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-
-/* -----------------------------------------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-© Copyright 1995 - 2013 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;
-
- FDK_ASSERT(log2_tl - 2 >= 0);
- FDK_ASSERT(log2_tl - 2 < 8*sizeof(int));
-
- /* 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;
-}
-
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