/* -----------------------------------------------------------------------------
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Software License for The Fraunhofer FDK AAC Codec Library for Android
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© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
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Forschung e.V. All rights reserved.
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1. INTRODUCTION
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The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software
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that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding
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scheme for digital audio. This FDK AAC Codec software is intended to be used on
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a wide variety of Android devices.
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AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient
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general perceptual audio codecs. AAC-ELD is considered the best-performing
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full-bandwidth communications codec by independent studies and is widely
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deployed. AAC has been standardized by ISO and IEC as part of the MPEG
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specifications.
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Patent licenses for necessary patent claims for the FDK AAC Codec (including
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those of Fraunhofer) may be obtained through Via Licensing
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(www.vialicensing.com) or through the respective patent owners individually for
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the purpose of encoding or decoding bit streams in products that are compliant
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with the ISO/IEC MPEG audio standards. Please note that most manufacturers of
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Android devices already license these patent claims through Via Licensing or
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directly from the patent owners, and therefore FDK AAC Codec software may
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already be covered under those patent licenses when it is used for those
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licensed purposes only.
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Commercially-licensed AAC software libraries, including floating-point versions
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with enhanced sound quality, are also available from Fraunhofer. Users are
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encouraged to check the Fraunhofer website for additional applications
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information and documentation.
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2. COPYRIGHT LICENSE
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Redistribution and use in source and binary forms, with or without modification,
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are permitted without payment of copyright license fees provided that you
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satisfy the following conditions:
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You must retain the complete text of this software license in redistributions of
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the FDK AAC Codec or your modifications thereto in source code form.
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You must retain the complete text of this software license in the documentation
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and/or other materials provided with redistributions of the FDK AAC Codec or
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your modifications thereto in binary form. You must make available free of
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charge copies of the complete source code of the FDK AAC Codec and your
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modifications thereto to recipients of copies in binary form.
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The name of Fraunhofer may not be used to endorse or promote products derived
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from this library without prior written permission.
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You may not charge copyright license fees for anyone to use, copy or distribute
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the FDK AAC Codec software or your modifications thereto.
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Your modified versions of the FDK AAC Codec must carry prominent notices stating
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that you changed the software and the date of any change. For modified versions
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of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android"
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must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK
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AAC Codec Library for Android."
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3. NO PATENT LICENSE
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NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without
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limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE.
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Fraunhofer provides no warranty of patent non-infringement with respect to this
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software.
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You may use this FDK AAC Codec software or modifications thereto only for
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purposes that are authorized by appropriate patent licenses.
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4. DISCLAIMER
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This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright
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holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES,
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including but not limited to the implied warranties of merchantability and
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fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
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CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary,
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or consequential damages, including but not limited to procurement of substitute
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goods or services; loss of use, data, or profits, or business interruption,
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however caused and on any theory of liability, whether in contract, strict
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liability, or tort (including negligence), arising in any way out of the use of
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this software, even if advised of the possibility of such damage.
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5. CONTACT INFORMATION
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Fraunhofer Institute for Integrated Circuits IIS
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Attention: Audio and Multimedia Departments - FDK AAC LL
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Am Wolfsmantel 33
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91058 Erlangen, Germany
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www.iis.fraunhofer.de/amm
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amm-info@iis.fraunhofer.de
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----------------------------------------------------------------------------- */
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/**************************** AAC decoder library ******************************
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Author(s):
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Description: low delay filterbank
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*******************************************************************************/
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#include "ldfiltbank.h"
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#include "aac_rom.h"
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#include "dct.h"
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#include "FDK_tools_rom.h"
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#include "mdct.h"
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#define LDFB_HEADROOM 2
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#if defined(__arm__)
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#endif
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static void multE2_DinvF_fdk(FIXP_PCM *output, FIXP_DBL *x, const FIXP_WTB *fb,
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FIXP_DBL *z, const int N) {
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int i;
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/* scale for FIXP_DBL -> INT_PCM conversion. */
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const int scale = (DFRACT_BITS - SAMPLE_BITS) - LDFB_HEADROOM;
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#if ((DFRACT_BITS - SAMPLE_BITS - LDFB_HEADROOM) > 0)
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FIXP_DBL rnd_val_wts0 = (FIXP_DBL)0;
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FIXP_DBL rnd_val_wts1 = (FIXP_DBL)0;
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if (-WTS0 - 1 + scale)
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rnd_val_wts0 = (FIXP_DBL)(1 << (-WTS0 - 1 + scale - 1));
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if (-WTS1 - 1 + scale)
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rnd_val_wts1 = (FIXP_DBL)(1 << (-WTS1 - 1 + scale - 1));
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#endif
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for (i = 0; i < N / 4; i++) {
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FIXP_DBL z0, z2, tmp;
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z2 = x[N / 2 + i];
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z0 = z2 + (fMultDiv2(z[N / 2 + i], fb[2 * N + i]) >> (-WTS2 - 1));
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z[N / 2 + i] = x[N / 2 - 1 - i] +
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(fMultDiv2(z[N + i], fb[2 * N + N / 2 + i]) >> (-WTS2 - 1));
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tmp = (fMultDiv2(z[N / 2 + i], fb[N + N / 2 - 1 - i]) +
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fMultDiv2(z[i], fb[N + N / 2 + i]));
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#if ((DFRACT_BITS - SAMPLE_BITS - LDFB_HEADROOM) > 0)
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FDK_ASSERT((-WTS1 - 1 + scale) >= 0);
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FDK_ASSERT(tmp <= ((FIXP_DBL)0x7FFFFFFF -
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rnd_val_wts1)); /* rounding must not cause overflow */
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output[(N * 3 / 4 - 1 - i)] = (FIXP_PCM)SATURATE_RIGHT_SHIFT(
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tmp + rnd_val_wts1, -WTS1 - 1 + scale, PCM_OUT_BITS);
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#else
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FDK_ASSERT((WTS1 + 1 - scale) >= 0);
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output[(N * 3 / 4 - 1 - i)] =
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(FIXP_PCM)SATURATE_LEFT_SHIFT(tmp, WTS1 + 1 - scale, PCM_OUT_BITS);
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#endif
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z[i] = z0;
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z[N + i] = z2;
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}
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for (i = N / 4; i < N / 2; i++) {
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FIXP_DBL z0, z2, tmp0, tmp1;
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z2 = x[N / 2 + i];
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z0 = z2 + (fMultDiv2(z[N / 2 + i], fb[2 * N + i]) >> (-WTS2 - 1));
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z[N / 2 + i] = x[N / 2 - 1 - i] +
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(fMultDiv2(z[N + i], fb[2 * N + N / 2 + i]) >> (-WTS2 - 1));
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tmp0 = (fMultDiv2(z[N / 2 + i], fb[N / 2 - 1 - i]) +
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fMultDiv2(z[i], fb[N / 2 + i]));
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tmp1 = (fMultDiv2(z[N / 2 + i], fb[N + N / 2 - 1 - i]) +
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fMultDiv2(z[i], fb[N + N / 2 + i]));
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#if ((DFRACT_BITS - SAMPLE_BITS - LDFB_HEADROOM) > 0)
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FDK_ASSERT((-WTS0 - 1 + scale) >= 0);
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FDK_ASSERT(tmp0 <= ((FIXP_DBL)0x7FFFFFFF -
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rnd_val_wts0)); /* rounding must not cause overflow */
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FDK_ASSERT(tmp1 <= ((FIXP_DBL)0x7FFFFFFF -
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rnd_val_wts1)); /* rounding must not cause overflow */
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output[(i - N / 4)] = (FIXP_PCM)SATURATE_RIGHT_SHIFT(
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tmp0 + rnd_val_wts0, -WTS0 - 1 + scale, PCM_OUT_BITS);
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output[(N * 3 / 4 - 1 - i)] = (FIXP_PCM)SATURATE_RIGHT_SHIFT(
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tmp1 + rnd_val_wts1, -WTS1 - 1 + scale, PCM_OUT_BITS);
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#else
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FDK_ASSERT((WTS0 + 1 - scale) >= 0);
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output[(i - N / 4)] =
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(FIXP_PCM)SATURATE_LEFT_SHIFT(tmp0, WTS0 + 1 - scale, PCM_OUT_BITS);
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output[(N * 3 / 4 - 1 - i)] =
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(FIXP_PCM)SATURATE_LEFT_SHIFT(tmp1, WTS1 + 1 - scale, PCM_OUT_BITS);
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#endif
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z[i] = z0;
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z[N + i] = z2;
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}
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/* Exchange quarter parts of x to bring them in the "right" order */
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for (i = 0; i < N / 4; i++) {
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FIXP_DBL tmp0 = fMultDiv2(z[i], fb[N / 2 + i]);
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#if ((DFRACT_BITS - SAMPLE_BITS - LDFB_HEADROOM) > 0)
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FDK_ASSERT((-WTS0 - 1 + scale) >= 0);
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FDK_ASSERT(tmp0 <= ((FIXP_DBL)0x7FFFFFFF -
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rnd_val_wts0)); /* rounding must not cause overflow */
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output[(N * 3 / 4 + i)] = (FIXP_PCM)SATURATE_RIGHT_SHIFT(
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tmp0 + rnd_val_wts0, -WTS0 - 1 + scale, PCM_OUT_BITS);
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#else
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FDK_ASSERT((WTS0 + 1 - scale) >= 0);
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output[(N * 3 / 4 + i)] =
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(FIXP_PCM)SATURATE_LEFT_SHIFT(tmp0, WTS0 + 1 - scale, PCM_OUT_BITS);
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#endif
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}
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}
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int InvMdctTransformLowDelay_fdk(FIXP_DBL *mdctData, const int mdctData_e,
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FIXP_PCM *output, FIXP_DBL *fs_buffer,
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const int N) {
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const FIXP_WTB *coef;
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FIXP_DBL gain = (FIXP_DBL)0;
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int scale = mdctData_e + MDCT_OUT_HEADROOM -
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LDFB_HEADROOM; /* The LDFB_HEADROOM is compensated inside
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multE2_DinvF_fdk() below */
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int i;
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/* Select LD window slope */
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switch (N) {
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case 256:
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coef = LowDelaySynthesis256;
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break;
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case 240:
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coef = LowDelaySynthesis240;
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break;
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case 160:
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coef = LowDelaySynthesis160;
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break;
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case 128:
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coef = LowDelaySynthesis128;
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break;
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case 120:
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coef = LowDelaySynthesis120;
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break;
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case 512:
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coef = LowDelaySynthesis512;
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break;
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case 480:
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default:
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coef = LowDelaySynthesis480;
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break;
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}
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/*
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Apply exponent and 1/N factor.
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Note: "scale" is off by one because for LD_MDCT the window length is twice
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the window length of a regular MDCT. This is corrected inside
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multE2_DinvF_fdk(). Refer to ISO/IEC 14496-3:2009 page 277,
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chapter 4.6.20.2 "Low Delay Window".
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*/
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imdct_gain(&gain, &scale, N);
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dct_IV(mdctData, N, &scale);
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if (N == 256 || N == 240 || N == 160) {
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scale -= 1;
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} else if (N == 128 || N == 120) {
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scale -= 2;
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}
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if (gain != (FIXP_DBL)0) {
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for (i = 0; i < N; i++) {
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mdctData[i] = fMult(mdctData[i], gain);
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}
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}
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scaleValuesSaturate(mdctData, N, scale);
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/* Since all exponent and factors have been applied, current exponent is zero.
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*/
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multE2_DinvF_fdk(output, mdctData, coef, fs_buffer, N);
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return (1);
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}
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