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author | The Android Open Source Project <initial-contribution@android.com> | 2012-07-11 10:15:24 -0700 |
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committer | The Android Open Source Project <initial-contribution@android.com> | 2012-07-11 10:15:24 -0700 |
commit | 2228e360595641dd906bf1773307f43d304f5b2e (patch) | |
tree | 57f3d390ebb0782cc0de0fb984c8ea7e45b4f386 /libFDK/src/mdct.cpp | |
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Snapshot 2bda038c163298531d47394bc2c09e1409c5d0db
Change-Id: If584e579464f28b97d50e51fc76ba654a5536c54
Diffstat (limited to 'libFDK/src/mdct.cpp')
-rw-r--r-- | libFDK/src/mdct.cpp | 385 |
1 files changed, 385 insertions, 0 deletions
diff --git a/libFDK/src/mdct.cpp b/libFDK/src/mdct.cpp new file mode 100644 index 0000000..7b9864f --- /dev/null +++ b/libFDK/src/mdct.cpp @@ -0,0 +1,385 @@ + +/* ----------------------------------------------------------------------------------------------------------- +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; +} + |