Include patched FDK-AAC in the repository

The initial idea was to get the DAB+ patch into upstream, but since
that follows the android source releases, there is no place for a custom
DAB+ patch there.

So instead of having to maintain a patched fdk-aac that has to have the
same .so version as the distribution package on which it is installed,
we prefer having a separate fdk-aac-dab library to avoid collision.

At that point, there's no reason to keep fdk-aac in a separate
repository, as odr-audioenc is the only tool that needs DAB+ encoding
support. Including it here simplifies installation, and makes it
consistent with toolame-dab, also shipped in this repository.

DAB+ decoding support (needed by ODR-SourceCompanion, dablin, etisnoop,
welle.io and others) can be done using upstream FDK-AAC.

1 files changed, 487 insertions, 0 deletions

diff --git a/fdk-aac/libFDK/src/FDK_lpc.cpp b/fdk-aac/libFDK/src/FDK_lpc.cpp
new file mode 100644
index 0000000..7d7e691
--- /dev/null
+++ b/fdk-aac/libFDK/src/FDK_lpc.cpp
@@ -0,0 +1,487 @@
+/* -----------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2018 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
+----------------------------------------------------------------------------- */
+
+/******************* Library for basic calculation routines ********************
+
+   Author(s):   Manuel Jander
+
+   Description: LPC related functions
+
+*******************************************************************************/
+
+#include "FDK_lpc.h"
+
+/* Internal scaling of LPC synthesis to avoid overflow of filte states.
+   This depends on the LPC order, because the LPC order defines the amount
+   of MAC operations. */
+static SCHAR order_ld[LPC_MAX_ORDER] = {
+    /* Assume that Synthesis filter output does not clip and filter
+       accu does change no more than 1.0 for each iteration.
+       ceil(0.5*log((1:24))/log(2)) */
+    0, 1, 1, 1, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 3, 3, 3, 3, 3, 3, 3, 3};
+
+/* IIRLattice */
+#ifndef FUNCTION_CLpc_SynthesisLattice_SGL
+void CLpc_SynthesisLattice(FIXP_DBL *signal, const int signal_size,
+                           const int signal_e, const int signal_e_out,
+                           const int inc, const FIXP_SGL *coeff,
+                           const int order, FIXP_DBL *state) {
+  int i, j;
+  FIXP_DBL *pSignal;
+  int shift;
+
+  FDK_ASSERT(order <= LPC_MAX_ORDER);
+  FDK_ASSERT(order > 0);
+
+  if (inc == -1)
+    pSignal = &signal[signal_size - 1];
+  else
+    pSignal = &signal[0];
+
+  /*
+    tmp = x(k) - K(M)*g(M);
+    for m=M-1:-1:1
+            tmp = tmp - K(m) * g(m);
+            g(m+1) = g(m) + K(m) * tmp;
+    endfor
+    g(1) = tmp;
+
+    y(k) = tmp;
+  */
+
+  shift = -order_ld[order - 1];
+
+  for (i = signal_size; i != 0; i--) {
+    FIXP_DBL *pState = state + order - 1;
+    const FIXP_SGL *pCoeff = coeff + order - 1;
+    FIXP_DBL tmp;
+
+    tmp = scaleValue(*pSignal, shift + signal_e) -
+          fMultDiv2(*pCoeff--, *pState--);
+    for (j = order - 1; j != 0; j--) {
+      tmp = fMultSubDiv2(tmp, pCoeff[0], pState[0]);
+      pState[1] = pState[0] + (fMultDiv2(*pCoeff--, tmp) << 2);
+      pState--;
+    }
+
+    *pSignal = scaleValueSaturate(tmp, -shift - signal_e_out);
+
+    /* exponent of state[] is -1 */
+    pState[1] = tmp << 1;
+    pSignal += inc;
+  }
+}
+#endif
+
+#ifndef FUNCTION_CLpc_SynthesisLattice_DBL
+void CLpc_SynthesisLattice(FIXP_DBL *signal, const int signal_size,
+                           const int signal_e, const int signal_e_out,
+                           const int inc, const FIXP_DBL *coeff,
+                           const int order, FIXP_DBL *state) {
+  int i, j;
+  FIXP_DBL *pSignal;
+
+  FDK_ASSERT(order <= LPC_MAX_ORDER);
+  FDK_ASSERT(order > 0);
+
+  if (inc == -1)
+    pSignal = &signal[signal_size - 1];
+  else
+    pSignal = &signal[0];
+
+  FDK_ASSERT(signal_size > 0);
+  for (i = signal_size; i != 0; i--) {
+    FIXP_DBL *pState = state + order - 1;
+    const FIXP_DBL *pCoeff = coeff + order - 1;
+    FIXP_DBL tmp, accu;
+
+    accu =
+        fMultSubDiv2(scaleValue(*pSignal, signal_e - 1), *pCoeff--, *pState--);
+    tmp = SATURATE_LEFT_SHIFT_ALT(accu, 1, DFRACT_BITS);
+
+    for (j = order - 1; j != 0; j--) {
+      accu = fMultSubDiv2(tmp >> 1, pCoeff[0], pState[0]);
+      tmp = SATURATE_LEFT_SHIFT_ALT(accu, 1, DFRACT_BITS);
+
+      accu = fMultAddDiv2(pState[0] >> 1, *pCoeff--, tmp);
+      pState[1] = SATURATE_LEFT_SHIFT_ALT(accu, 1, DFRACT_BITS);
+
+      pState--;
+    }
+
+    *pSignal = scaleValue(tmp, -signal_e_out);
+
+    /* exponent of state[] is 0 */
+    pState[1] = tmp;
+    pSignal += inc;
+  }
+}
+
+#endif
+
+/* LPC_SYNTHESIS_IIR version */
+void CLpc_Synthesis(FIXP_DBL *signal, const int signal_size, const int signal_e,
+                    const int inc, const FIXP_LPC_TNS *lpcCoeff_m,
+                    const int lpcCoeff_e, const int order, FIXP_DBL *state,
+                    int *pStateIndex) {
+  int i, j;
+  FIXP_DBL *pSignal;
+  int stateIndex = *pStateIndex;
+
+  FIXP_LPC_TNS coeff[2 * LPC_MAX_ORDER];
+  FDKmemcpy(&coeff[0], lpcCoeff_m, order * sizeof(FIXP_LPC_TNS));
+  FDKmemcpy(&coeff[order], lpcCoeff_m, order * sizeof(FIXP_LPC_TNS));
+
+  FDK_ASSERT(order <= LPC_MAX_ORDER);
+  FDK_ASSERT(stateIndex < order);
+
+  if (inc == -1)
+    pSignal = &signal[signal_size - 1];
+  else
+    pSignal = &signal[0];
+
+  /* y(n) = x(n) - lpc[1]*y(n-1) - ... - lpc[order]*y(n-order) */
+
+  for (i = 0; i < signal_size; i++) {
+    FIXP_DBL x;
+    const FIXP_LPC_TNS *pCoeff = coeff + order - stateIndex;
+
+    x = scaleValue(*pSignal, -(lpcCoeff_e + 1));
+    for (j = 0; j < order; j++) {
+      x -= fMultDiv2(state[j], pCoeff[j]);
+    }
+    x = SATURATE_SHIFT(x, -lpcCoeff_e - 1, DFRACT_BITS);
+
+    /* Update states */
+    stateIndex = ((stateIndex - 1) < 0) ? (order - 1) : (stateIndex - 1);
+    state[stateIndex] = x;
+
+    *pSignal = scaleValue(x, signal_e);
+    pSignal += inc;
+  }
+
+  *pStateIndex = stateIndex;
+}
+/* default version */
+void CLpc_Synthesis(FIXP_DBL *signal, const int signal_size, const int signal_e,
+                    const int inc, const FIXP_LPC *lpcCoeff_m,
+                    const int lpcCoeff_e, const int order, FIXP_DBL *state,
+                    int *pStateIndex) {
+  int i, j;
+  FIXP_DBL *pSignal;
+  int stateIndex = *pStateIndex;
+
+  FIXP_LPC coeff[2 * LPC_MAX_ORDER];
+  FDKmemcpy(&coeff[0], lpcCoeff_m, order * sizeof(FIXP_LPC));
+  FDKmemcpy(&coeff[order], lpcCoeff_m, order * sizeof(FIXP_LPC));
+
+  FDK_ASSERT(order <= LPC_MAX_ORDER);
+  FDK_ASSERT(stateIndex < order);
+
+  if (inc == -1)
+    pSignal = &signal[signal_size - 1];
+  else
+    pSignal = &signal[0];
+
+  /* y(n) = x(n) - lpc[1]*y(n-1) - ... - lpc[order]*y(n-order) */
+
+  for (i = 0; i < signal_size; i++) {
+    FIXP_DBL x;
+    const FIXP_LPC *pCoeff = coeff + order - stateIndex;
+
+    x = scaleValue(*pSignal, -(lpcCoeff_e + 1));
+    for (j = 0; j < order; j++) {
+      x -= fMultDiv2(state[j], pCoeff[j]);
+    }
+    x = SATURATE_SHIFT(x, -lpcCoeff_e - 1, DFRACT_BITS);
+
+    /* Update states */
+    stateIndex = ((stateIndex - 1) < 0) ? (order - 1) : (stateIndex - 1);
+    state[stateIndex] = x;
+
+    *pSignal = scaleValue(x, signal_e);
+    pSignal += inc;
+  }
+
+  *pStateIndex = stateIndex;
+}
+
+/* FIR */
+void CLpc_Analysis(FIXP_DBL *RESTRICT signal, const int signal_size,
+                   const FIXP_LPC lpcCoeff_m[], const int lpcCoeff_e,
+                   const int order, FIXP_DBL *RESTRICT filtState,
+                   int *filtStateIndex) {
+  int stateIndex;
+  INT i, j, shift = lpcCoeff_e + 1; /* +1, because fMultDiv2 */
+  FIXP_DBL tmp;
+
+  if (order <= 0) {
+    return;
+  }
+  if (filtStateIndex != NULL) {
+    stateIndex = *filtStateIndex;
+  } else {
+    stateIndex = 0;
+  }
+
+  /* keep filter coefficients twice and save memory copy operation in
+     modulo state buffer */
+  FIXP_LPC coeff[2 * LPC_MAX_ORDER];
+  FIXP_LPC *pCoeff;
+  FDKmemcpy(&coeff[0], lpcCoeff_m, order * sizeof(FIXP_LPC));
+  FDKmemcpy(&coeff[order], lpcCoeff_m, order * sizeof(FIXP_LPC));
+
+  /*
+      # Analysis filter, obtain residual.
+      for k = 0:BL-1
+              err(i-BL+k) = a * inputSignal(i-BL+k:-1:i-BL-M+k);
+      endfor
+   */
+
+  FDK_ASSERT(shift >= 0);
+
+  for (j = 0; j < signal_size; j++) {
+    pCoeff = &coeff[(order - stateIndex)];
+
+    tmp = signal[j] >> shift;
+    for (i = 0; i < order; i++) {
+      tmp = fMultAddDiv2(tmp, pCoeff[i], filtState[i]);
+    }
+
+    stateIndex =
+        ((stateIndex - 1) < 0) ? (stateIndex - 1 + order) : (stateIndex - 1);
+    filtState[stateIndex] = signal[j];
+
+    signal[j] = tmp << shift;
+  }
+
+  if (filtStateIndex != NULL) {
+    *filtStateIndex = stateIndex;
+  }
+}
+
+/* For the LPC_SYNTHESIS_IIR version */
+INT CLpc_ParcorToLpc(const FIXP_LPC_TNS reflCoeff[], FIXP_LPC_TNS LpcCoeff[],
+                     INT numOfCoeff, FIXP_DBL workBuffer[]) {
+  INT i, j;
+  INT shiftval,
+      par2LpcShiftVal = 6; /* 6 should be enough, bec. max(numOfCoeff) = 20 */
+  FIXP_DBL maxVal = (FIXP_DBL)0;
+
+  workBuffer[0] = FX_LPC_TNS2FX_DBL(reflCoeff[0]) >> par2LpcShiftVal;
+  for (i = 1; i < numOfCoeff; i++) {
+    for (j = 0; j < i / 2; j++) {
+      FIXP_DBL tmp1, tmp2;
+
+      tmp1 = workBuffer[j];
+      tmp2 = workBuffer[i - 1 - j];
+      workBuffer[j] += fMult(reflCoeff[i], tmp2);
+      workBuffer[i - 1 - j] += fMult(reflCoeff[i], tmp1);
+    }
+    if (i & 1) {
+      workBuffer[j] += fMult(reflCoeff[i], workBuffer[j]);
+    }
+
+    workBuffer[i] = FX_LPC_TNS2FX_DBL(reflCoeff[i]) >> par2LpcShiftVal;
+  }
+
+  /* calculate exponent */
+  for (i = 0; i < numOfCoeff; i++) {
+    maxVal = fMax(maxVal, fAbs(workBuffer[i]));
+  }
+
+  shiftval = fMin(fNorm(maxVal), par2LpcShiftVal);
+
+  for (i = 0; i < numOfCoeff; i++) {
+    LpcCoeff[i] = FX_DBL2FX_LPC_TNS(workBuffer[i] << shiftval);
+  }
+
+  return (par2LpcShiftVal - shiftval);
+}
+/* Default version */
+INT CLpc_ParcorToLpc(const FIXP_LPC reflCoeff[], FIXP_LPC LpcCoeff[],
+                     INT numOfCoeff, FIXP_DBL workBuffer[]) {
+  INT i, j;
+  INT shiftval,
+      par2LpcShiftVal = 6; /* 6 should be enough, bec. max(numOfCoeff) = 20 */
+  FIXP_DBL maxVal = (FIXP_DBL)0;
+
+  workBuffer[0] = FX_LPC2FX_DBL(reflCoeff[0]) >> par2LpcShiftVal;
+  for (i = 1; i < numOfCoeff; i++) {
+    for (j = 0; j < i / 2; j++) {
+      FIXP_DBL tmp1, tmp2;
+
+      tmp1 = workBuffer[j];
+      tmp2 = workBuffer[i - 1 - j];
+      workBuffer[j] += fMult(reflCoeff[i], tmp2);
+      workBuffer[i - 1 - j] += fMult(reflCoeff[i], tmp1);
+    }
+    if (i & 1) {
+      workBuffer[j] += fMult(reflCoeff[i], workBuffer[j]);
+    }
+
+    workBuffer[i] = FX_LPC2FX_DBL(reflCoeff[i]) >> par2LpcShiftVal;
+  }
+
+  /* calculate exponent */
+  for (i = 0; i < numOfCoeff; i++) {
+    maxVal = fMax(maxVal, fAbs(workBuffer[i]));
+  }
+
+  shiftval = fMin(fNorm(maxVal), par2LpcShiftVal);
+
+  for (i = 0; i < numOfCoeff; i++) {
+    LpcCoeff[i] = FX_DBL2FX_LPC(workBuffer[i] << shiftval);
+  }
+
+  return (par2LpcShiftVal - shiftval);
+}
+
+void CLpc_AutoToParcor(FIXP_DBL acorr[], const int acorr_e,
+                       FIXP_LPC reflCoeff[], const int numOfCoeff,
+                       FIXP_DBL *pPredictionGain_m, INT *pPredictionGain_e) {
+  INT i, j, scale = 0;
+  FIXP_DBL parcorWorkBuffer[LPC_MAX_ORDER];
+
+  FIXP_DBL *workBuffer = parcorWorkBuffer;
+  FIXP_DBL autoCorr_0 = acorr[0];
+
+  FDKmemclear(reflCoeff, numOfCoeff * sizeof(FIXP_LPC));
+
+  if (autoCorr_0 == FL2FXCONST_DBL(0.0)) {
+    if (pPredictionGain_m != NULL) {
+      *pPredictionGain_m = FL2FXCONST_DBL(0.5f);
+      *pPredictionGain_e = 1;
+    }
+    return;
+  }
+
+  FDKmemcpy(workBuffer, acorr + 1, numOfCoeff * sizeof(FIXP_DBL));
+  for (i = 0; i < numOfCoeff; i++) {
+    LONG sign = ((LONG)workBuffer[0] >> (DFRACT_BITS - 1));
+    FIXP_DBL tmp = (FIXP_DBL)((LONG)workBuffer[0] ^ sign);
+
+    /* Check preconditions for division function: num<=denum             */
+    /* For 1st iteration acorr[0] cannot be 0, it is checked before loop */
+    /* Due to exor operation with "sign", num(=tmp) is greater/equal 0   */
+    if (acorr[0] < tmp) break;
+
+    /* tmp = div(num, denum, 16) */
+    tmp = (FIXP_DBL)((LONG)schur_div(tmp, acorr[0], FRACT_BITS) ^ (~sign));
+
+    reflCoeff[i] = FX_DBL2FX_LPC(tmp);
+
+    for (j = numOfCoeff - i - 1; j >= 0; j--) {
+      FIXP_DBL accu1 = fMult(tmp, acorr[j]);
+      FIXP_DBL accu2 = fMult(tmp, workBuffer[j]);
+      workBuffer[j] += accu1;
+      acorr[j] += accu2;
+    }
+    /* Check preconditions for division function: denum (=acorr[0]) > 0 */
+    if (acorr[0] == (FIXP_DBL)0) break;
+
+    workBuffer++;
+  }
+
+  if (pPredictionGain_m != NULL) {
+    if (acorr[0] > (FIXP_DBL)0) {
+      /* prediction gain = signal power / error (residual) power */
+      *pPredictionGain_m = fDivNormSigned(autoCorr_0, acorr[0], &scale);
+      *pPredictionGain_e = scale;
+    } else {
+      *pPredictionGain_m = (FIXP_DBL)0;
+      *pPredictionGain_e = 0;
+    }
+  }
+}