Snapshot 2bda038c163298531d47394bc2c09e1409c5d0db

Change-Id: If584e579464f28b97d50e51fc76ba654a5536c54

1 files changed, 554 insertions, 0 deletions

diff --git a/libSYS/src/wav_file.cpp b/libSYS/src/wav_file.cpp
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+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 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
+software or your modifications thereto.
+
+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."
+
+3.    NO PATENT LICENSE
+
+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,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+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;
+      UINT channelMask;
+      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(&(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->fp) {
+      FDKfclose(wav->fp);
+      wav->fp = NULL;
+    }
+
+    if (wav) {
+      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 (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->fp)
+    FDKfclose(wav->fp);
+  if (wav)
+    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;
+}
+