtc_Word32 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)
	WebRtc_Word32 InitPCMReading(InStream& stream,
	const WebRtc_UWord32 startPointMs = 0,
	const WebRtc_UWord32 stopPointMs = 0,
	const WebRtc_UWord32 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.
	WebRtc_Word32 ReadPCMData(InStream& stream, WebRtc_Word8* audioBuffer,
	const WebRtc_UWord32 dataLengthInBytes);

	// Prepare for recording audio to stream.
	// freqInHz is the PCM sampling frequency.
	// NOTE, allowed frequencies are 8000, 16000 and 32000 (Hz)
	WebRtc_Word32 InitPCMWriting(OutStream& stream,
	const WebRtc_UWord32 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.
	WebRtc_Word32 WritePCMData(OutStream& stream,
	const WebRtc_Word8* audioBuffer,
	WebRtc_UWord32 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.
	WebRtc_Word32 InitCompressedReading(InStream& stream,
	const WebRtc_UWord32 startPointMs = 0,
	const WebRtc_UWord32 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.
	WebRtc_Word32 ReadCompressedData(InStream& stream,
	WebRtc_Word8* audioBuffer,
	const WebRtc_UWord32 dataLengthInBytes);

	// Prepare for recording audio to stream.
	// codecInst specifies the encoding of the audio data.
	WebRtc_Word32 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.
	WebRtc_Word32 WriteCompressedData(OutStream& stream,
	const WebRtc_Word8* audioBuffer,
	const WebRtc_UWord32 bufferLength);

	// Prepare for playing audio from stream.
	// codecInst specifies the encoding of the audio data.
	WebRtc_Word32 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.
	WebRtc_Word32 ReadPreEncodedData(InStream& stream,
	WebRtc_Word8* audioBuffer,
	const WebRtc_UWord32 dataLengthInBytes);

	// Prepare for recording audio to stream.
	// codecInst specifies the encoding of the audio data.
	WebRtc_Word32 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.
	WebRtc_Word32 WritePreEncodedData(OutStream& stream,
	const WebRtc_Word8* inData,
	const WebRtc_UWord32 dataLengthInBytes);

	// Set durationMs to the size of the file (in ms) specified by fileName.
	// freqInHz specifies the sampling frequency of the file.
	WebRtc_Word32 FileDurationMs(const WebRtc_Word8* fileName,
	const FileFormats fileFormat,
	const WebRtc_UWord32 freqInHz = 16000);

	// Return the number of ms that have been played so far.
	WebRtc_UWord32 PlayoutPositionMs();

	// Update codecInst according to the current audio codec being used for
	// reading or writing.
	WebRtc_Word32 codec_info(CodecInst& codecInst);

	private:
	// Biggest WAV frame supported is 10 ms at 48kHz of 2 channel, 16 bit audio.
	enum{WAV_MAX_BUFFER_SIZE = 48022};


	WebRtc_Word32 InitWavCodec(WebRtc_UWord32 samplesPerSec,
	WebRtc_UWord32 channels,
	WebRtc_UWord32 bitsPerSample,
	WebRtc_UWord32 formatTag);

	// Parse the WAV header in stream.
	WebRtc_Word32 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.
	WebRtc_Word32 WriteWavHeader(OutStream& stream,
	const WebRtc_UWord32 freqInHz,
	const WebRtc_UWord32 bytesPerSample,
	const WebRtc_UWord32 channels,
	const WebRtc_UWord32 format,
	const WebRtc_UWord32 lengthInBytes);

	// Put dataLengthInBytes of audio data from stream into the audioBuffer.
	// The return value is the number of bytes written to audioBuffer.
	WebRtc_Word32 ReadWavData(InStream& stream, WebRtc_UWord8* audioBuffer,
	const WebRtc_UWord32 dataLengthInBytes);

	// Update the current audio codec being used for reading or writing
	// according to codecInst.
	WebRtc_Word32 set_codec_info(const CodecInst& codecInst);

	struct WAVE_FMTINFO_header
	{
	WebRtc_Word16 formatTag;
	WebRtc_Word16 nChannels;
	WebRtc_Word32 nSamplesPerSec;
	WebRtc_Word32 nAvgBytesPerSec;
	WebRtc_Word16 nBlockAlign;
	WebRtc_Word16 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,
	kCodecSpeex8Khz,
	kCodecSpeex16Khz
	};

	// TODO (hellner): why store multiple formats. Just store either codec_info_
	// or _wavFormatObj and supply conversion functions.
	WAVE_FMTINFO_header _wavFormatObj;
	WebRtc_Word32 _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.
	WebRtc_Word32 _readSizeBytes;

	WebRtc_Word32 _id;

	WebRtc_UWord32 _stopPointInMs;
	WebRtc_UWord32 _startPointInMs;
	WebRtc_UWord32 _playoutPositionMs;
	WebRtc_UWord32 _bytesWritten;

	CodecInst codec_info_;
	MediaFileUtility_CodecType _codecId;

	// The amount of bytes, on average, used for one audio sample.
	WebRtc_Word32 _bytesPerSample;
	WebRtc_Word32 _readPos;

	// Only reading or writing can be enabled, not both.
	bool _reading;
	bool _writing;

	// Scratch buffer used for turning stereo audio to mono.
	WebRtc_UWord8 _tempData[WAV_MAX_BUFFER_SIZE];

	#ifdef WEBRTC_MODULE_UTILITY_VIDEO
	AviFile* _aviAudioInFile;
	AviFile* _aviVideoInFile;
	AviFile* _aviOutFile;
	VideoCodec _videoCodec;
	#endif
	};
	} // namespace webrtc
	#endif // WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_UTILITY_H_