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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;
+}
+