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, 276 insertions, 0 deletions

diff --git a/fdk-aac/libAACdec/src/ldfiltbank.cpp b/fdk-aac/libAACdec/src/ldfiltbank.cpp
new file mode 100644
index 0000000..c7d2928
--- /dev/null
+++ b/fdk-aac/libAACdec/src/ldfiltbank.cpp
@@ -0,0 +1,276 @@
+/* -----------------------------------------------------------------------------
+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
+----------------------------------------------------------------------------- */
+
+/**************************** AAC decoder library ******************************
+
+   Author(s):
+
+   Description: low delay filterbank
+
+*******************************************************************************/
+
+#include "ldfiltbank.h"
+
+#include "aac_rom.h"
+#include "dct.h"
+#include "FDK_tools_rom.h"
+#include "mdct.h"
+
+#define LDFB_HEADROOM 2
+
+#if defined(__arm__)
+#endif
+
+static void multE2_DinvF_fdk(FIXP_PCM *output, FIXP_DBL *x, const FIXP_WTB *fb,
+                             FIXP_DBL *z, const int N) {
+  int i;
+
+  /*  scale for FIXP_DBL -> INT_PCM conversion. */
+  const int scale = (DFRACT_BITS - SAMPLE_BITS) - LDFB_HEADROOM;
+#if ((DFRACT_BITS - SAMPLE_BITS - LDFB_HEADROOM) > 0)
+  FIXP_DBL rnd_val_wts0 = (FIXP_DBL)0;
+  FIXP_DBL rnd_val_wts1 = (FIXP_DBL)0;
+  if (-WTS0 - 1 + scale)
+    rnd_val_wts0 = (FIXP_DBL)(1 << (-WTS0 - 1 + scale - 1));
+  if (-WTS1 - 1 + scale)
+    rnd_val_wts1 = (FIXP_DBL)(1 << (-WTS1 - 1 + scale - 1));
+#endif
+
+  for (i = 0; i < N / 4; i++) {
+    FIXP_DBL z0, z2, tmp;
+
+    z2 = x[N / 2 + i];
+    z0 = z2 + (fMultDiv2(z[N / 2 + i], fb[2 * N + i]) >> (-WTS2 - 1));
+
+    z[N / 2 + i] = x[N / 2 - 1 - i] +
+                   (fMultDiv2(z[N + i], fb[2 * N + N / 2 + i]) >> (-WTS2 - 1));
+
+    tmp = (fMultDiv2(z[N / 2 + i], fb[N + N / 2 - 1 - i]) +
+           fMultDiv2(z[i], fb[N + N / 2 + i]));
+
+#if ((DFRACT_BITS - SAMPLE_BITS - LDFB_HEADROOM) > 0)
+    FDK_ASSERT((-WTS1 - 1 + scale) >= 0);
+    FDK_ASSERT(tmp <= ((FIXP_DBL)0x7FFFFFFF -
+                       rnd_val_wts1)); /* rounding must not cause overflow */
+    output[(N * 3 / 4 - 1 - i)] = (FIXP_PCM)SATURATE_RIGHT_SHIFT(
+        tmp + rnd_val_wts1, -WTS1 - 1 + scale, PCM_OUT_BITS);
+#else
+    FDK_ASSERT((WTS1 + 1 - scale) >= 0);
+    output[(N * 3 / 4 - 1 - i)] =
+        (FIXP_PCM)SATURATE_LEFT_SHIFT(tmp, WTS1 + 1 - scale, PCM_OUT_BITS);
+#endif
+
+    z[i] = z0;
+    z[N + i] = z2;
+  }
+
+  for (i = N / 4; i < N / 2; i++) {
+    FIXP_DBL z0, z2, tmp0, tmp1;
+
+    z2 = x[N / 2 + i];
+    z0 = z2 + (fMultDiv2(z[N / 2 + i], fb[2 * N + i]) >> (-WTS2 - 1));
+
+    z[N / 2 + i] = x[N / 2 - 1 - i] +
+                   (fMultDiv2(z[N + i], fb[2 * N + N / 2 + i]) >> (-WTS2 - 1));
+
+    tmp0 = (fMultDiv2(z[N / 2 + i], fb[N / 2 - 1 - i]) +
+            fMultDiv2(z[i], fb[N / 2 + i]));
+    tmp1 = (fMultDiv2(z[N / 2 + i], fb[N + N / 2 - 1 - i]) +
+            fMultDiv2(z[i], fb[N + N / 2 + i]));
+
+#if ((DFRACT_BITS - SAMPLE_BITS - LDFB_HEADROOM) > 0)
+    FDK_ASSERT((-WTS0 - 1 + scale) >= 0);
+    FDK_ASSERT(tmp0 <= ((FIXP_DBL)0x7FFFFFFF -
+                        rnd_val_wts0)); /* rounding must not cause overflow */
+    FDK_ASSERT(tmp1 <= ((FIXP_DBL)0x7FFFFFFF -
+                        rnd_val_wts1)); /* rounding must not cause overflow */
+    output[(i - N / 4)] = (FIXP_PCM)SATURATE_RIGHT_SHIFT(
+        tmp0 + rnd_val_wts0, -WTS0 - 1 + scale, PCM_OUT_BITS);
+    output[(N * 3 / 4 - 1 - i)] = (FIXP_PCM)SATURATE_RIGHT_SHIFT(
+        tmp1 + rnd_val_wts1, -WTS1 - 1 + scale, PCM_OUT_BITS);
+#else
+    FDK_ASSERT((WTS0 + 1 - scale) >= 0);
+    output[(i - N / 4)] =
+        (FIXP_PCM)SATURATE_LEFT_SHIFT(tmp0, WTS0 + 1 - scale, PCM_OUT_BITS);
+    output[(N * 3 / 4 - 1 - i)] =
+        (FIXP_PCM)SATURATE_LEFT_SHIFT(tmp1, WTS1 + 1 - scale, PCM_OUT_BITS);
+#endif
+    z[i] = z0;
+    z[N + i] = z2;
+  }
+
+  /* Exchange quarter parts of x to bring them in the "right" order */
+  for (i = 0; i < N / 4; i++) {
+    FIXP_DBL tmp0 = fMultDiv2(z[i], fb[N / 2 + i]);
+
+#if ((DFRACT_BITS - SAMPLE_BITS - LDFB_HEADROOM) > 0)
+    FDK_ASSERT((-WTS0 - 1 + scale) >= 0);
+    FDK_ASSERT(tmp0 <= ((FIXP_DBL)0x7FFFFFFF -
+                        rnd_val_wts0)); /* rounding must not cause overflow */
+    output[(N * 3 / 4 + i)] = (FIXP_PCM)SATURATE_RIGHT_SHIFT(
+        tmp0 + rnd_val_wts0, -WTS0 - 1 + scale, PCM_OUT_BITS);
+#else
+    FDK_ASSERT((WTS0 + 1 - scale) >= 0);
+    output[(N * 3 / 4 + i)] =
+        (FIXP_PCM)SATURATE_LEFT_SHIFT(tmp0, WTS0 + 1 - scale, PCM_OUT_BITS);
+#endif
+  }
+}
+
+int InvMdctTransformLowDelay_fdk(FIXP_DBL *mdctData, const int mdctData_e,
+                                 FIXP_PCM *output, FIXP_DBL *fs_buffer,
+                                 const int N) {
+  const FIXP_WTB *coef;
+  FIXP_DBL gain = (FIXP_DBL)0;
+  int scale = mdctData_e + MDCT_OUT_HEADROOM -
+              LDFB_HEADROOM; /* The LDFB_HEADROOM is compensated inside
+                                multE2_DinvF_fdk() below */
+  int i;
+
+  /* Select LD window slope */
+  switch (N) {
+    case 256:
+      coef = LowDelaySynthesis256;
+      break;
+    case 240:
+      coef = LowDelaySynthesis240;
+      break;
+    case 160:
+      coef = LowDelaySynthesis160;
+      break;
+    case 128:
+      coef = LowDelaySynthesis128;
+      break;
+    case 120:
+      coef = LowDelaySynthesis120;
+      break;
+    case 512:
+      coef = LowDelaySynthesis512;
+      break;
+    case 480:
+    default:
+      coef = LowDelaySynthesis480;
+      break;
+  }
+
+  /*
+     Apply exponent and 1/N factor.
+     Note: "scale" is off by one because for LD_MDCT the window length is twice
+     the window length of a regular MDCT. This is corrected inside
+     multE2_DinvF_fdk(). Refer to ISO/IEC 14496-3:2009 page 277,
+     chapter 4.6.20.2 "Low Delay Window".
+   */
+  imdct_gain(&gain, &scale, N);
+
+  dct_IV(mdctData, N, &scale);
+
+  if (N == 256 || N == 240 || N == 160) {
+    scale -= 1;
+  } else if (N == 128 || N == 120) {
+    scale -= 2;
+  }
+
+  if (gain != (FIXP_DBL)0) {
+    for (i = 0; i < N; i++) {
+      mdctData[i] = fMult(mdctData[i], gain);
+    }
+  }
+  scaleValuesSaturate(mdctData, N, scale);
+
+  /* Since all exponent and factors have been applied, current exponent is zero.
+   */
+  multE2_DinvF_fdk(output, mdctData, coef, fs_buffer, N);
+
+  return (1);
+}