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authorThe Android Open Source Project <initial-contribution@android.com>2012-07-11 10:15:24 -0700
committerThe Android Open Source Project <initial-contribution@android.com>2012-07-11 10:15:24 -0700
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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 Tools **********************
+
+ Author(s): Josef Hoepfl, Manuel Jander
+ Description: MDCT routines
+
+******************************************************************************/
+
+#include "mdct.h"
+
+
+#include "FDK_tools_rom.h"
+#include "dct.h"
+#include "fixpoint_math.h"
+
+
+void mdct_init( H_MDCT hMdct,
+ FIXP_DBL *overlap,
+ INT overlapBufferSize )
+{
+ hMdct->overlap.freq = overlap;
+ //FDKmemclear(overlap, overlapBufferSize*sizeof(FIXP_DBL));
+ hMdct->prev_fr = 0;
+ hMdct->prev_nr = 0;
+ hMdct->prev_tl = 0;
+ hMdct->ov_size = overlapBufferSize;
+}
+
+
+void imdct_gain(FIXP_DBL *pGain_m, int *pGain_e, int tl)
+{
+ FIXP_DBL gain_m = *pGain_m;
+ int gain_e = *pGain_e;
+ int log2_tl;
+
+ log2_tl = DFRACT_BITS-1-fNormz((FIXP_DBL)tl);
+
+ gain_e += -MDCT_OUTPUT_GAIN - log2_tl - MDCT_OUT_HEADROOM + 1;
+
+ /* Detect non-radix 2 transform length and add amplitude compensation factor
+ which cannot be included into the exponent above */
+ switch ( (tl) >> (log2_tl - 2) ) {
+ case 0x7: /* 10 ms, 1/tl = 1.0/(FDKpow(2.0, -log2_tl) * 0.53333333333333333333) */
+ if (gain_m == (FIXP_DBL)0) {
+ gain_m = FL2FXCONST_DBL(0.53333333333333333333f);
+ } else {
+ gain_m = fMult(gain_m, FL2FXCONST_DBL(0.53333333333333333333f));
+ }
+ break;
+ case 0x6: /* 3/4 of radix 2, 1/tl = 1.0/(FDKpow(2.0, -log2_tl) * 2.0/3.0) */
+ if (gain_m == (FIXP_DBL)0) {
+ gain_m = FL2FXCONST_DBL(2.0/3.0f);
+ } else {
+ gain_m = fMult(gain_m, FL2FXCONST_DBL(2.0/3.0f));
+ }
+ break;
+ case 0x4:
+ /* radix 2, nothing to do. */
+ break;
+ default:
+ /* unsupported */
+ FDK_ASSERT(0);
+ break;
+ }
+
+ *pGain_m = gain_m;
+ *pGain_e = gain_e;
+}
+
+INT imdct_drain(
+ H_MDCT hMdct,
+ FIXP_DBL *output,
+ INT nrSamplesRoom
+ )
+{
+ int buffered_samples = 0;
+
+ if (nrSamplesRoom > 0) {
+ buffered_samples = hMdct->ov_offset;
+
+ FDK_ASSERT(buffered_samples <= nrSamplesRoom);
+
+ if (buffered_samples > 0) {
+ FDKmemcpy(output, hMdct->overlap.time, buffered_samples*sizeof(FIXP_DBL));
+ hMdct->ov_offset = 0;
+ }
+ }
+ return buffered_samples;
+}
+
+INT imdct_copy_ov_and_nr(
+ H_MDCT hMdct,
+ FIXP_DBL * pTimeData,
+ INT nrSamples
+ )
+{
+ FIXP_DBL *pOvl;
+ int nt, nf, i;
+
+ nt = fMin(hMdct->ov_offset, nrSamples);
+ nrSamples -= nt;
+ nf = fMin(hMdct->prev_nr, nrSamples);
+ nrSamples -= nf;
+ FDKmemcpy(pTimeData, hMdct->overlap.time, nt*sizeof(FIXP_DBL));
+ pTimeData += nt;
+
+ pOvl = hMdct->overlap.freq + hMdct->ov_size - 1;
+ for (i=0; i<nf; i++) {
+ FIXP_DBL x = - (*pOvl--);
+ *pTimeData = IMDCT_SCALE_DBL(x);
+ pTimeData ++;
+ }
+
+ return (nt+nf);
+}
+
+void imdct_adapt_parameters(H_MDCT hMdct, int *pfl, int *pnl, int tl, const FIXP_WTP *wls, int noOutSamples)
+{
+ int fl = *pfl, nl = *pnl;
+ int window_diff, use_current = 0, use_previous = 0;
+ if (hMdct->prev_tl == 0) {
+ hMdct->prev_wrs = wls;
+ hMdct->prev_fr = fl;
+ hMdct->prev_nr = (noOutSamples-fl)>>1;
+ hMdct->prev_tl = noOutSamples;
+ hMdct->ov_offset = 0;
+ use_current = 1;
+ }
+
+ window_diff = (hMdct->prev_fr - fl)>>1;
+
+ /* check if the previous window slope can be adjusted to match the current window slope */
+ if (hMdct->prev_nr + window_diff > 0) {
+ use_current = 1;
+ }
+ /* check if the current window slope can be adjusted to match the previous window slope */
+ if (nl - window_diff > 0 ) {
+ use_previous = 1;
+ }
+
+ /* if both is possible choose the larger of both window slope lengths */
+ if (use_current && use_previous) {
+ if (fl < hMdct->prev_fr) {
+ use_current = 0;
+ } else {
+ use_previous = 0;
+ }
+ }
+ /*
+ * If the previous transform block is big enough, enlarge previous window overlap,
+ * if not, then shrink current window overlap.
+ */
+ if (use_current) {
+ hMdct->prev_nr += window_diff;
+ hMdct->prev_fr = fl;
+ hMdct->prev_wrs = wls;
+ } else {
+ nl -= window_diff;
+ fl = hMdct->prev_fr;
+ }
+
+ *pfl = fl;
+ *pnl = nl;
+}
+
+INT imdct_block(
+ H_MDCT hMdct,
+ FIXP_DBL *output,
+ FIXP_DBL *spectrum,
+ const SHORT scalefactor[],
+ const INT nSpec,
+ const INT noOutSamples,
+ const INT tl,
+ const FIXP_WTP *wls,
+ INT fl,
+ const FIXP_WTP *wrs,
+ const INT fr,
+ FIXP_DBL gain
+ )
+{
+ FIXP_DBL *pOvl;
+ FIXP_DBL *pOut0 = output, *pOut1;
+ INT nl, nr;
+ int w, i, nrSamples = 0, specShiftScale, transform_gain_e = 0;
+
+ /* Derive NR and NL */
+ nr = (tl - fr)>>1;
+ nl = (tl - fl)>>1;
+
+ /* Include 2/N IMDCT gain into gain factor and exponent. */
+ imdct_gain(&gain, &transform_gain_e, tl);
+
+ /* Detect FRprevious / FL mismatches and override parameters accordingly */
+ if (hMdct->prev_fr != fl) {
+ imdct_adapt_parameters(hMdct, &fl, &nl, tl, wls, noOutSamples);
+ }
+
+ pOvl = hMdct->overlap.freq + hMdct->ov_size - 1;
+
+ if ( noOutSamples > nrSamples ) {
+ /* Purge buffered output. */
+ for (i=0; i<hMdct->ov_offset; i++) {
+ *pOut0 = hMdct->overlap.time[i];
+ pOut0 ++;
+ }
+ nrSamples = hMdct->ov_offset;
+ hMdct->ov_offset = 0;
+ }
+
+ for (w=0; w<nSpec; w++)
+ {
+ FIXP_DBL *pSpec, *pCurr;
+ const FIXP_WTP *pWindow;
+
+ specShiftScale = transform_gain_e;
+
+ /* Setup window pointers */
+ pWindow = hMdct->prev_wrs;
+
+ /* Current spectrum */
+ pSpec = spectrum+w*tl;
+
+ /* DCT IV of current spectrum. */
+ dct_IV(pSpec, tl, &specShiftScale);
+
+ /* Optional scaling of time domain - no yet windowed - of current spectrum */
+ /* and de-scale current spectrum signal (time domain, no yet windowed) */
+ if (gain != (FIXP_DBL)0) {
+ scaleValuesWithFactor(pSpec, gain, tl, scalefactor[w] + specShiftScale);
+ } else {
+ scaleValues(pSpec, tl, scalefactor[w] + specShiftScale);
+ }
+
+ if ( noOutSamples <= nrSamples ) {
+ /* Divert output first half to overlap buffer if we already got enough output samples. */
+ pOut0 = hMdct->overlap.time + hMdct->ov_offset;
+ hMdct->ov_offset += hMdct->prev_nr + fl/2;
+ } else {
+ /* Account output samples */
+ nrSamples += hMdct->prev_nr + fl/2;
+ }
+
+ /* NR output samples 0 .. NR. -overlap[TL/2..TL/2-NR] */
+ for (i=0; i<hMdct->prev_nr; i++) {
+ FIXP_DBL x = - (*pOvl--);
+ *pOut0 = IMDCT_SCALE_DBL(x);
+ pOut0 ++;
+ }
+
+ if ( noOutSamples <= nrSamples ) {
+ /* Divert output second half to overlap buffer if we already got enough output samples. */
+ pOut1 = hMdct->overlap.time + hMdct->ov_offset + fl/2 - 1;
+ hMdct->ov_offset += fl/2 + nl;
+ } else {
+ pOut1 = pOut0 + (fl - 1);
+ nrSamples += fl/2 + nl;
+ }
+
+ /* output samples before window crossing point NR .. TL/2. -overlap[TL/2-NR..TL/2-NR-FL/2] + current[NR..TL/2] */
+ /* output samples after window crossing point TL/2 .. TL/2+FL/2. -overlap[0..FL/2] - current[TL/2..FL/2] */
+ pCurr = pSpec + tl - fl/2;
+ for (i=0; i<fl/2; i++) {
+ FIXP_DBL x0, x1;
+
+ cplxMult(&x1, &x0, *pCurr++, - *pOvl--, pWindow[i]);
+ *pOut0 = IMDCT_SCALE_DBL(x0);
+ *pOut1 = IMDCT_SCALE_DBL(-x1);
+ pOut0 ++;
+ pOut1 --;
+ }
+ pOut0 += (fl/2);
+
+ /* NL output samples TL/2+FL/2..TL. - current[FL/2..0] */
+ pOut1 += (fl/2) + 1;
+ pCurr = pSpec + tl - fl/2 - 1;
+ for (i=0; i<nl; i++) {
+ FIXP_DBL x = - (*pCurr--);
+ *pOut1 = IMDCT_SCALE_DBL(x);
+ pOut1 ++;
+ }
+
+ /* Set overlap source pointer for next window pOvl = pSpec + tl/2 - 1; */
+ pOvl = pSpec + tl/2 - 1;
+
+ /* Previous window values. */
+ hMdct->prev_nr = nr;
+ hMdct->prev_fr = fr;
+ hMdct->prev_tl = tl;
+ hMdct->prev_wrs = wrs;
+ }
+
+ /* Save overlap */
+
+ pOvl = hMdct->overlap.freq + hMdct->ov_size - tl/2;
+ FDK_ASSERT(pOvl >= hMdct->overlap.time + hMdct->ov_offset);
+ FDK_ASSERT(tl/2 <= hMdct->ov_size);
+ for (i=0; i<tl/2; i++) {
+ pOvl[i] = spectrum[i+(nSpec-1)*tl];
+ }
+
+ return nrSamples;
+}
+