wav_file.cpp - Android社区 - https://www.androidos.net.cn/

/* -----------------------------------------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android

1.    INTRODUCTION
The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
This FDK AAC Codec software is intended to be used on a wide variety of Android devices.

AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
of the MPEG specifications.

Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
individually for the purpose of encoding or decoding bit streams in products that are compliant with
the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
software may already be covered under those patent licenses when it is used for those licensed purposes only.

Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
applications information and documentation.

2.    COPYRIGHT LICENSE

Redistribution and use in source and binary forms, with or without modification, are permitted without
payment of copyright license fees provided that you satisfy the following conditions:

You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
your modifications thereto in source code form.

You must retain the complete text of this software license in the documentation and/or other materials
provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
modifications thereto to recipients of copies in binary form.

The name of Fraunhofer may not be used to endorse or promote products derived from this library without
prior written permission.

You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
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Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
and the date of any change. For modified versions of the FDK AAC Codec, the term
"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."

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NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
respect to this software.

You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
by appropriate patent licenses.

4.    DISCLAIMER

This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
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or business interruption, however caused and on any theory of liability, whether in contract, strict
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advised of the possibility of such damage.

5.    CONTACT INFORMATION

Fraunhofer Institute for Integrated Circuits IIS
Attention: Audio and Multimedia Departments - FDK AAC LL
Am Wolfsmantel 33
91058 Erlangen, Germany

www.iis.fraunhofer.de/amm
amm-info@iis.fraunhofer.de
----------------------------------------------------------------------------------------------------------- */

/**************************  Fraunhofer IIS FDK SysLib  **********************

Author(s):   Eric Allamanche
   Description: a rudimentary wav file interface

******************************************************************************/

#include "wav_file.h"
#include "genericStds.h"

static INT_PCM ulaw2pcm (UCHAR ulawbyte);

/*!
 *
 *  \brief Read header from a WAVEfile. Host endianess is handled accordingly.
 *  \wav->fp filepointer of type FILE*.
 *  \wavinfo SWavInfo struct where the decoded header info is stored into.
 *  \return 0 on success and non-zero on failure.
 *
 */
INT WAV_InputOpen (HANDLE_WAV *pWav, const char *filename)
{
    HANDLE_WAV wav = (HANDLE_WAV)FDKcalloc(1, sizeof(struct WAV));
    INT offset;

if (wav == NULL) {
      FDKprintfErr("WAV_InputOpen(): Unable to allocate WAV struct.\n");
      goto error;
    }

wav->fp = FDKfopen(filename, "rb");
    if (wav->fp == NULL) {
      FDKprintfErr("WAV_InputOpen(): Unable to open wav file. %s\n", filename);
      goto error;
    }

/* read RIFF-chunk */
    if (FDKfread(&(wav->header.riffType), 1, 4, wav->fp) != 4) {
      FDKprintfErr("WAV_InputOpen(): couldn't read RIFF_ID\n");
      goto error;  /* bad error "couldn't read RIFF_ID" */
    }
    if (FDKstrncmp("RIFF", wav->header.riffType, 4)) {
      FDKprintfErr("WAV_InputOpen(): RIFF descriptor not found.\n") ;
      goto error;
    }

/* Read RIFF size. Ignored. */
    FDKfread_EL(&(wav->header.riffSize), 4, 1, wav->fp);

/* read WAVE-chunk */
    if (FDKfread(&wav->header.waveType, 1, 4, wav->fp) !=4) {
      FDKprintfErr("WAV_InputOpen(): couldn't read format\n");
      goto error;  /* bad error "couldn't read format" */
    }
    if (FDKstrncmp("WAVE", wav->header.waveType, 4)) {
      FDKprintfErr("WAV_InputOpen(): WAVE chunk ID not found.\n") ;
      goto error;
    }

/* read format-chunk */
    if (FDKfread(&(wav->header.formatType), 1, 4, wav->fp) != 4) {
      FDKprintfErr("WAV_InputOpen(): couldn't read format_ID\n");
      goto error;  /* bad error "couldn't read format_ID" */
    }
    if (FDKstrncmp("fmt", wav->header.formatType, 3)) {
      FDKprintfErr("WAV_InputOpen(): fmt chunk format not found.\n") ;
     goto error;
    }

FDKfread_EL(&wav->header.formatSize, 4, 1, wav->fp);   /* should be 16 for PCM-format (uncompressed) */

/* read  info */
    FDKfread_EL(&(wav->header.compressionCode), 2, 1, wav->fp);
    FDKfread_EL(&(wav->header.numChannels), 2, 1, wav->fp);
    FDKfread_EL(&(wav->header.sampleRate), 4, 1, wav->fp);
    FDKfread_EL(&(wav->header.bytesPerSecond), 4, 1, wav->fp);
    FDKfread_EL(&(wav->header.blockAlign), 2, 1, wav->fp);
    FDKfread_EL(&(wav->header.bitsPerSample), 2, 1, wav->fp);

offset = wav->header.formatSize - 16;

/* Wave format extensible */
    if (wav->header.compressionCode == 0xFFFE) {
      static const UCHAR guidPCM[16] = {
          0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x10, 0x00,
          0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71
      };
      USHORT extraFormatBytes, validBitsPerSample;
      UCHAR guid[16];
      INT i;

/* read extra bytes */
      FDKfread_EL(&(extraFormatBytes), 2, 1, wav->fp);
      offset -= 2;

if (extraFormatBytes >= 22) {
        FDKfread_EL(&(validBitsPerSample), 2, 1, wav->fp);
        FDKfread_EL(&(wav->channelMask), 4, 1, wav->fp);
        FDKfread_EL(&(guid), 16, 1, wav->fp);

/* check for PCM GUID */
        for (i = 0; i < 16; i++) if (guid[i] != guidPCM[i]) break;
        if (i == 16) wav->header.compressionCode = 0x01;

offset -= 22;
      }
    }

/* Skip rest of fmt header if any. */
    for (;offset > 0; offset--) {
      FDKfread(&wav->header.formatSize, 1, 1, wav->fp);
    }

do {
      /* Read data chunk ID */
      if (FDKfread(wav->header.dataType, 1, 4, wav->fp) != 4) {
        FDKprintfErr("WAV_InputOpen(): Unable to read data chunk ID.\n");
        FDKfree(wav);
        goto error;
      }

/* Read chunk length. */
      FDKfread_EL(&offset, 4, 1, wav->fp);

/* Check for data chunk signature. */
      if (FDKstrncmp("data", wav->header.dataType, 4) == 0) {
        wav->header.dataSize = offset;
        break;
      }
      /* Jump over non data chunk. */
      for (;offset > 0; offset--) {
        FDKfread(&(wav->header.dataSize), 1, 1, wav->fp);
      }
    } while (!FDKfeof(wav->fp));

/* return success */
    *pWav = wav;
    return 0;

/* Error path */
error:

if (wav) {
      if (wav->fp) {
        FDKfclose(wav->fp);
        wav->fp = NULL;
      }
      FDKfree(wav);
    }

*pWav = NULL;

return -1;
}

/*!
 *
 *  \brief Read samples from a WAVEfile. The samples are automatically reorder to the native
 *    host endianess and scaled to full scale of the INT_PCM type, from whatever bps the WAVEfile
 *    had specified in its haader data.
 *
 *  \wav HANDLE_WAV of the wav file.
 *  \buffer Pointer to store read data.
 *  \numSamples Desired number of samples to read.
 *  \nBits sample size in bits to be used for the buffer
 *
 *  \return Number of samples actually read.
 *
 */

INT WAV_InputRead (HANDLE_WAV wav, void *buffer, UINT numSamples, int nBits)
{
  UINT result = 0 ;
  UINT i;
  SCHAR *bptr = (SCHAR*)buffer;
  LONG  *lptr = (LONG*)buffer;
  SHORT *sptr = (SHORT*)buffer;

switch (wav->header.compressionCode)
  {
    case 0x01:  /* PCM uncompressed */
      if (nBits == wav->header.bitsPerSample) {
        result = FDKfread_EL(buffer, wav->header.bitsPerSample >> 3, numSamples, wav->fp) ;
      } else {
        result = 0;
        for (i=0; i<numSamples; i++)
        {
          LONG tmp = 0;
          result += FDKfread_EL(&tmp, wav->header.bitsPerSample >> 3, 1, wav->fp) ;

/* Move read bits to lower bits of LONG. */
          if ( !IS_LITTLE_ENDIAN() && wav->header.bitsPerSample != 24 && wav->header.bitsPerSample < 32) {
            tmp >>= (32-wav->header.bitsPerSample);
          }

/* Full scale */
          if (wav->header.bitsPerSample > nBits)
            tmp >>= (wav->header.bitsPerSample-nBits);
          else
            tmp <<= (nBits-wav->header.bitsPerSample);
            
          if (nBits == 8)
            *bptr++ = (SCHAR) tmp;
          if (nBits == 16)
            *sptr++ = (SHORT) tmp;
          if (nBits == 32)
            *lptr++ = (LONG) tmp;
        }
      }
      break;

case 0x07:  /* u-Law compression */
      for (i=0; i<numSamples; i++) {
        result += FDKfread(&(bptr[i<<1]), 1, 1, wav->fp) ;
        sptr[i] = ulaw2pcm(bptr[i<<1]) ;
      }
      break ;

default:
      FDKprintf("WAV_InputRead(): unsupported data-compression!!") ;
      break ;
  }
  return result ;
}

void WAV_InputClose(HANDLE_WAV *pWav)
{
  HANDLE_WAV wav = *pWav;

if (wav != NULL) {
    if (wav->fp != NULL) {
       FDKfclose(wav->fp);
       wav->fp = NULL;
    }
    if (wav) {
       FDKfree(wav);
    }
  }
  *pWav = NULL;
}

/* conversion of u-law to linear coding */
static INT_PCM ulaw2pcm (UCHAR ulawbyte)
{
  static const INT exp_lut[8] = { 0, 132, 396, 924, 1980, 4092, 8316, 16764 } ;
  INT sign, exponent, mantissa, sample ;

ulawbyte = (UCHAR)~ulawbyte ;
  sign = (ulawbyte & 0x80) ;
  exponent = (ulawbyte >> 4) & 0x07 ;
  mantissa = ulawbyte & 0x0F ;

sample = exp_lut[exponent] + (mantissa << (exponent + 3)) ;
  if (sign != 0)
    sample = -sample ;

return (INT_PCM)sample ;
}

/************** Writer ***********************/

static UINT LittleEndian32(UINT v)
{
    if (IS_LITTLE_ENDIAN())
        return v ;
    else
        return (v & 0x000000FF) << 24 | (v & 0x0000FF00) <<  8 | (v & 0x00FF0000) >>  8 | (v & 0xFF000000) >> 24;
}

static SHORT LittleEndian16(SHORT v)
{
    if (IS_LITTLE_ENDIAN())
        return v;
    else
        return (SHORT)(((v << 8) & 0xFF00) | ((v >> 8) & 0x00FF));
}

static USHORT Unpack(USHORT v)
{
    if (IS_LITTLE_ENDIAN())
      return v;
    else
      return (SHORT)(((v << 8) & 0xFF00) | ((v >> 8) & 0x00FF));
}

/**
 * WAV_OutputOpen
 * \brief Open WAV output/writer handle
 * \param pWav pointer to WAV handle to be returned
 * \param sampleRate desired samplerate of the resulting WAV file
 * \param numChannels desired number of audio channels of the resulting WAV file
 * \param bitsPerSample desired number of bits per audio sample of the resulting WAV file
 *
 * \return value:   0: ok
 *                 -1: error
 */
INT WAV_OutputOpen(HANDLE_WAV *pWav, const char *outputFilename, INT sampleRate, INT numChannels, INT bitsPerSample)
{
  HANDLE_WAV wav = (HANDLE_WAV)FDKcalloc(1, sizeof(struct WAV));
  UINT size = 0;

if (wav == NULL) {
    FDKprintfErr("WAV_OutputOpen(): Unable to allocate WAV struct.\n");
    goto bail;
  }

if (bitsPerSample != 16 && bitsPerSample != 24 && bitsPerSample != 32)
  {
      FDKprintfErr("WAV_OutputOpen(): Invalid argument (bitsPerSample).\n");
      goto bail;
  }

wav->fp = FDKfopen(outputFilename, "wb");
  if (wav->fp == NULL)
  {
      FDKprintfErr("WAV_OutputOpen(): unable to create file %s\n", outputFilename);
      goto bail;
  }

FDKstrcpy(wav->header.riffType, "RIFF");
  wav->header.riffSize = LittleEndian32(0x7fffffff);  /* in case fseek() doesn't work later in WAV_OutputClose() */
  FDKstrcpy(wav->header.waveType, "WAVE");

FDKstrcpy(wav->header.formatType, "fmt ");
  wav->header.formatSize = LittleEndian32(16);

wav->header.compressionCode = LittleEndian16(0x01);
  wav->header.bitsPerSample   = LittleEndian16((SHORT)bitsPerSample);
  wav->header.numChannels     = LittleEndian16((SHORT)numChannels);
  wav->header.blockAlign      = LittleEndian16((SHORT)(numChannels * (bitsPerSample >> 3)));
  wav->header.sampleRate      = LittleEndian32(sampleRate);
  wav->header.bytesPerSecond  = LittleEndian32(sampleRate * wav->header.blockAlign);
  FDKstrcpy(wav->header.dataType, "data");
  wav->header.dataSize        = LittleEndian32(0x7fffffff - 36);

size = sizeof(WAV_HEADER);
  if (FDKfwrite(&wav->header, 1, size, wav->fp) != size)
  {
      FDKprintfErr("WAV_OutputOpen(): error writing to output file %s\n", outputFilename);
      goto bail;
  }

wav->header.dataSize = wav->header.riffSize = 0;

*pWav = wav;

return 0;

bail:
  if (wav) {
    if (wav->fp) {
      FDKfclose(wav->fp);
    }
    FDKfree(wav);
  }

pWav = NULL;

return -1;
}

/**
 * WAV_OutputWrite
 * \brief Write data to WAV file asociated to WAV handle
 *
 * \param wav handle of wave file
 * \param sampleBuffer pointer to audio samples, right justified integer values
 * \param nBufBits size in bits of each audio sample in sampleBuffer
 * \param nSigBits amount of significant bits of each nBufBits in sampleBuffer
 *
 * \return value:    0: ok
 *                  -1: error
 */
INT WAV_OutputWrite(HANDLE_WAV wav, void *sampleBuffer, UINT numberOfSamples, int nBufBits, int nSigBits)
{
    SCHAR *bptr = (SCHAR*)sampleBuffer;
    SHORT *sptr = (SHORT*)sampleBuffer;
    LONG  *lptr = (LONG*)sampleBuffer;
    LONG tmp;
    
    int bps = Unpack(wav->header.bitsPerSample);
    UINT i;

/* Pack samples if required */
    if (bps == nBufBits && bps == nSigBits) {
      if (FDKfwrite_EL(sampleBuffer, (bps>>3), numberOfSamples, wav->fp) != numberOfSamples)
      {
        FDKprintfErr("WAV_OutputWrite(): error: unable to write to file %d\n", wav->fp);
        return -1;
      }
    } else  {
      for (i=0; i<numberOfSamples; i++)
      {
        int result;
        int shift;
        
        switch (nBufBits) {
          case 8:  tmp = *bptr++; break;
          case 16: tmp = *sptr++; break;
          case 32: tmp = *lptr++; break;
          default: return -1;
        }
        /* Adapt sample size */
        shift = (nBufBits-nSigBits)-(32-bps);

/* Correct alignment difference between 32 bit data buffer "tmp" and 24 bits to be written. */
        if ( !IS_LITTLE_ENDIAN() && bps == 24) {
          shift += 8;
        }

if (shift < 0)
          tmp >>= -shift;
        else
          tmp <<= shift;

/* Write sample */        
        result=FDKfwrite_EL(&tmp, bps>>3, 1, wav->fp);
        if (result <= 0) {
          FDKprintfErr("WAV_OutputWrite(): error: unable to write to file %d\n", wav->fp);
          return -1;
        }
      }
    }

wav->header.dataSize += (numberOfSamples * (bps>>3));
    return 0;
}

/**
 * WAV_OutputClose
 * \brief Close WAV Output handle
 * \param pWav pointer to WAV handle. *pWav is set to NULL.
 */
void WAV_OutputClose(HANDLE_WAV *pWav)
{
  HANDLE_WAV wav = *pWav;
  UINT size = 0;

if ( wav == NULL ) {
      return;
  }

wav->header.dataSize = LittleEndian32(wav->header.dataSize);
  wav->header.riffSize = LittleEndian32(wav->header.dataSize + 36);

if (wav->fp != NULL)
  {
    if (FDKfseek(wav->fp, 0, FDKSEEK_SET)) {
       FDKprintf("WAV_OutputClose(): fseek() failed.\n");
    }

size = sizeof(WAV_HEADER);
    if (FDKfwrite(&wav->header.riffType, 1, size, wav->fp) != size)
    {
        FDKprintfErr("WAV_OutputClose(): unable to write header\n");
	}

if (FDKfclose(wav->fp))
    {
        FDKprintfErr("WAV_OutputClose(): unable to close wav file\n");
    }
    wav->fp = NULL;
  }

FDKfree(wav);
  *pWav = NULL;
}

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