text: /*
   * AAC decoder
   * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
   * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
   * Copyright (c) 2008-2013 Alex Converse <alex.converse@gmail.com>
   *
   * AAC LATM decoder
   * Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
   * Copyright (c) 2010      Janne Grunau <janne-libav@jannau.net>
   *
   * This file is part of FFmpeg.
   *
   * FFmpeg is free software; you can redistribute it and/or
   * modify it under the terms of the GNU Lesser General Public
   * License as published by the Free Software Foundation; either
   * version 2.1 of the License, or (at your option) any later version.
   *
   * FFmpeg is distributed in the hope that it will be useful,
   * but WITHOUT ANY WARRANTY; without even the implied warranty of
   * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
   * Lesser General Public License for more details.
   *
   * You should have received a copy of the GNU Lesser General Public
   * License along with FFmpeg; if not, write to the Free Software
   * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
   */
  
  /**
   * @file
   * AAC decoder
   * @author Oded Shimon  ( ods15 ods15 dyndns org )
   * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
   */
  
  #define FFT_FLOAT 1
  #define FFT_FIXED_32 0
  #define USE_FIXED 0
  
  #include "libavutil/float_dsp.h"
  #include "libavutil/opt.h"
  #include "avcodec.h"
  #include "internal.h"
  #include "get_bits.h"
  #include "fft.h"
  #include "mdct15.h"
  #include "lpc.h"
  #include "kbdwin.h"
  #include "sinewin.h"
  
  #include "aac.h"
  #include "aactab.h"
  #include "aacdectab.h"
  #include "adts_header.h"
  #include "cbrt_data.h"
  #include "sbr.h"
  #include "aacsbr.h"
  #include "mpeg4audio.h"
  #include "profiles.h"
  #include "libavutil/intfloat.h"
  
  #include <errno.h>
  #include <math.h>
  #include <stdint.h>
  #include <string.h>
  
  #if ARCH_ARM
  #   include "arm/aac.h"
  #elif ARCH_MIPS
  #   include "mips/aacdec_mips.h"
  #endif
  
  static av_always_inline void reset_predict_state(PredictorState *ps)
  {
      ps->r0   = 0.0f;
      ps->r1   = 0.0f;
      ps->cor0 = 0.0f;
      ps->cor1 = 0.0f;
      ps->var0 = 1.0f;
      ps->var1 = 1.0f;
  }
  
  #ifndef VMUL2
  static inline float *VMUL2(float *dst, const float *v, unsigned idx,
                             const float *scale)
  {
      float s = *scale;
      *dst++ = v[idx    & 15] * s;
      *dst++ = v[idx>>4 & 15] * s;
      return dst;
  }
  #endif
  
  #ifndef VMUL4
  static inline float *VMUL4(float *dst, const float *v, unsigned idx,
                             const float *scale)
  {
      float s = *scale;
      *dst++ = v[idx    & 3] * s;
      *dst++ = v[idx>>2 & 3] * s;
      *dst++ = v[idx>>4 & 3] * s;
      *dst++ = v[idx>>6 & 3] * s;
      return dst;
  }
  #endif
  
  #ifndef VMUL2S
  static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
                              unsigned sign, const float *scale)
  {
      union av_intfloat32 s0, s1;
  
      s0.f = s1.f = *scale;
      s0.i ^= sign >> 1 << 31;
      s1.i ^= sign      << 31;
  
      *dst++ = v[idx    & 15] * s0.f;
      *dst++ = v[idx>>4 & 15] * s1.f;
  
      return dst;
  }
  #endif
  
  #ifndef VMUL4S
  static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
                              unsigned sign, const float *scale)
  {
      unsigned nz = idx >> 12;
      union av_intfloat32 s = { .f = *scale };
      union av_intfloat32 t;
  
      t.i = s.i ^ (sign & 1U<<31);
      *dst++ = v[idx    & 3] * t.f;
  
      sign <<= nz & 1; nz >>= 1;
      t.i = s.i ^ (sign & 1U<<31);
      *dst++ = v[idx>>2 & 3] * t.f;
  
      sign <<= nz & 1; nz >>= 1;
      t.i = s.i ^ (sign & 1U<<31);
      *dst++ = v[idx>>4 & 3] * t.f;
  
      sign <<= nz & 1;
      t.i = s.i ^ (sign & 1U<<31);
      *dst++ = v[idx>>6 & 3] * t.f;
  
      return dst;
  }
  #endif
  
  static av_always_inline float flt16_round(float pf)
  {
      union av_intfloat32 tmp;
      tmp.f = pf;
      tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
      return tmp.f;
  }
  
  static av_always_inline float flt16_even(float pf)
  {
      union av_intfloat32 tmp;
      tmp.f = pf;
      tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
      return tmp.f;
  }
  
  static av_always_inline float flt16_trunc(float pf)
  {
      union av_intfloat32 pun;
      pun.f = pf;
      pun.i &= 0xFFFF0000U;
      return pun.f;
  }
  
  static av_always_inline void predict(PredictorState *ps, float *coef,
                                       int output_enable)
  {
      const float a     = 0.953125; // 61.0 / 64
      const float alpha = 0.90625;  // 29.0 / 32
      float e0, e1;
      float pv;
      float k1, k2;
      float   r0 = ps->r0,     r1 = ps->r1;
      float cor0 = ps->cor0, cor1 = ps->cor1;
      float var0 = ps->var0, var1 = ps->var1;
  
      k1 = var0 > 1 ? cor0 * flt16_even(a / var0) : 0;
      k2 = var1 > 1 ? cor1 * flt16_even(a / var1) : 0;
  
      pv = flt16_round(k1 * r0 + k2 * r1);
      if (output_enable)
          *coef += pv;
  
      e0 = *coef;
      e1 = e0 - k1 * r0;
  
      ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
      ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1));
      ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0);
      ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0));
  
      ps->r1 = flt16_trunc(a * (r0 - k1 * e0));
      ps->r0 = flt16_trunc(a * e0);
  }
  
  /**
   * Apply dependent channel coupling (applied before IMDCT).
   *
   * @param   index   index into coupling gain array
   */
  static void apply_dependent_coupling(AACContext *ac,
                                       SingleChannelElement *target,
                                       ChannelElement *cce, int index)
  {
      IndividualChannelStream *ics = &cce->ch[0].ics;
      const uint16_t *offsets = ics->swb_offset;
      float *dest = target->coeffs;
      const float *src = cce->ch[0].coeffs;
      int g, i, group, k, idx = 0;
      if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
          av_log(ac->avctx, AV_LOG_ERROR,
                 "Dependent coupling is not supported together with LTP\n");
          return;
      }
      for (g = 0; g < ics->num_window_groups; g++) {
          for (i = 0; i < ics->max_sfb; i++, idx++) {
              if (cce->ch[0].band_type[idx] != ZERO_BT) {
                  const float gain = cce->coup.gain[index][idx];
                  for (group = 0; group < ics->group_len[g]; group++) {
                      for (k = offsets[i]; k < offsets[i + 1]; k++) {
                          // FIXME: SIMDify
                          dest[group * 128 + k] += gain * src[group * 128 + k];
                      }
                  }
              }
          }
          dest += ics->group_len[g] * 128;
          src  += ics->group_len[g] * 128;
      }
  }
  
  /**
   * Apply independent channel coupling (applied after IMDCT).
   *
   * @param   index   index into coupling gain array
   */
  static void apply_independent_coupling(AACContext *ac,
                                         SingleChannelElement *target,
                                         ChannelElement *cce, int index)
  {
      const float gain = cce->coup.gain[index][0];
      const float *src = cce->ch[0].ret;
      float *dest = target->ret;
      const int len = 1024 << (ac->oc[1].m4ac.sbr == 1);
  
      ac->fdsp->vector_fmac_scalar(dest, src, gain, len);
  }
  
  #include "aacdec_template.c"
  
  #define LOAS_SYNC_WORD   0x2b7       ///< 11 bits LOAS sync word
  
  struct LATMContext {
      AACContext aac_ctx;     ///< containing AACContext
      int initialized;        ///< initialized after a valid extradata was seen
  
      // parser data
      int audio_mux_version_A; ///< LATM syntax version
      int frame_length_type;   ///< 0/1 variable/fixed frame length
      int frame_length;        ///< frame length for fixed frame length
  };
  
  static inline uint32_t latm_get_value(GetBitContext *b)
  {
      int length = get_bits(b, 2);
  
      return get_bits_long(b, (length+1)*8);
  }
  
  static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
                                               GetBitContext *gb, int asclen)
  {
      AACContext *ac        = &latmctx->aac_ctx;
      AVCodecContext *avctx = ac->avctx;
      MPEG4AudioConfig m4ac = { 0 };
      GetBitContext gbc;
      int config_start_bit  = get_bits_count(gb);
      int sync_extension    = 0;
      int bits_consumed, esize, i;
  
      if (asclen > 0) {
          sync_extension = 1;
          asclen         = FFMIN(asclen, get_bits_left(gb));
          init_get_bits(&gbc, gb->buffer, config_start_bit + asclen);
          skip_bits_long(&gbc, config_start_bit);
      } else if (asclen == 0) {
          gbc = *gb;
      } else {
          return AVERROR_INVALIDDATA;
      }
  
      if (get_bits_left(gb) <= 0)
          return AVERROR_INVALIDDATA;
  
      bits_consumed = decode_audio_specific_config_gb(NULL, avctx, &m4ac,
                                                      &gbc, config_start_bit,
                                                      sync_extension);
  
      if (bits_consumed < config_start_bit)
          return AVERROR_INVALIDDATA;
      bits_consumed -= config_start_bit;
  
      if (asclen == 0)
        asclen = bits_consumed;
  
      if (!latmctx->initialized ||
          ac->oc[1].m4ac.sample_rate != m4ac.sample_rate ||
          ac->oc[1].m4ac.chan_config != m4ac.chan_config) {
  
          if (latmctx->initialized) {
              av_log(avctx, AV_LOG_INFO, "audio config changed (sample_rate=%d, chan_config=%d)\n", m4ac.sample_rate, m4ac.chan_config);
          } else {
              av_log(avctx, AV_LOG_DEBUG, "initializing latmctx\n");
          }
          latmctx->initialized = 0;
  
          esize = (asclen + 7) / 8;
  
          if (avctx->extradata_size < esize) {
              av_free(avctx->extradata);
              avctx->extradata = av_malloc(esize + AV_INPUT_BUFFER_PADDING_SIZE);
              if (!avctx->extradata)
                  return AVERROR(ENOMEM);
          }
  
          avctx->extradata_size = esize;
          gbc = *gb;
          for (i = 0; i < esize; i++) {
            avctx->extradata[i] = get_bits(&gbc, 8);
          }
          memset(avctx->extradata+esize, 0, AV_INPUT_BUFFER_PADDING_SIZE);
      }
      skip_bits_long(gb, asclen);
  
      return 0;
  }
  
  static int read_stream_mux_config(struct LATMContext *latmctx,
                                    GetBitContext *gb)
  {
      int ret, audio_mux_version = get_bits(gb, 1);
  
      latmctx->audio_mux_version_A = 0;
      if (audio_mux_version)
          latmctx->audio_mux_version_A = get_bits(gb, 1);
  
      if (!latmctx->audio_mux_version_A) {
  
          if (audio_mux_version)
              latm_get_value(gb);                 // taraFullness
  
          skip_bits(gb, 1);                       // allStreamSameTimeFraming
          skip_bits(gb, 6);                       // numSubFrames
          // numPrograms
          if (get_bits(gb, 4)) {                  // numPrograms
              avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple programs");
              return AVERROR_PATCHWELCOME;
          }
  
          // for each program (which there is only one in DVB)
  
          // for each layer (which there is only one in DVB)
          if (get_bits(gb, 3)) {                   // numLayer
              avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple layers");
              return AVERROR_PATCHWELCOME;
          }
  
          // for all but first stream: use_same_config = get_bits(gb, 1);
          if (!audio_mux_version) {
              if ((ret = latm_decode_audio_specific_config(latmctx, gb, 0)) < 0)
                  return ret;
          } else {
              int ascLen = latm_get_value(gb);
              if ((ret = latm_decode_audio_specific_config(latmctx, gb, ascLen)) < 0)
                  return ret;
          }
  
          latmctx->frame_length_type = get_bits(gb, 3);
          switch (latmctx->frame_length_type) {
          case 0:
              skip_bits(gb, 8);       // latmBufferFullness
              break;
          case 1:
              latmctx->frame_length = get_bits(gb, 9);
              break;
          case 3:
          case 4:
          case 5:
              skip_bits(gb, 6);       // CELP frame length table index
              break;
          case 6:
          case 7:
              skip_bits(gb, 1);       // HVXC frame length table index
              break;
          }
  
          if (get_bits(gb, 1)) {                  // other data
              if (audio_mux_version) {
                  latm_get_value(gb);             // other_data_bits
              } else {
                  int esc;
                  do {
                      if (get_bits_left(gb) < 9)
                          return AVERROR_INVALIDDATA;
                      esc = get_bits(gb, 1);
                      skip_bits(gb, 8);
                  } while (esc);
              }
          }
  
          if (get_bits(gb, 1))                     // crc present
              skip_bits(gb, 8);                    // config_crc
      }
  
      return 0;
  }
  
  static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb)
  {
      uint8_t tmp;
  
      if (ctx->frame_length_type == 0) {
          int mux_slot_length = 0;
          do {
              if (get_bits_left(gb) < 8)
                  return AVERROR_INVALIDDATA;
              tmp = get_bits(gb, 8);
              mux_slot_length += tmp;
          } while (tmp == 255);
          return mux_slot_length;
      } else if (ctx->frame_length_type == 1) {
          return ctx->frame_length;
      } else if (ctx->frame_length_type == 3 ||
                 ctx->frame_length_type == 5 ||
                 ctx->frame_length_type == 7) {
          skip_bits(gb, 2);          // mux_slot_length_coded
      }
      return 0;
  }
  
  static int read_audio_mux_element(struct LATMContext *latmctx,
                                    GetBitContext *gb)
  {
      int err;
      uint8_t use_same_mux = get_bits(gb, 1);
      if (!use_same_mux) {
          if ((err = read_stream_mux_config(latmctx, gb)) < 0)
              return err;
      } else if (!latmctx->aac_ctx.avctx->extradata) {
          av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG,
                 "no decoder config found\n");
          return 1;
      }
      if (latmctx->audio_mux_version_A == 0) {
          int mux_slot_length_bytes = read_payload_length_info(latmctx, gb);
          if (mux_slot_length_bytes < 0 || mux_slot_length_bytes * 8LL > get_bits_left(gb)) {
              av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n");
              return AVERROR_INVALIDDATA;
          } else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) {
              av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
                     "frame length mismatch %d << %d\n",
                     mux_slot_length_bytes * 8, get_bits_left(gb));
              return AVERROR_INVALIDDATA;
          }
      }
      return 0;
  }
  
  
  static int latm_decode_frame(AVCodecContext *avctx, void *out,
                               int *got_frame_ptr, AVPacket *avpkt)
  {
      struct LATMContext *latmctx = avctx->priv_data;
      int                 muxlength, err;
      GetBitContext       gb;
  
      if ((err = init_get_bits8(&gb, avpkt->data, avpkt->size)) < 0)
          return err;
  
      // check for LOAS sync word
      if (get_bits(&gb, 11) != LOAS_SYNC_WORD)
          return AVERROR_INVALIDDATA;
  
      muxlength = get_bits(&gb, 13) + 3;
      // not enough data, the parser should have sorted this out
      if (muxlength > avpkt->size)
          return AVERROR_INVALIDDATA;
  
      if ((err = read_audio_mux_element(latmctx, &gb)))
          return (err < 0) ? err : avpkt->size;
  
      if (!latmctx->initialized) {
          if (!avctx->extradata) {
              *got_frame_ptr = 0;
              return avpkt->size;
          } else {
              push_output_configuration(&latmctx->aac_ctx);
              if ((err = decode_audio_specific_config(
                      &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1].m4ac,
                      avctx->extradata, avctx->extradata_size*8LL, 1)) < 0) {
                  pop_output_configuration(&latmctx->aac_ctx);
                  return err;
              }
              latmctx->initialized = 1;
          }
      }
  
      if (show_bits(&gb, 12) == 0xfff) {
          av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
                 "ADTS header detected, probably as result of configuration "
                 "misparsing\n");
          return AVERROR_INVALIDDATA;
      }
  
      switch (latmctx->aac_ctx.oc[1].m4ac.object_type) {
      case AOT_ER_AAC_LC:
      case AOT_ER_AAC_LTP:
      case AOT_ER_AAC_LD:
      case AOT_ER_AAC_ELD:
          err = aac_decode_er_frame(avctx, out, got_frame_ptr, &gb);
          break;
      default:
          err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb, avpkt);
      }
      if (err < 0)
          return err;
  
      return muxlength;
  }
  
  static av_cold int latm_decode_init(AVCodecContext *avctx)
  {
      struct LATMContext *latmctx = avctx->priv_data;
      int ret = aac_decode_init(avctx);
  
      if (avctx->extradata_size > 0)
          latmctx->initialized = !ret;
  
      return ret;
  }
  
  AVCodec ff_aac_decoder = {
      .name            = "aac",
      .long_name       = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
      .type            = AVMEDIA_TYPE_AUDIO,
      .id              = AV_CODEC_ID_AAC,
      .priv_data_size  = sizeof(AACContext),
      .init            = aac_decode_init,
      .close           = aac_decode_close,
      .decode          = aac_decode_frame,
      .sample_fmts     = (const enum AVSampleFormat[]) {
          AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
      },
      .capabilities    = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
      .caps_internal   = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
      .channel_layouts = aac_channel_layout,
      .flush = flush,
      .priv_class      = &aac_decoder_class,
      .profiles        = NULL_IF_CONFIG_SMALL(ff_aac_profiles),
  };
  
  /*
      Note: This decoder filter is intended to decode LATM streams transferred
      in MPEG transport streams which only contain one program.
      To do a more complex LATM demuxing a separate LATM demuxer should be used.
  */
  AVCodec ff_aac_latm_decoder = {
      .name            = "aac_latm",
      .long_name       = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Coding LATM syntax)"),
      .type            = AVMEDIA_TYPE_AUDIO,
      .id              = AV_CODEC_ID_AAC_LATM,
      .priv_data_size  = sizeof(struct LATMContext),
      .init            = latm_decode_init,
      .close           = aac_decode_close,
      .decode          = latm_decode_frame,
      .sample_fmts     = (const enum AVSampleFormat[]) {
          AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
      },
      .capabilities    = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
      .caps_internal   = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
      .channel_layouts = aac_channel_layout,
      .flush = flush,
      .profiles        = NULL_IF_CONFIG_SMALL(ff_aac_profiles),
  };
  