/*
  * Copyright (C) 2016 foo86
  *
  * 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
  */
 
 #include "dcadec.h"
 #include "dcadata.h"
 #include "dcahuff.h"
 #include "dcamath.h"
 #include "dca_syncwords.h"
 
 #if ARCH_ARM
 #include "arm/dca.h"
 #endif
 
 enum HeaderType {
     HEADER_CORE,
     HEADER_XCH,
     HEADER_XXCH
 };
 
 enum AudioMode {
     AMODE_MONO,             // Mode 0: A (mono)
     AMODE_MONO_DUAL,        // Mode 1: A + B (dual mono)
     AMODE_STEREO,           // Mode 2: L + R (stereo)
     AMODE_STEREO_SUMDIFF,   // Mode 3: (L+R) + (L-R) (sum-diff)
     AMODE_STEREO_TOTAL,     // Mode 4: LT + RT (left and right total)
     AMODE_3F,               // Mode 5: C + L + R
     AMODE_2F1R,             // Mode 6: L + R + S
     AMODE_3F1R,             // Mode 7: C + L + R + S
     AMODE_2F2R,             // Mode 8: L + R + SL + SR
     AMODE_3F2R,             // Mode 9: C + L + R + SL + SR
 
     AMODE_COUNT
 };
 
 enum ExtAudioType {
     EXT_AUDIO_XCH   = 0,
     EXT_AUDIO_X96   = 2,
     EXT_AUDIO_XXCH  = 6
 };
 
 enum LFEFlag {
     LFE_FLAG_NONE,
     LFE_FLAG_128,
     LFE_FLAG_64,
     LFE_FLAG_INVALID
 };
 
 static const int8_t prm_ch_to_spkr_map[AMODE_COUNT][5] = {
     { DCA_SPEAKER_C,            -1,             -1,             -1,             -1 },
     { DCA_SPEAKER_L, DCA_SPEAKER_R,             -1,             -1,             -1 },
     { DCA_SPEAKER_L, DCA_SPEAKER_R,             -1,             -1,             -1 },
     { DCA_SPEAKER_L, DCA_SPEAKER_R,             -1,             -1,             -1 },
     { DCA_SPEAKER_L, DCA_SPEAKER_R,             -1,             -1,             -1 },
     { DCA_SPEAKER_C, DCA_SPEAKER_L, DCA_SPEAKER_R ,             -1,             -1 },
     { DCA_SPEAKER_L, DCA_SPEAKER_R, DCA_SPEAKER_Cs,             -1,             -1 },
     { DCA_SPEAKER_C, DCA_SPEAKER_L, DCA_SPEAKER_R , DCA_SPEAKER_Cs,             -1 },
     { DCA_SPEAKER_L, DCA_SPEAKER_R, DCA_SPEAKER_Ls, DCA_SPEAKER_Rs,             -1 },
     { DCA_SPEAKER_C, DCA_SPEAKER_L, DCA_SPEAKER_R,  DCA_SPEAKER_Ls, DCA_SPEAKER_Rs }
 };
 
 static const uint8_t audio_mode_ch_mask[AMODE_COUNT] = {
     DCA_SPEAKER_LAYOUT_MONO,
     DCA_SPEAKER_LAYOUT_STEREO,
     DCA_SPEAKER_LAYOUT_STEREO,
     DCA_SPEAKER_LAYOUT_STEREO,
     DCA_SPEAKER_LAYOUT_STEREO,
     DCA_SPEAKER_LAYOUT_3_0,
     DCA_SPEAKER_LAYOUT_2_1,
     DCA_SPEAKER_LAYOUT_3_1,
     DCA_SPEAKER_LAYOUT_2_2,
     DCA_SPEAKER_LAYOUT_5POINT0
 };
 
 static const uint8_t block_code_nbits[7] = {
     7, 10, 12, 13, 15, 17, 19
 };
 

 static int dca_get_vlc(GetBitContext *s, DCAVLC *v, int i)

 {
     return get_vlc2(s, v->vlc[i].table, v->vlc[i].bits, v->max_depth) + v->offset;
 }
 
 static void get_array(GetBitContext *s, int32_t *array, int size, int n)
 {
     int i;
 
     for (i = 0; i < size; i++)
         array[i] = get_sbits(s, n);
 }
 
 // 5.3.1 - Bit stream header
 static int parse_frame_header(DCACoreDecoder *s)
 {
     int normal_frame, pcmr_index;
 
     // Frame type
     normal_frame = get_bits1(&s->gb);
 
     // Deficit sample count
     if (get_bits(&s->gb, 5) != DCA_PCMBLOCK_SAMPLES - 1) {
         av_log(s->avctx, AV_LOG_ERROR, "Deficit samples are not supported\n");
         return normal_frame ? AVERROR_INVALIDDATA : AVERROR_PATCHWELCOME;
     }
 
     // CRC present flag
     s->crc_present = get_bits1(&s->gb);
 
     // Number of PCM sample blocks
     s->npcmblocks = get_bits(&s->gb, 7) + 1;
     if (s->npcmblocks & (DCA_SUBBAND_SAMPLES - 1)) {
         av_log(s->avctx, AV_LOG_ERROR, "Unsupported number of PCM sample blocks (%d)\n", s->npcmblocks);
         return (s->npcmblocks < 6 || normal_frame) ? AVERROR_INVALIDDATA : AVERROR_PATCHWELCOME;
     }
 
     // Primary frame byte size
     s->frame_size = get_bits(&s->gb, 14) + 1;
     if (s->frame_size < 96) {
         av_log(s->avctx, AV_LOG_ERROR, "Invalid core frame size (%d bytes)\n", s->frame_size);
         return AVERROR_INVALIDDATA;
     }
 
     // Audio channel arrangement
     s->audio_mode = get_bits(&s->gb, 6);
     if (s->audio_mode >= AMODE_COUNT) {
         av_log(s->avctx, AV_LOG_ERROR, "Unsupported audio channel arrangement (%d)\n", s->audio_mode);
         return AVERROR_PATCHWELCOME;
     }
 
     // Core audio sampling frequency
     s->sample_rate = avpriv_dca_sample_rates[get_bits(&s->gb, 4)];
     if (!s->sample_rate) {
         av_log(s->avctx, AV_LOG_ERROR, "Invalid core audio sampling frequency\n");
         return AVERROR_INVALIDDATA;
     }
 
     // Transmission bit rate
     s->bit_rate = ff_dca_bit_rates[get_bits(&s->gb, 5)];
 
     // Reserved field
     skip_bits1(&s->gb);
 
     // Embedded dynamic range flag
     s->drc_present = get_bits1(&s->gb);
 
     // Embedded time stamp flag
     s->ts_present = get_bits1(&s->gb);
 
     // Auxiliary data flag
     s->aux_present = get_bits1(&s->gb);
 
     // HDCD mastering flag
     skip_bits1(&s->gb);
 
     // Extension audio descriptor flag
     s->ext_audio_type = get_bits(&s->gb, 3);
 
     // Extended coding flag
     s->ext_audio_present = get_bits1(&s->gb);
 
     // Audio sync word insertion flag
     s->sync_ssf = get_bits1(&s->gb);
 
     // Low frequency effects flag
     s->lfe_present = get_bits(&s->gb, 2);
     if (s->lfe_present == LFE_FLAG_INVALID) {
         av_log(s->avctx, AV_LOG_ERROR, "Invalid low frequency effects flag\n");
         return AVERROR_INVALIDDATA;
     }
 
     // Predictor history flag switch
     s->predictor_history = get_bits1(&s->gb);
 
     // Header CRC check bytes
     if (s->crc_present)
         skip_bits(&s->gb, 16);
 
     // Multirate interpolator switch
     s->filter_perfect = get_bits1(&s->gb);
 
     // Encoder software revision
     skip_bits(&s->gb, 4);
 
     // Copy history
     skip_bits(&s->gb, 2);
 
     // Source PCM resolution
     s->source_pcm_res = ff_dca_bits_per_sample[pcmr_index = get_bits(&s->gb, 3)];
     if (!s->source_pcm_res) {
         av_log(s->avctx, AV_LOG_ERROR, "Invalid source PCM resolution\n");
         return AVERROR_INVALIDDATA;
     }
     s->es_format = pcmr_index & 1;
 
     // Front sum/difference flag
     s->sumdiff_front = get_bits1(&s->gb);
 
     // Surround sum/difference flag
     s->sumdiff_surround = get_bits1(&s->gb);
 
     // Dialog normalization / unspecified
     skip_bits(&s->gb, 4);
 
     return 0;
 }
 
 // 5.3.2 - Primary audio coding header
 static int parse_coding_header(DCACoreDecoder *s, enum HeaderType header, int xch_base)
 {
     int n, ch, nchannels, header_size = 0, header_pos = get_bits_count(&s->gb);
     unsigned int mask, index;
 
     if (get_bits_left(&s->gb) < 0)
         return AVERROR_INVALIDDATA;
 
     switch (header) {
     case HEADER_CORE:
         // Number of subframes
         s->nsubframes = get_bits(&s->gb, 4) + 1;
 
         // Number of primary audio channels
         s->nchannels = get_bits(&s->gb, 3) + 1;
         if (s->nchannels != ff_dca_channels[s->audio_mode]) {
             av_log(s->avctx, AV_LOG_ERROR, "Invalid number of primary audio channels (%d) for audio channel arrangement (%d)\n", s->nchannels, s->audio_mode);
             return AVERROR_INVALIDDATA;
         }
         av_assert1(s->nchannels <= DCA_CHANNELS - 2);
 
         s->ch_mask = audio_mode_ch_mask[s->audio_mode];
 
         // Add LFE channel if present
         if (s->lfe_present)
             s->ch_mask |= DCA_SPEAKER_MASK_LFE1;
         break;
 
     case HEADER_XCH:
         s->nchannels = ff_dca_channels[s->audio_mode] + 1;
         av_assert1(s->nchannels <= DCA_CHANNELS - 1);
         s->ch_mask |= DCA_SPEAKER_MASK_Cs;
         break;
 
     case HEADER_XXCH:
         // Channel set header length
         header_size = get_bits(&s->gb, 7) + 1;
 
         // Check CRC
         if (s->xxch_crc_present

             && ff_dca_check_crc(s->avctx, &s->gb, header_pos, header_pos + header_size * 8)) {

             av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH channel set header checksum\n");
             return AVERROR_INVALIDDATA;
         }
 
         // Number of channels in a channel set
         nchannels = get_bits(&s->gb, 3) + 1;
         if (nchannels > DCA_XXCH_CHANNELS_MAX) {
             avpriv_request_sample(s->avctx, "%d XXCH channels", nchannels);
             return AVERROR_PATCHWELCOME;
         }
         s->nchannels = ff_dca_channels[s->audio_mode] + nchannels;
         av_assert1(s->nchannels <= DCA_CHANNELS);
 
         // Loudspeaker layout mask
         mask = get_bits_long(&s->gb, s->xxch_mask_nbits - DCA_SPEAKER_Cs);
         s->xxch_spkr_mask = mask << DCA_SPEAKER_Cs;
 
         if (av_popcount(s->xxch_spkr_mask) != nchannels) {
             av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH speaker layout mask (%#x)\n", s->xxch_spkr_mask);
             return AVERROR_INVALIDDATA;
         }
 
         if (s->xxch_core_mask & s->xxch_spkr_mask) {
             av_log(s->avctx, AV_LOG_ERROR, "XXCH speaker layout mask (%#x) overlaps with core (%#x)\n", s->xxch_spkr_mask, s->xxch_core_mask);
             return AVERROR_INVALIDDATA;
         }
 
         // Combine core and XXCH masks together
         s->ch_mask = s->xxch_core_mask | s->xxch_spkr_mask;
 
         // Downmix coefficients present in stream
         if (get_bits1(&s->gb)) {
             int *coeff_ptr = s->xxch_dmix_coeff;
 
             // Downmix already performed by encoder
             s->xxch_dmix_embedded = get_bits1(&s->gb);
 
             // Downmix scale factor
             index = get_bits(&s->gb, 6) * 4 - FF_DCA_DMIXTABLE_OFFSET - 3;
             if (index >= FF_DCA_INV_DMIXTABLE_SIZE) {
                 av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH downmix scale index (%d)\n", index);
                 return AVERROR_INVALIDDATA;
             }
             s->xxch_dmix_scale_inv = ff_dca_inv_dmixtable[index];
 
             // Downmix channel mapping mask
             for (ch = 0; ch < nchannels; ch++) {
                 mask = get_bits_long(&s->gb, s->xxch_mask_nbits);
                 if ((mask & s->xxch_core_mask) != mask) {
                     av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH downmix channel mapping mask (%#x)\n", mask);
                     return AVERROR_INVALIDDATA;
                 }
                 s->xxch_dmix_mask[ch] = mask;
             }
 
             // Downmix coefficients
             for (ch = 0; ch < nchannels; ch++) {
                 for (n = 0; n < s->xxch_mask_nbits; n++) {
                     if (s->xxch_dmix_mask[ch] & (1U << n)) {
                         int code = get_bits(&s->gb, 7);
                         int sign = (code >> 6) - 1;
                         if (code &= 63) {
                             index = code * 4 - 3;
                             if (index >= FF_DCA_DMIXTABLE_SIZE) {
                                 av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH downmix coefficient index (%d)\n", index);
                                 return AVERROR_INVALIDDATA;
                             }
                             *coeff_ptr++ = (ff_dca_dmixtable[index] ^ sign) - sign;
                         } else {
                             *coeff_ptr++ = 0;
                         }
                     }
                 }
             }
         } else {
             s->xxch_dmix_embedded = 0;
         }
 
         break;
     }
 
     // Subband activity count
     for (ch = xch_base; ch < s->nchannels; ch++) {
         s->nsubbands[ch] = get_bits(&s->gb, 5) + 2;
         if (s->nsubbands[ch] > DCA_SUBBANDS) {
             av_log(s->avctx, AV_LOG_ERROR, "Invalid subband activity count\n");
             return AVERROR_INVALIDDATA;
         }
     }
 
     // High frequency VQ start subband
     for (ch = xch_base; ch < s->nchannels; ch++)
         s->subband_vq_start[ch] = get_bits(&s->gb, 5) + 1;
 
     // Joint intensity coding index
     for (ch = xch_base; ch < s->nchannels; ch++) {
         if ((n = get_bits(&s->gb, 3)) && header == HEADER_XXCH)
             n += xch_base - 1;
         if (n > s->nchannels) {
             av_log(s->avctx, AV_LOG_ERROR, "Invalid joint intensity coding index\n");
             return AVERROR_INVALIDDATA;
         }
         s->joint_intensity_index[ch] = n;
     }
 
     // Transient mode code book
     for (ch = xch_base; ch < s->nchannels; ch++)
         s->transition_mode_sel[ch] = get_bits(&s->gb, 2);
 
     // Scale factor code book
     for (ch = xch_base; ch < s->nchannels; ch++) {
         s->scale_factor_sel[ch] = get_bits(&s->gb, 3);
         if (s->scale_factor_sel[ch] == 7) {
             av_log(s->avctx, AV_LOG_ERROR, "Invalid scale factor code book\n");
             return AVERROR_INVALIDDATA;
         }
     }
 
     // Bit allocation quantizer select
     for (ch = xch_base; ch < s->nchannels; ch++) {
         s->bit_allocation_sel[ch] = get_bits(&s->gb, 3);
         if (s->bit_allocation_sel[ch] == 7) {
             av_log(s->avctx, AV_LOG_ERROR, "Invalid bit allocation quantizer select\n");
             return AVERROR_INVALIDDATA;
         }
     }
 
     // Quantization index codebook select
     for (n = 0; n < DCA_CODE_BOOKS; n++)
         for (ch = xch_base; ch < s->nchannels; ch++)

             s->quant_index_sel[ch][n] = get_bits(&s->gb, ff_dca_quant_index_sel_nbits[n]);

 
     // Scale factor adjustment index
     for (n = 0; n < DCA_CODE_BOOKS; n++)
         for (ch = xch_base; ch < s->nchannels; ch++)

             if (s->quant_index_sel[ch][n] < ff_dca_quant_index_group_size[n])

                 s->scale_factor_adj[ch][n] = ff_dca_scale_factor_adj[get_bits(&s->gb, 2)];
 
     if (header == HEADER_XXCH) {
         // Reserved
         // Byte align
         // CRC16 of channel set header
         if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8)) {
             av_log(s->avctx, AV_LOG_ERROR, "Read past end of XXCH channel set header\n");
             return AVERROR_INVALIDDATA;
         }
     } else {
         // Audio header CRC check word
         if (s->crc_present)
             skip_bits(&s->gb, 16);
     }
 
     return 0;
 }
 
 static inline int parse_scale(DCACoreDecoder *s, int *scale_index, int sel)
 {
     const uint32_t *scale_table;
     unsigned int scale_size;
 
     // Select the root square table
     if (sel > 5) {
         scale_table = ff_dca_scale_factor_quant7;
         scale_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant7);
     } else {
         scale_table = ff_dca_scale_factor_quant6;
         scale_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant6);
     }
 
     // If Huffman code was used, the difference of scales was encoded
     if (sel < 5)

         *scale_index += dca_get_vlc(&s->gb, &ff_dca_vlc_scale_factor, sel);

     else
         *scale_index = get_bits(&s->gb, sel + 1);
 
     // Look up scale factor from the root square table
     if ((unsigned int)*scale_index >= scale_size) {
         av_log(s->avctx, AV_LOG_ERROR, "Invalid scale factor index\n");
         return AVERROR_INVALIDDATA;
     }
 
     return scale_table[*scale_index];
 }
 
 static inline int parse_joint_scale(DCACoreDecoder *s, int sel)
 {
     int scale_index;
 
     // Absolute value was encoded even when Huffman code was used
     if (sel < 5)

         scale_index = dca_get_vlc(&s->gb, &ff_dca_vlc_scale_factor, sel);

     else
         scale_index = get_bits(&s->gb, sel + 1);
 
     // Bias by 64
     scale_index += 64;
 
     // Look up joint scale factor
     if ((unsigned int)scale_index >= FF_ARRAY_ELEMS(ff_dca_joint_scale_factors)) {
         av_log(s->avctx, AV_LOG_ERROR, "Invalid joint scale factor index\n");
         return AVERROR_INVALIDDATA;
     }
 
     return ff_dca_joint_scale_factors[scale_index];
 }
 
 // 5.4.1 - Primary audio coding side information
 static int parse_subframe_header(DCACoreDecoder *s, int sf,
                                  enum HeaderType header, int xch_base)
 {
     int ch, band, ret;
 
     if (get_bits_left(&s->gb) < 0)
         return AVERROR_INVALIDDATA;
 
     if (header == HEADER_CORE) {
         // Subsubframe count
         s->nsubsubframes[sf] = get_bits(&s->gb, 2) + 1;
 
         // Partial subsubframe sample count
         skip_bits(&s->gb, 3);
     }
 
     // Prediction mode
     for (ch = xch_base; ch < s->nchannels; ch++)
         for (band = 0; band < s->nsubbands[ch]; band++)
             s->prediction_mode[ch][band] = get_bits1(&s->gb);
 
     // Prediction coefficients VQ address
     for (ch = xch_base; ch < s->nchannels; ch++)
         for (band = 0; band < s->nsubbands[ch]; band++)
             if (s->prediction_mode[ch][band])
                 s->prediction_vq_index[ch][band] = get_bits(&s->gb, 12);
 
     // Bit allocation index
     for (ch = xch_base; ch < s->nchannels; ch++) {
         int sel = s->bit_allocation_sel[ch];
 
         for (band = 0; band < s->subband_vq_start[ch]; band++) {
             int abits;
 
             if (sel < 5)

                 abits = dca_get_vlc(&s->gb, &ff_dca_vlc_bit_allocation, sel);

             else
                 abits = get_bits(&s->gb, sel - 1);
 
             if (abits > DCA_ABITS_MAX) {
                 av_log(s->avctx, AV_LOG_ERROR, "Invalid bit allocation index\n");
                 return AVERROR_INVALIDDATA;
             }
 
             s->bit_allocation[ch][band] = abits;
         }
     }
 
     // Transition mode
     for (ch = xch_base; ch < s->nchannels; ch++) {
         // Clear transition mode for all subbands
         memset(s->transition_mode[sf][ch], 0, sizeof(s->transition_mode[0][0]));
 
         // Transient possible only if more than one subsubframe
         if (s->nsubsubframes[sf] > 1) {
             int sel = s->transition_mode_sel[ch];
             for (band = 0; band < s->subband_vq_start[ch]; band++)
                 if (s->bit_allocation[ch][band])

                     s->transition_mode[sf][ch][band] = dca_get_vlc(&s->gb, &ff_dca_vlc_transition_mode, sel);

         }
     }
 
     // Scale factors
     for (ch = xch_base; ch < s->nchannels; ch++) {
         int sel = s->scale_factor_sel[ch];
         int scale_index = 0;
 
         // Extract scales for subbands up to VQ
         for (band = 0; band < s->subband_vq_start[ch]; band++) {
             if (s->bit_allocation[ch][band]) {
                 if ((ret = parse_scale(s, &scale_index, sel)) < 0)
                     return ret;
                 s->scale_factors[ch][band][0] = ret;
                 if (s->transition_mode[sf][ch][band]) {
                     if ((ret = parse_scale(s, &scale_index, sel)) < 0)
                         return ret;
                     s->scale_factors[ch][band][1] = ret;
                 }
             } else {
                 s->scale_factors[ch][band][0] = 0;
             }
         }
 
         // High frequency VQ subbands
         for (band = s->subband_vq_start[ch]; band < s->nsubbands[ch]; band++) {
             if ((ret = parse_scale(s, &scale_index, sel)) < 0)
                 return ret;
             s->scale_factors[ch][band][0] = ret;
         }
     }
 
     // Joint subband codebook select
     for (ch = xch_base; ch < s->nchannels; ch++) {
         if (s->joint_intensity_index[ch]) {
             s->joint_scale_sel[ch] = get_bits(&s->gb, 3);
             if (s->joint_scale_sel[ch] == 7) {
                 av_log(s->avctx, AV_LOG_ERROR, "Invalid joint scale factor code book\n");
                 return AVERROR_INVALIDDATA;
             }
         }
     }
 
     // Scale factors for joint subband coding
     for (ch = xch_base; ch < s->nchannels; ch++) {
         int src_ch = s->joint_intensity_index[ch] - 1;
         if (src_ch >= 0) {
             int sel = s->joint_scale_sel[ch];
             for (band = s->nsubbands[ch]; band < s->nsubbands[src_ch]; band++) {
                 if ((ret = parse_joint_scale(s, sel)) < 0)
                     return ret;
                 s->joint_scale_factors[ch][band] = ret;
             }
         }
     }
 
     // Dynamic range coefficient
     if (s->drc_present && header == HEADER_CORE)
         skip_bits(&s->gb, 8);
 
     // Side information CRC check word
     if (s->crc_present)
         skip_bits(&s->gb, 16);
 
     return 0;
 }
 
 #ifndef decode_blockcodes
 static inline int decode_blockcodes(int code1, int code2, int levels, int32_t *audio)
 {
     int offset = (levels - 1) / 2;
     int n, div;
 
     for (n = 0; n < DCA_SUBBAND_SAMPLES / 2; n++) {
         div = FASTDIV(code1, levels);
         audio[n] = code1 - div * levels - offset;
         code1 = div;
     }
     for (; n < DCA_SUBBAND_SAMPLES; n++) {
         div = FASTDIV(code2, levels);
         audio[n] = code2 - div * levels - offset;
         code2 = div;
     }
 
     return code1 | code2;
 }
 #endif
 
 static inline int parse_block_codes(DCACoreDecoder *s, int32_t *audio, int abits)
 {
     // Extract block code indices from the bit stream
     int code1 = get_bits(&s->gb, block_code_nbits[abits - 1]);
     int code2 = get_bits(&s->gb, block_code_nbits[abits - 1]);
     int levels = ff_dca_quant_levels[abits];
 
     // Look up samples from the block code book
     if (decode_blockcodes(code1, code2, levels, audio)) {
         av_log(s->avctx, AV_LOG_ERROR, "Failed to decode block code(s)\n");
         return AVERROR_INVALIDDATA;
     }
 
     return 0;
 }
 
 static inline int parse_huffman_codes(DCACoreDecoder *s, int32_t *audio, int abits, int sel)
 {
     int i;
 
     // Extract Huffman codes from the bit stream
     for (i = 0; i < DCA_SUBBAND_SAMPLES; i++)

         audio[i] = dca_get_vlc(&s->gb, &ff_dca_vlc_quant_index[abits - 1], sel);

 
     return 1;
 }
 
 static inline int extract_audio(DCACoreDecoder *s, int32_t *audio, int abits, int ch)
 {
     av_assert1(abits >= 0 && abits <= DCA_ABITS_MAX);
 
     if (abits == 0) {
         // No bits allocated
         memset(audio, 0, DCA_SUBBAND_SAMPLES * sizeof(*audio));
         return 0;
     }
 
     if (abits <= DCA_CODE_BOOKS) {
         int sel = s->quant_index_sel[ch][abits - 1];

         if (sel < ff_dca_quant_index_group_size[abits - 1]) {

             // Huffman codes
             return parse_huffman_codes(s, audio, abits, sel);
         }
         if (abits <= 7) {
             // Block codes
             return parse_block_codes(s, audio, abits);
         }
     }
 
     // No further encoding
     get_array(&s->gb, audio, DCA_SUBBAND_SAMPLES, abits - 3);
     return 0;
 }
 
 static inline void dequantize(int32_t *output, const int32_t *input,
                               int32_t step_size, int32_t scale, int residual)
 {
     // Account for quantizer step size
     int64_t step_scale = (int64_t)step_size * scale;
     int n, shift = 0;
 
     // Limit scale factor resolution to 22 bits
     if (step_scale > (1 << 23)) {
         shift = av_log2(step_scale >> 23) + 1;
         step_scale >>= shift;
     }
 
     // Scale the samples
     if (residual) {
         for (n = 0; n < DCA_SUBBAND_SAMPLES; n++)
             output[n] += clip23(norm__(input[n] * step_scale, 22 - shift));
     } else {
         for (n = 0; n < DCA_SUBBAND_SAMPLES; n++)
             output[n]  = clip23(norm__(input[n] * step_scale, 22 - shift));
     }
 }
 
 static inline void inverse_adpcm(int32_t **subband_samples,
                                  const int16_t *vq_index,
                                  const int8_t *prediction_mode,
                                  int sb_start, int sb_end,
                                  int ofs, int len)
 {
     int i, j, k;
 
     for (i = sb_start; i < sb_end; i++) {
         if (prediction_mode[i]) {
             const int16_t *coeff = ff_dca_adpcm_vb[vq_index[i]];
             int32_t *ptr = subband_samples[i] + ofs;
             for (j = 0; j < len; j++) {
                 int64_t err = 0;
                 for (k = 0; k < DCA_ADPCM_COEFFS; k++)
                     err += (int64_t)ptr[j - k - 1] * coeff[k];
                 ptr[j] = clip23(ptr[j] + clip23(norm13(err)));
             }
         }
     }
 }
 
 // 5.5 - Primary audio data arrays
 static int parse_subframe_audio(DCACoreDecoder *s, int sf, enum HeaderType header,
                                 int xch_base, int *sub_pos, int *lfe_pos)
 {
     int32_t audio[16], scale;
     int n, ssf, ofs, ch, band;
 
     // Check number of subband samples in this subframe
     int nsamples = s->nsubsubframes[sf] * DCA_SUBBAND_SAMPLES;
     if (*sub_pos + nsamples > s->npcmblocks) {
         av_log(s->avctx, AV_LOG_ERROR, "Subband sample buffer overflow\n");
         return AVERROR_INVALIDDATA;
     }
 
     if (get_bits_left(&s->gb) < 0)
         return AVERROR_INVALIDDATA;
 
     // VQ encoded subbands
     for (ch = xch_base; ch < s->nchannels; ch++) {
         int32_t vq_index[DCA_SUBBANDS];
 
         for (band = s->subband_vq_start[ch]; band < s->nsubbands[ch]; band++)
             // Extract the VQ address from the bit stream
             vq_index[band] = get_bits(&s->gb, 10);
 
         if (s->subband_vq_start[ch] < s->nsubbands[ch]) {
             s->dcadsp->decode_hf(s->subband_samples[ch], vq_index,
                                  ff_dca_high_freq_vq, s->scale_factors[ch],
                                  s->subband_vq_start[ch], s->nsubbands[ch],
                                  *sub_pos, nsamples);
         }
     }
 
     // Low frequency effect data
     if (s->lfe_present && header == HEADER_CORE) {
         unsigned int index;
 
         // Determine number of LFE samples in this subframe
         int nlfesamples = 2 * s->lfe_present * s->nsubsubframes[sf];
         av_assert1((unsigned int)nlfesamples <= FF_ARRAY_ELEMS(audio));
 
         // Extract LFE samples from the bit stream
         get_array(&s->gb, audio, nlfesamples, 8);
 
         // Extract scale factor index from the bit stream
         index = get_bits(&s->gb, 8);
         if (index >= FF_ARRAY_ELEMS(ff_dca_scale_factor_quant7)) {
             av_log(s->avctx, AV_LOG_ERROR, "Invalid LFE scale factor index\n");
             return AVERROR_INVALIDDATA;
         }
 
         // Look up the 7-bit root square quantization table
         scale = ff_dca_scale_factor_quant7[index];
 
         // Account for quantizer step size which is 0.035
         scale = mul23(4697620 /* 0.035 * (1 << 27) */, scale);
 
         // Scale and take the LFE samples
         for (n = 0, ofs = *lfe_pos; n < nlfesamples; n++, ofs++)
             s->lfe_samples[ofs] = clip23(audio[n] * scale >> 4);
 
         // Advance LFE sample pointer for the next subframe
         *lfe_pos = ofs;
     }
 
     // Audio data
     for (ssf = 0, ofs = *sub_pos; ssf < s->nsubsubframes[sf]; ssf++) {
         for (ch = xch_base; ch < s->nchannels; ch++) {
             if (get_bits_left(&s->gb) < 0)
                 return AVERROR_INVALIDDATA;
 
             // Not high frequency VQ subbands
             for (band = 0; band < s->subband_vq_start[ch]; band++) {
                 int ret, trans_ssf, abits = s->bit_allocation[ch][band];
                 int32_t step_size;
 
                 // Extract bits from the bit stream
                 if ((ret = extract_audio(s, audio, abits, ch)) < 0)
                     return ret;
 
                 // Select quantization step size table and look up
                 // quantization step size
                 if (s->bit_rate == 3)
                     step_size = ff_dca_lossless_quant[abits];
                 else
                     step_size = ff_dca_lossy_quant[abits];
 
                 // Identify transient location
                 trans_ssf = s->transition_mode[sf][ch][band];
 
                 // Determine proper scale factor
                 if (trans_ssf == 0 || ssf < trans_ssf)
                     scale = s->scale_factors[ch][band][0];
                 else
                     scale = s->scale_factors[ch][band][1];
 
                 // Adjust scale factor when SEL indicates Huffman code
                 if (ret > 0) {
                     int64_t adj = s->scale_factor_adj[ch][abits - 1];
                     scale = clip23(adj * scale >> 22);
                 }
 
                 dequantize(s->subband_samples[ch][band] + ofs,
                            audio, step_size, scale, 0);
             }
         }
 
         // DSYNC
         if ((ssf == s->nsubsubframes[sf] - 1 || s->sync_ssf) && get_bits(&s->gb, 16) != 0xffff) {
             av_log(s->avctx, AV_LOG_ERROR, "DSYNC check failed\n");
             return AVERROR_INVALIDDATA;
         }
 
         ofs += DCA_SUBBAND_SAMPLES;
     }
 
     // Inverse ADPCM
     for (ch = xch_base; ch < s->nchannels; ch++) {
         inverse_adpcm(s->subband_samples[ch], s->prediction_vq_index[ch],
                       s->prediction_mode[ch], 0, s->nsubbands[ch],
                       *sub_pos, nsamples);
     }
 
     // Joint subband coding
     for (ch = xch_base; ch < s->nchannels; ch++) {
         int src_ch = s->joint_intensity_index[ch] - 1;
         if (src_ch >= 0) {
             s->dcadsp->decode_joint(s->subband_samples[ch], s->subband_samples[src_ch],
                                     s->joint_scale_factors[ch], s->nsubbands[ch],
                                     s->nsubbands[src_ch], *sub_pos, nsamples);
         }
     }
 
     // Advance subband sample pointer for the next subframe
     *sub_pos = ofs;
     return 0;
 }
 
 static void erase_adpcm_history(DCACoreDecoder *s)
 {
     int ch, band;
 
     // Erase ADPCM history from previous frame if
     // predictor history switch was disabled
     for (ch = 0; ch < DCA_CHANNELS; ch++)
         for (band = 0; band < DCA_SUBBANDS; band++)
             AV_ZERO128(s->subband_samples[ch][band] - DCA_ADPCM_COEFFS);

 
     emms_c();

 }
 
 static int alloc_sample_buffer(DCACoreDecoder *s)
 {
     int nchsamples = DCA_ADPCM_COEFFS + s->npcmblocks;
     int nframesamples = nchsamples * DCA_CHANNELS * DCA_SUBBANDS;
     int nlfesamples = DCA_LFE_HISTORY + s->npcmblocks / 2;
     unsigned int size = s->subband_size;
     int ch, band;
 
     // Reallocate subband sample buffer
     av_fast_mallocz(&s->subband_buffer, &s->subband_size,
                     (nframesamples + nlfesamples) * sizeof(int32_t));
     if (!s->subband_buffer)
         return AVERROR(ENOMEM);
 
     if (size != s->subband_size) {
         for (ch = 0; ch < DCA_CHANNELS; ch++)
             for (band = 0; band < DCA_SUBBANDS; band++)
                 s->subband_samples[ch][band] = s->subband_buffer +
                     (ch * DCA_SUBBANDS + band) * nchsamples + DCA_ADPCM_COEFFS;
         s->lfe_samples = s->subband_buffer + nframesamples;
     }
 
     if (!s->predictor_history)
         erase_adpcm_history(s);
 
     return 0;
 }
 
 static int parse_frame_data(DCACoreDecoder *s, enum HeaderType header, int xch_base)
 {
     int sf, ch, ret, band, sub_pos, lfe_pos;
 
     if ((ret = parse_coding_header(s, header, xch_base)) < 0)
         return ret;
 
     for (sf = 0, sub_pos = 0, lfe_pos = DCA_LFE_HISTORY; sf < s->nsubframes; sf++) {
         if ((ret = parse_subframe_header(s, sf, header, xch_base)) < 0)
             return ret;
         if ((ret = parse_subframe_audio(s, sf, header, xch_base, &sub_pos, &lfe_pos)) < 0)
             return ret;
     }
 
     for (ch = xch_base; ch < s->nchannels; ch++) {
         // Determine number of active subbands for this channel
         int nsubbands = s->nsubbands[ch];
         if (s->joint_intensity_index[ch])
             nsubbands = FFMAX(nsubbands, s->nsubbands[s->joint_intensity_index[ch] - 1]);
 
         // Update history for ADPCM
         for (band = 0; band < nsubbands; band++) {
             int32_t *samples = s->subband_samples[ch][band] - DCA_ADPCM_COEFFS;
             AV_COPY128(samples, samples + s->npcmblocks);
         }
 
         // Clear inactive subbands
         for (; band < DCA_SUBBANDS; band++) {
             int32_t *samples = s->subband_samples[ch][band] - DCA_ADPCM_COEFFS;
             memset(samples, 0, (DCA_ADPCM_COEFFS + s->npcmblocks) * sizeof(int32_t));
         }
     }
 

     emms_c();
 

     return 0;
 }
 
 static int parse_xch_frame(DCACoreDecoder *s)
 {
     int ret;
 
     if (s->ch_mask & DCA_SPEAKER_MASK_Cs) {
         av_log(s->avctx, AV_LOG_ERROR, "XCH with Cs speaker already present\n");
         return AVERROR_INVALIDDATA;
     }
 
     if ((ret = parse_frame_data(s, HEADER_XCH, s->nchannels)) < 0)
         return ret;
 
     // Seek to the end of core frame, don't trust XCH frame size
     if (ff_dca_seek_bits(&s->gb, s->frame_size * 8)) {
         av_log(s->avctx, AV_LOG_ERROR, "Read past end of XCH frame\n");
         return AVERROR_INVALIDDATA;
     }
 
     return 0;
 }
 
 static int parse_xxch_frame(DCACoreDecoder *s)
 {
     int xxch_nchsets, xxch_frame_size;
     int ret, mask, header_size, header_pos = get_bits_count(&s->gb);
 
     // XXCH sync word
     if (get_bits_long(&s->gb, 32) != DCA_SYNCWORD_XXCH) {
         av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH sync word\n");
         return AVERROR_INVALIDDATA;
     }
 
     // XXCH frame header length
     header_size = get_bits(&s->gb, 6) + 1;
 
     // Check XXCH frame header CRC

     if (ff_dca_check_crc(s->avctx, &s->gb, header_pos + 32, header_pos + header_size * 8)) {

         av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH frame header checksum\n");
         return AVERROR_INVALIDDATA;
     }
 
     // CRC presence flag for channel set header
     s->xxch_crc_present = get_bits1(&s->gb);
 
     // Number of bits for loudspeaker mask
     s->xxch_mask_nbits = get_bits(&s->gb, 5) + 1;
     if (s->xxch_mask_nbits <= DCA_SPEAKER_Cs) {
         av_log(s->avctx, AV_LOG_ERROR, "Invalid number of bits for XXCH speaker mask (%d)\n", s->xxch_mask_nbits);
         return AVERROR_INVALIDDATA;
     }
 
     // Number of channel sets
     xxch_nchsets = get_bits(&s->gb, 2) + 1;
     if (xxch_nchsets > 1) {
         avpriv_request_sample(s->avctx, "%d XXCH channel sets", xxch_nchsets);
         return AVERROR_PATCHWELCOME;
     }
 
     // Channel set 0 data byte size
     xxch_frame_size = get_bits(&s->gb, 14) + 1;
 
     // Core loudspeaker activity mask
     s->xxch_core_mask = get_bits_long(&s->gb, s->xxch_mask_nbits);
 
     // Validate the core mask
     mask = s->ch_mask;
 
     if ((mask & DCA_SPEAKER_MASK_Ls) && (s->xxch_core_mask & DCA_SPEAKER_MASK_Lss))
         mask = (mask & ~DCA_SPEAKER_MASK_Ls) | DCA_SPEAKER_MASK_Lss;
 
     if ((mask & DCA_SPEAKER_MASK_Rs) && (s->xxch_core_mask & DCA_SPEAKER_MASK_Rss))
         mask = (mask & ~DCA_SPEAKER_MASK_Rs) | DCA_SPEAKER_MASK_Rss;
 
     if (mask != s->xxch_core_mask) {
         av_log(s->avctx, AV_LOG_ERROR, "XXCH core speaker activity mask (%#x) disagrees with core (%#x)\n", s->xxch_core_mask, mask);
         return AVERROR_INVALIDDATA;
     }
 
     // Reserved
     // Byte align
     // CRC16 of XXCH frame header
     if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8)) {
         av_log(s->avctx, AV_LOG_ERROR, "Read past end of XXCH frame header\n");
         return AVERROR_INVALIDDATA;
     }
 
     // Parse XXCH channel set 0
     if ((ret = parse_frame_data(s, HEADER_XXCH, s->nchannels)) < 0)
         return ret;
 
     if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8 + xxch_frame_size * 8)) {
         av_log(s->avctx, AV_LOG_ERROR, "Read past end of XXCH channel set\n");
         return AVERROR_INVALIDDATA;
     }
 
     return 0;
 }
 
 static int parse_xbr_subframe(DCACoreDecoder *s, int xbr_base_ch, int xbr_nchannels,
                               int *xbr_nsubbands, int xbr_transition_mode, int sf, int *sub_pos)
 {
     int     xbr_nabits[DCA_CHANNELS];
     int     xbr_bit_allocation[DCA_CHANNELS][DCA_SUBBANDS];
     int     xbr_scale_nbits[DCA_CHANNELS];
     int32_t xbr_scale_factors[DCA_CHANNELS][DCA_SUBBANDS][2];
     int     ssf, ch, band, ofs;
 
     // Check number of subband samples in this subframe
     if (*sub_pos + s->nsubsubframes[sf] * DCA_SUBBAND_SAMPLES > s->npcmblocks) {
         av_log(s->avctx, AV_LOG_ERROR, "Subband sample buffer overflow\n");
         return AVERROR_INVALIDDATA;
     }
 
     if (get_bits_left(&s->gb) < 0)
         return AVERROR_INVALIDDATA;
 
     // Number of bits for XBR bit allocation index
     for (ch = xbr_base_ch; ch < xbr_nchannels; ch++)
         xbr_nabits[ch] = get_bits(&s->gb, 2) + 2;
 
     // XBR bit allocation index
     for (ch = xbr_base_ch; ch < xbr_nchannels; ch++) {
         for (band = 0; band < xbr_nsubbands[ch]; band++) {
             xbr_bit_allocation[ch][band] = get_bits(&s->gb, xbr_nabits[ch]);
             if (xbr_bit_allocation[ch][band] > DCA_ABITS_MAX) {
                 av_log(s->avctx, AV_LOG_ERROR, "Invalid XBR bit allocation index\n");
                 return AVERROR_INVALIDDATA;
             }
         }
     }
 
     // Number of bits for scale indices
     for (ch = xbr_base_ch; ch < xbr_nchannels; ch++) {
         xbr_scale_nbits[ch] = get_bits(&s->gb, 3);
         if (!xbr_scale_nbits[ch]) {
             av_log(s->avctx, AV_LOG_ERROR, "Invalid number of bits for XBR scale factor index\n");
             return AVERROR_INVALIDDATA;
         }
     }
 
     // XBR scale factors
     for (ch = xbr_base_ch; ch < xbr_nchannels; ch++) {
         const uint32_t *scale_table;
         int scale_size;
 
         // Select the root square table
         if (s->scale_factor_sel[ch] > 5) {
             scale_table = ff_dca_scale_factor_quant7;
             scale_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant7);
         } else {
             scale_table = ff_dca_scale_factor_quant6;
             scale_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant6);
         }
 
         // Parse scale factor indices and look up scale factors from the root
         // square table
         for (band = 0; band < xbr_nsubbands[ch]; band++) {
             if (xbr_bit_allocation[ch][band]) {
                 int scale_index = get_bits(&s->gb, xbr_scale_nbits[ch]);
                 if (scale_index >= scale_size) {
                     av_log(s->avctx, AV_LOG_ERROR, "Invalid XBR scale factor index\n");
                     return AVERROR_INVALIDDATA;
                 }
                 xbr_scale_factors[ch][band][0] = scale_table[scale_index];
                 if (xbr_transition_mode && s->transition_mode[sf][ch][band]) {
                     scale_index = get_bits(&s->gb, xbr_scale_nbits[ch]);
                     if (scale_index >= scale_size) {
                         av_log(s->avctx, AV_LOG_ERROR, "Invalid XBR scale factor index\n");
                         return AVERROR_INVALIDDATA;
                     }
                     xbr_scale_factors[ch][band][1] = scale_table[scale_index];
                 }
             }
         }
     }
 
     // Audio data
     for (ssf = 0, ofs = *sub_pos; ssf < s->nsubsubframes[sf]; ssf++) {
         for (ch = xbr_base_ch; ch < xbr_nchannels; ch++) {
             if (get_bits_left(&s->gb) < 0)
                 return AVERROR_INVALIDDATA;
 
             for (band = 0; band < xbr_nsubbands[ch]; band++) {
                 int ret, trans_ssf, abits = xbr_bit_allocation[ch][band];
                 int32_t audio[DCA_SUBBAND_SAMPLES], step_size, scale;
 
                 // Extract bits from the bit stream
                 if (abits > 7) {
                     // No further encoding
                     get_array(&s->gb, audio, DCA_SUBBAND_SAMPLES, abits - 3);
                 } else if (abits > 0) {
                     // Block codes
                     if ((ret = parse_block_codes(s, audio, abits)) < 0)
                         return ret;
                 } else {
                     // No bits allocated
                     continue;
                 }
 
                 // Look up quantization step size
                 step_size = ff_dca_lossless_quant[abits];
 
                 // Identify transient location
                 if (xbr_transition_mode)
                     trans_ssf = s->transition_mode[sf][ch][band];
                 else
                     trans_ssf = 0;
 
                 // Determine proper scale factor
                 if (trans_ssf == 0 || ssf < trans_ssf)
                     scale = xbr_scale_factors[ch][band][0];
                 else
                     scale = xbr_scale_factors[ch][band][1];
 
                 dequantize(s->subband_samples[ch][band] + ofs,
                            audio, step_size, scale, 1);
             }
         }
 
         // DSYNC
         if ((ssf == s->nsubsubframes[sf] - 1 || s->sync_ssf) && get_bits(&s->gb, 16) != 0xffff) {
             av_log(s->avctx, AV_LOG_ERROR, "XBR-DSYNC check failed\n");
             return AVERROR_INVALIDDATA;
         }
 
         ofs += DCA_SUBBAND_SAMPLES;
     }
 
     // Advance subband sample pointer for the next subframe
     *sub_pos = ofs;
     return 0;
 }
 
 static int parse_xbr_frame(DCACoreDecoder *s)
 {
     int     xbr_frame_size[DCA_EXSS_CHSETS_MAX];
     int     xbr_nchannels[DCA_EXSS_CHSETS_MAX];
     int     xbr_nsubbands[DCA_EXSS_CHSETS_MAX * DCA_EXSS_CHANNELS_MAX];
     int     xbr_nchsets, xbr_transition_mode, xbr_band_nbits, xbr_base_ch;
     int     i, ch1, ch2, ret, header_size, header_pos = get_bits_count(&s->gb);
 
     // XBR sync word
     if (get_bits_long(&s->gb, 32) != DCA_SYNCWORD_XBR) {
         av_log(s->avctx, AV_LOG_ERROR, "Invalid XBR sync word\n");
         return AVERROR_INVALIDDATA;
     }
 
     // XBR frame header length
     header_size = get_bits(&s->gb, 6) + 1;
 
     // Check XBR frame header CRC

     if (ff_dca_check_crc(s->avctx, &s->gb, header_pos + 32, header_pos + header_size * 8)) {

         av_log(s->avctx, AV_LOG_ERROR, "Invalid XBR frame header checksum\n");
         return AVERROR_INVALIDDATA;
     }
 
     // Number of channel sets
     xbr_nchsets = get_bits(&s->gb, 2) + 1;
 
     // Channel set data byte size
     for (i = 0; i < xbr_nchsets; i++)
         xbr_frame_size[i] = get_bits(&s->gb, 14) + 1;
 
     // Transition mode flag
     xbr_transition_mode = get_bits1(&s->gb);
 
     // Channel set headers
     for (i = 0, ch2 = 0; i < xbr_nchsets; i++) {
         xbr_nchannels[i] = get_bits(&s->gb, 3) + 1;
         xbr_band_nbits = get_bits(&s->gb, 2) + 5;
         for (ch1 = 0; ch1 < xbr_nchannels[i]; ch1++, ch2++) {
             xbr_nsubbands[ch2] = get_bits(&s->gb, xbr_band_nbits) + 1;
             if (xbr_nsubbands[ch2] > DCA_SUBBANDS) {
                 av_log(s->avctx, AV_LOG_ERROR, "Invalid number of active XBR subbands (%d)\n", xbr_nsubbands[ch2]);
                 return AVERROR_INVALIDDATA;
             }
         }
     }
 
     // Reserved
     // Byte align
     // CRC16 of XBR frame header
     if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8)) {
         av_log(s->avctx, AV_LOG_ERROR, "Read past end of XBR frame header\n");
         return AVERROR_INVALIDDATA;
     }
 
     // Channel set data
     for (i = 0, xbr_base_ch = 0; i < xbr_nchsets; i++) {
         header_pos = get_bits_count(&s->gb);
 
         if (xbr_base_ch + xbr_nchannels[i] <= s->nchannels) {
             int sf, sub_pos;
 
             for (sf = 0, sub_pos = 0; sf < s->nsubframes; sf++) {
                 if ((ret = parse_xbr_subframe(s, xbr_base_ch,
                                               xbr_base_ch + xbr_nchannels[i],
                                               xbr_nsubbands, xbr_transition_mode,
                                               sf, &sub_pos)) < 0)
                     return ret;
             }
         }
 
         xbr_base_ch += xbr_nchannels[i];
 
         if (ff_dca_seek_bits(&s->gb, header_pos + xbr_frame_size[i] * 8)) {
             av_log(s->avctx, AV_LOG_ERROR, "Read past end of XBR channel set\n");
             return AVERROR_INVALIDDATA;
         }
     }
 
     return 0;
 }
 
 // Modified ISO/IEC 9899 linear congruential generator
 // Returns pseudorandom integer in range [-2^30, 2^30 - 1]
 static int rand_x96(DCACoreDecoder *s)
 {
     s->x96_rand = 1103515245U * s->x96_rand + 12345U;
     return (s->x96_rand & 0x7fffffff) - 0x40000000;
 }
 
 static int parse_x96_subframe_audio(DCACoreDecoder *s, int sf, int xch_base, int *sub_pos)
 {
     int n, ssf, ch, band, ofs;
 
     // Check number of subband samples in this subframe
     int nsamples = s->nsubsubframes[sf] * DCA_SUBBAND_SAMPLES;
     if (*sub_pos + nsamples > s->npcmblocks) {
         av_log(s->avctx, AV_LOG_ERROR, "Subband sample buffer overflow\n");
         return AVERROR_INVALIDDATA;
     }
 
     if (get_bits_left(&s->gb) < 0)
         return AVERROR_INVALIDDATA;
 
     // VQ encoded or unallocated subbands
     for (ch = xch_base; ch < s->x96_nchannels; ch++) {
         for (band = s->x96_subband_start; band < s->nsubbands[ch]; band++) {
             // Get the sample pointer and scale factor
             int32_t *samples = s->x96_subband_samples[ch][band] + *sub_pos;
             int32_t scale    = s->scale_factors[ch][band >> 1][band & 1];
 
             switch (s->bit_allocation[ch][band]) {
             case 0: // No bits allocated for subband
                 if (scale <= 1)
                     memset(samples, 0, nsamples * sizeof(int32_t));
                 else for (n = 0; n < nsamples; n++)
                     // Generate scaled random samples
                     samples[n] = mul31(rand_x96(s), scale);
                 break;
 
             case 1: // VQ encoded subband
                 for (ssf = 0; ssf < (s->nsubsubframes[sf] + 1) / 2; ssf++) {
                     // Extract the VQ address from the bit stream and look up
                     // the VQ code book for up to 16 subband samples
                     const int8_t *vq_samples = ff_dca_high_freq_vq[get_bits(&s->gb, 10)];
                     // Scale and take the samples
                     for (n = 0; n < FFMIN(nsamples - ssf * 16, 16); n++)
                         *samples++ = clip23(vq_samples[n] * scale + (1 << 3) >> 4);
                 }
                 break;
             }
         }
     }
 
     // Audio data
     for (ssf = 0, ofs = *sub_pos; ssf < s->nsubsubframes[sf]; ssf++) {
         for (ch = xch_base; ch < s->x96_nchannels; ch++) {
             if (get_bits_left(&s->gb) < 0)
                 return AVERROR_INVALIDDATA;
 
             for (band = s->x96_subband_start; band < s->nsubbands[ch]; band++) {
                 int ret, abits = s->bit_allocation[ch][band] - 1;
                 int32_t audio[DCA_SUBBAND_SAMPLES], step_size, scale;
 
                 // Not VQ encoded or unallocated subbands
                 if (abits < 1)
                     continue;
 
                 // Extract bits from the bit stream
                 if ((ret = extract_audio(s, audio, abits, ch)) < 0)
                     return ret;
 
                 // Select quantization step size table and look up quantization
                 // step size
                 if (s->bit_rate == 3)
                     step_size = ff_dca_lossless_quant[abits];
                 else
                     step_size = ff_dca_lossy_quant[abits];
 
                 // Get the scale factor
                 scale = s->scale_factors[ch][band >> 1][band & 1];
 
                 dequantize(s->x96_subband_samples[ch][band] + ofs,
                            audio, step_size, scale, 0);
             }
         }
 
         // DSYNC
         if ((ssf == s->nsubsubframes[sf] - 1 || s->sync_ssf) && get_bits(&s->gb, 16) != 0xffff) {
             av_log(s->avctx, AV_LOG_ERROR, "X96-DSYNC check failed\n");
             return AVERROR_INVALIDDATA;
         }
 
         ofs += DCA_SUBBAND_SAMPLES;
     }
 
     // Inverse ADPCM
     for (ch = xch_base; ch < s->x96_nchannels; ch++) {
         inverse_adpcm(s->x96_subband_samples[ch], s->prediction_vq_index[ch],
                       s->prediction_mode[ch], s->x96_subband_start, s->nsubbands[ch],
                       *sub_pos, nsamples);
     }
 
     // Joint subband coding
     for (ch = xch_base; ch < s->x96_nchannels; ch++) {
         int src_ch = s->joint_intensity_index[ch] - 1;
         if (src_ch >= 0) {
             s->dcadsp->decode_joint(s->x96_subband_samples[ch], s->x96_subband_samples[src_ch],
                                     s->joint_scale_factors[ch], s->nsubbands[ch],
                                     s->nsubbands[src_ch], *sub_pos, nsamples);
         }
     }
 
     // Advance subband sample pointer for the next subframe
     *sub_pos = ofs;
     return 0;
 }
 
 static void erase_x96_adpcm_history(DCACoreDecoder *s)
 {
     int ch, band;
 
     // Erase ADPCM history from previous frame if
     // predictor history switch was disabled
     for (ch = 0; ch < DCA_CHANNELS; ch++)
         for (band = 0; band < DCA_SUBBANDS_X96; band++)
             AV_ZERO128(s->x96_subband_samples[ch][band] - DCA_ADPCM_COEFFS);

 
     emms_c();

 }
 
 static int alloc_x96_sample_buffer(DCACoreDecoder *s)
 {
     int nchsamples = DCA_ADPCM_COEFFS + s->npcmblocks;
     int nframesamples = nchsamples * DCA_CHANNELS * DCA_SUBBANDS_X96;
     unsigned int size = s->x96_subband_size;
     int ch, band;
 
     // Reallocate subband sample buffer
     av_fast_mallocz(&s->x96_subband_buffer, &s->x96_subband_size,
                     nframesamples * sizeof(int32_t));
     if (!s->x96_subband_buffer)
         return AVERROR(ENOMEM);
 
     if (size != s->x96_subband_size) {
         for (ch = 0; ch < DCA_CHANNELS; ch++)
             for (band = 0; band < DCA_SUBBANDS_X96; band++)
                 s->x96_subband_samples[ch][band] = s->x96_subband_buffer +
                     (ch * DCA_SUBBANDS_X96 + band) * nchsamples + DCA_ADPCM_COEFFS;
     }
 
     if (!s->predictor_history)
         erase_x96_adpcm_history(s);
 
     return 0;
 }
 
 static int parse_x96_subframe_header(DCACoreDecoder *s, int xch_base)
 {
     int ch, band, ret;
 
     if (get_bits_left(&s->gb) < 0)
         return AVERROR_INVALIDDATA;
 
     // Prediction mode
     for (ch = xch_base; ch < s->x96_nchannels; ch++)
         for (band = s->x96_subband_start; band < s->nsubbands[ch]; band++)
             s->prediction_mode[ch][band] = get_bits1(&s->gb);
 
     // Prediction coefficients VQ address
     for (ch = xch_base; ch < s->x96_nchannels; ch++)
         for (band = s->x96_subband_start; band < s->nsubbands[ch]; band++)
             if (s->prediction_mode[ch][band])
                 s->prediction_vq_index[ch][band] = get_bits(&s->gb, 12);
 
     // Bit allocation index
     for (ch = xch_base; ch < s->x96_nchannels; ch++) {
         int sel = s->bit_allocation_sel[ch];
         int abits = 0;
 
         for (band = s->x96_subband_start; band < s->nsubbands[ch]; band++) {
             // If Huffman code was used, the difference of abits was encoded
             if (sel < 7)

                 abits += dca_get_vlc(&s->gb, &ff_dca_vlc_quant_index[5 + 2 * s->x96_high_res], sel);

             else
                 abits = get_bits(&s->gb, 3 + s->x96_high_res);
 
             if (abits < 0 || abits > 7 + 8 * s->x96_high_res) {
                 av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 bit allocation index\n");
                 return AVERROR_INVALIDDATA;
             }
 
             s->bit_allocation[ch][band] = abits;
         }
     }
 
     // Scale factors
     for (ch = xch_base; ch < s->x96_nchannels; ch++) {
         int sel = s->scale_factor_sel[ch];
         int scale_index = 0;
 
         // Extract scales for subbands which are transmitted even for
         // unallocated subbands
         for (band = s->x96_subband_start; band < s->nsubbands[ch]; band++) {
             if ((ret = parse_scale(s, &scale_index, sel)) < 0)
                 return ret;
             s->scale_factors[ch][band >> 1][band & 1] = ret;
         }
     }
 
     // Joint subband codebook select
     for (ch = xch_base; ch < s->x96_nchannels; ch++) {
         if (s->joint_intensity_index[ch]) {
             s->joint_scale_sel[ch] = get_bits(&s->gb, 3);
             if (s->joint_scale_sel[ch] == 7) {
                 av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 joint scale factor code book\n");
                 return AVERROR_INVALIDDATA;
             }
         }
     }
 
     // Scale factors for joint subband coding
     for (ch = xch_base; ch < s->x96_nchannels; ch++) {
         int src_ch = s->joint_intensity_index[ch] - 1;
         if (src_ch >= 0) {
             int sel = s->joint_scale_sel[ch];
             for (band = s->nsubbands[ch]; band < s->nsubbands[src_ch]; band++) {
                 if ((ret = parse_joint_scale(s, sel)) < 0)
                     return ret;
                 s->joint_scale_factors[ch][band] = ret;
             }
         }
     }
 
     // Side information CRC check word
     if (s->crc_present)
         skip_bits(&s->gb, 16);
 
     return 0;
 }
 
 static int parse_x96_coding_header(DCACoreDecoder *s, int exss, int xch_base)
 {
     int n, ch, header_size = 0, header_pos = get_bits_count(&s->gb);
 
     if (get_bits_left(&s->gb) < 0)
         return AVERROR_INVALIDDATA;
 
     if (exss) {
         // Channel set header length
         header_size = get_bits(&s->gb, 7) + 1;
 
         // Check CRC
         if (s->x96_crc_present

             && ff_dca_check_crc(s->avctx, &s->gb, header_pos, header_pos + header_size * 8)) {

             av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 channel set header checksum\n");
             return AVERROR_INVALIDDATA;
         }
     }
 
     // High resolution flag
     s->x96_high_res = get_bits1(&s->gb);
 
     // First encoded subband
     if (s->x96_rev_no < 8) {
         s->x96_subband_start = get_bits(&s->gb, 5);
         if (s->x96_subband_start > 27) {
             av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 subband start index (%d)\n", s->x96_subband_start);
             return AVERROR_INVALIDDATA;
         }
     } else {
         s->x96_subband_start = DCA_SUBBANDS;
     }
 
     // Subband activity count
     for (ch = xch_base; ch < s->x96_nchannels; ch++) {
         s->nsubbands[ch] = get_bits(&s->gb, 6) + 1;
         if (s->nsubbands[ch] < DCA_SUBBANDS) {
             av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 subband activity count (%d)\n", s->nsubbands[ch]);
             return AVERROR_INVALIDDATA;
         }
     }
 
     // Joint intensity coding index
     for (ch = xch_base; ch < s->x96_nchannels; ch++) {
         if ((n = get_bits(&s->gb, 3)) && xch_base)
             n += xch_base - 1;
         if (n > s->x96_nchannels) {
             av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 joint intensity coding index\n");
             return AVERROR_INVALIDDATA;
         }
         s->joint_intensity_index[ch] = n;
     }
 
     // Scale factor code book
     for (ch = xch_base; ch < s->x96_nchannels; ch++) {
         s->scale_factor_sel[ch] = get_bits(&s->gb, 3);
         if (s->scale_factor_sel[ch] >= 6) {
             av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 scale factor code book\n");
             return AVERROR_INVALIDDATA;
         }
     }
 
     // Bit allocation quantizer select
     for (ch = xch_base; ch < s->x96_nchannels; ch++)
         s->bit_allocation_sel[ch] = get_bits(&s->gb, 3);
 
     // Quantization index codebook select
     for (n = 0; n < 6 + 4 * s->x96_high_res; n++)
         for (ch = xch_base; ch < s->x96_nchannels; ch++)

             s->quant_index_sel[ch][n] = get_bits(&s->gb, ff_dca_quant_index_sel_nbits[n]);

 
     if (exss) {
         // Reserved
         // Byte align
         // CRC16 of channel set header
         if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8)) {
             av_log(s->avctx, AV_LOG_ERROR, "Read past end of X96 channel set header\n");
             return AVERROR_INVALIDDATA;
         }
     } else {
         if (s->crc_present)
             skip_bits(&s->gb, 16);
     }
 
     return 0;
 }
 
 static int parse_x96_frame_data(DCACoreDecoder *s, int exss, int xch_base)
 {
     int sf, ch, ret, band, sub_pos;
 
     if ((ret = parse_x96_coding_header(s, exss, xch_base)) < 0)
         return ret;
 
     for (sf = 0, sub_pos = 0; sf < s->nsubframes; sf++) {
         if ((ret = parse_x96_subframe_header(s, xch_base)) < 0)
             return ret;
         if ((ret = parse_x96_subframe_audio(s, sf, xch_base, &sub_pos)) < 0)
             return ret;
     }
 
     for (ch = xch_base; ch < s->x96_nchannels; ch++) {
         // Determine number of active subbands for this channel
         int nsubbands = s->nsubbands[ch];
         if (s->joint_intensity_index[ch])
             nsubbands = FFMAX(nsubbands, s->nsubbands[s->joint_intensity_index[ch] - 1]);
 
         // Update history for ADPCM and clear inactive subbands
         for (band = 0; band < DCA_SUBBANDS_X96; band++) {
             int32_t *samples = s->x96_subband_samples[ch][band] - DCA_ADPCM_COEFFS;
             if (band >= s->x96_subband_start && band < nsubbands)
                 AV_COPY128(samples, samples + s->npcmblocks);
             else
                 memset(samples, 0, (DCA_ADPCM_COEFFS + s->npcmblocks) * sizeof(int32_t));
         }
     }
 

     emms_c();
 

     return 0;
 }
 
 static int parse_x96_frame(DCACoreDecoder *s)
 {
     int ret;
 
     // Revision number
     s->x96_rev_no = get_bits(&s->gb, 4);
     if (s->x96_rev_no < 1 || s->x96_rev_no > 8) {
         av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 revision (%d)\n", s->x96_rev_no);
         return AVERROR_INVALIDDATA;
     }
 
     s->x96_crc_present = 0;
     s->x96_nchannels = s->nchannels;
 
     if ((ret = alloc_x96_sample_buffer(s)) < 0)
         return ret;
 
     if ((ret = parse_x96_frame_data(s, 0, 0)) < 0)
         return ret;
 
     // Seek to the end of core frame
     if (ff_dca_seek_bits(&s->gb, s->frame_size * 8)) {
         av_log(s->avctx, AV_LOG_ERROR, "Read past end of X96 frame\n");
         return AVERROR_INVALIDDATA;
     }
 
     return 0;
 }
 
 static int parse_x96_frame_exss(DCACoreDecoder *s)
 {
     int     x96_frame_size[DCA_EXSS_CHSETS_MAX];
     int     x96_nchannels[DCA_EXSS_CHSETS_MAX];
     int     x96_nchsets, x96_base_ch;
     int     i, ret, header_size, header_pos = get_bits_count(&s->gb);
 
     // X96 sync word
     if (get_bits_long(&s->gb, 32) != DCA_SYNCWORD_X96) {
         av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 sync word\n");
         return AVERROR_INVALIDDATA;
     }
 
     // X96 frame header length
     header_size = get_bits(&s->gb, 6) + 1;
 
     // Check X96 frame header CRC

     if (ff_dca_check_crc(s->avctx, &s->gb, header_pos + 32, header_pos + header_size * 8)) {

         av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 frame header checksum\n");
         return AVERROR_INVALIDDATA;
     }
 
     // Revision number
     s->x96_rev_no = get_bits(&s->gb, 4);
     if (s->x96_rev_no < 1 || s->x96_rev_no > 8) {
         av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 revision (%d)\n", s->x96_rev_no);
         return AVERROR_INVALIDDATA;
     }
 
     // CRC presence flag for channel set header
     s->x96_crc_present = get_bits1(&s->gb);
 
     // Number of channel sets
     x96_nchsets = get_bits(&s->gb, 2) + 1;
 
     // Channel set data byte size
     for (i = 0; i < x96_nchsets; i++)
         x96_frame_size[i] = get_bits(&s->gb, 12) + 1;
 
     // Number of channels in channel set
     for (i = 0; i < x96_nchsets; i++)
         x96_nchannels[i] = get_bits(&s->gb, 3) + 1;
 
     // Reserved
     // Byte align
     // CRC16 of X96 frame header
     if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8)) {
         av_log(s->avctx, AV_LOG_ERROR, "Read past end of X96 frame header\n");
         return AVERROR_INVALIDDATA;
     }
 
     if ((ret = alloc_x96_sample_buffer(s)) < 0)
         return ret;
 
     // Channel set data

     s->x96_nchannels = 0;

     for (i = 0, x96_base_ch = 0; i < x96_nchsets; i++) {
         header_pos = get_bits_count(&s->gb);
 
         if (x96_base_ch + x96_nchannels[i] <= s->nchannels) {
             s->x96_nchannels = x96_base_ch + x96_nchannels[i];
             if ((ret = parse_x96_frame_data(s, 1, x96_base_ch)) < 0)
                 return ret;
         }
 
         x96_base_ch += x96_nchannels[i];
 
         if (ff_dca_seek_bits(&s->gb, header_pos + x96_frame_size[i] * 8)) {
             av_log(s->avctx, AV_LOG_ERROR, "Read past end of X96 channel set\n");
             return AVERROR_INVALIDDATA;
         }
     }
 
     return 0;
 }
 
 static int parse_aux_data(DCACoreDecoder *s)
 {
     int aux_pos;
 
     if (get_bits_left(&s->gb) < 0)
         return AVERROR_INVALIDDATA;
 
     // Auxiliary data byte count (can't be trusted)
     skip_bits(&s->gb, 6);
 
     // 4-byte align
     skip_bits_long(&s->gb, -get_bits_count(&s->gb) & 31);
 
     // Auxiliary data sync word
     if (get_bits_long(&s->gb, 32) != DCA_SYNCWORD_REV1AUX) {
         av_log(s->avctx, AV_LOG_ERROR, "Invalid auxiliary data sync word\n");
         return AVERROR_INVALIDDATA;
     }
 
     aux_pos = get_bits_count(&s->gb);
 
     // Auxiliary decode time stamp flag
     if (get_bits1(&s->gb))
         skip_bits_long(&s->gb, 47);
 
     // Auxiliary dynamic downmix flag
     if (s->prim_dmix_embedded = get_bits1(&s->gb)) {
         int i, m, n;
 
         // Auxiliary primary channel downmix type
         s->prim_dmix_type = get_bits(&s->gb, 3);
         if (s->prim_dmix_type >= DCA_DMIX_TYPE_COUNT) {
             av_log(s->avctx, AV_LOG_ERROR, "Invalid primary channel set downmix type\n");
             return AVERROR_INVALIDDATA;
         }
 
         // Size of downmix coefficients matrix
         m = ff_dca_dmix_primary_nch[s->prim_dmix_type];
         n = ff_dca_channels[s->audio_mode] + !!s->lfe_present;
 
         // Dynamic downmix code coefficients
         for (i = 0; i < m * n; i++) {
             int code = get_bits(&s->gb, 9);
             int sign = (code >> 8) - 1;
             unsigned int index = code & 0xff;
             if (index >= FF_DCA_DMIXTABLE_SIZE) {
                 av_log(s->avctx, AV_LOG_ERROR, "Invalid downmix coefficient index\n");
                 return AVERROR_INVALIDDATA;
             }
             s->prim_dmix_coeff[i] = (ff_dca_dmixtable[index] ^ sign) - sign;
         }
     }
 
     // Byte align
     skip_bits(&s->gb, -get_bits_count(&s->gb) & 7);
 
     // CRC16 of auxiliary data
     skip_bits(&s->gb, 16);
 
     // Check CRC

     if (ff_dca_check_crc(s->avctx, &s->gb, aux_pos, get_bits_count(&s->gb))) {

         av_log(s->avctx, AV_LOG_ERROR, "Invalid auxiliary data checksum\n");
         return AVERROR_INVALIDDATA;
     }
 
     return 0;
 }
 
 static int parse_optional_info(DCACoreDecoder *s)
 {
     DCAContext *dca = s->avctx->priv_data;
     int ret = -1;
 
     // Time code stamp
     if (s->ts_present)
         skip_bits_long(&s->gb, 32);
 
     // Auxiliary data
     if (s->aux_present && (ret = parse_aux_data(s)) < 0
         && (s->avctx->err_recognition & AV_EF_EXPLODE))
         return ret;
 
     if (ret < 0)
         s->prim_dmix_embedded = 0;
 
     // Core extensions
     if (s->ext_audio_present && !dca->core_only) {
         int sync_pos = FFMIN(s->frame_size / 4, s->gb.size_in_bits / 32) - 1;
         int last_pos = get_bits_count(&s->gb) / 32;
         int size, dist;
 
         // Search for extension sync words aligned on 4-byte boundary. Search
         // must be done backwards from the end of core frame to work around
         // sync word aliasing issues.
         switch (s->ext_audio_type) {
         case EXT_AUDIO_XCH:
             if (dca->request_channel_layout)
                 break;
 
             // The distance between XCH sync word and end of the core frame
             // must be equal to XCH frame size. Off by one error is allowed for
             // compatibility with legacy bitstreams. Minimum XCH frame size is
             // 96 bytes. AMODE and PCHS are further checked to reduce
             // probability of alias sync detection.
             for (; sync_pos >= last_pos; sync_pos--) {
                 if (AV_RB32(s->gb.buffer + sync_pos * 4) == DCA_SYNCWORD_XCH) {
                     s->gb.index = (sync_pos + 1) * 32;
                     size = get_bits(&s->gb, 10) + 1;
                     dist = s->frame_size - sync_pos * 4;
                     if (size >= 96
                         && (size == dist || size - 1 == dist)
                         && get_bits(&s->gb, 7) == 0x08) {
                         s->xch_pos = get_bits_count(&s->gb);
                         break;
                     }
                 }
             }
 

             if (!s->xch_pos) {

                 av_log(s->avctx, AV_LOG_ERROR, "XCH sync word not found\n");

                 if (s->avctx->err_recognition & AV_EF_EXPLODE)
                     return AVERROR_INVALIDDATA;

             }
             break;
 
         case EXT_AUDIO_X96:
             // The distance between X96 sync word and end of the core frame
             // must be equal to X96 frame size. Minimum X96 frame size is 96
             // bytes.
             for (; sync_pos >= last_pos; sync_pos--) {
                 if (AV_RB32(s->gb.buffer + sync_pos * 4) == DCA_SYNCWORD_X96) {
                     s->gb.index = (sync_pos + 1) * 32;
                     size = get_bits(&s->gb, 12) + 1;
                     dist = s->frame_size - sync_pos * 4;
                     if (size >= 96 && size == dist) {
                         s->x96_pos = get_bits_count(&s->gb);
                         break;
                     }
                 }
             }
 

             if (!s->x96_pos) {

                 av_log(s->avctx, AV_LOG_ERROR, "X96 sync word not found\n");

                 if (s->avctx->err_recognition & AV_EF_EXPLODE)
                     return AVERROR_INVALIDDATA;

             }
             break;
 
         case EXT_AUDIO_XXCH:
             if (dca->request_channel_layout)
                 break;
 
             // XXCH frame header CRC must be valid. Minimum XXCH frame header
             // size is 11 bytes.
             for (; sync_pos >= last_pos; sync_pos--) {
                 if (AV_RB32(s->gb.buffer + sync_pos * 4) == DCA_SYNCWORD_XXCH) {
                     s->gb.index = (sync_pos + 1) * 32;
                     size = get_bits(&s->gb, 6) + 1;

                     dist = s->gb.size_in_bits / 8 - sync_pos * 4;
                     if (size >= 11 && size <= dist &&
                         !av_crc(dca->crctab, 0xffff, s->gb.buffer +
                                 (sync_pos + 1) * 4, size - 4)) {

                         s->xxch_pos = sync_pos * 32;
                         break;
                     }
                 }
             }
 

             if (!s->xxch_pos) {

                 av_log(s->avctx, AV_LOG_ERROR, "XXCH sync word not found\n");

                 if (s->avctx->err_recognition & AV_EF_EXPLODE)
                     return AVERROR_INVALIDDATA;

             }
             break;
         }
     }
 
     return 0;
 }
 
 int ff_dca_core_parse(DCACoreDecoder *s, uint8_t *data, int size)
 {
     int ret;
 
     s->ext_audio_mask = 0;
     s->xch_pos = s->xxch_pos = s->x96_pos = 0;
 
     if ((ret = init_get_bits8(&s->gb, data, size)) < 0)
         return ret;
 
     skip_bits_long(&s->gb, 32);
     if ((ret = parse_frame_header(s)) < 0)
         return ret;
     if ((ret = alloc_sample_buffer(s)) < 0)
         return ret;
     if ((ret = parse_frame_data(s, HEADER_CORE, 0)) < 0)
         return ret;
     if ((ret = parse_optional_info(s)) < 0)
         return ret;
 
     // Workaround for DTS in WAV

     if (s->frame_size > size && s->frame_size < size + 4)

         s->frame_size = size;
 
     if (ff_dca_seek_bits(&s->gb, s->frame_size * 8)) {
         av_log(s->avctx, AV_LOG_ERROR, "Read past end of core frame\n");
         if (s->avctx->err_recognition & AV_EF_EXPLODE)
             return AVERROR_INVALIDDATA;
     }
 
     return 0;
 }
 
 int ff_dca_core_parse_exss(DCACoreDecoder *s, uint8_t *data, DCAExssAsset *asset)
 {
     AVCodecContext *avctx = s->avctx;
     DCAContext *dca = avctx->priv_data;
     GetBitContext gb = s->gb;
     int exss_mask = asset ? asset->extension_mask : 0;
     int ret = 0, ext = 0;
 
     // Parse (X)XCH unless downmixing
     if (!dca->request_channel_layout) {
         if (exss_mask & DCA_EXSS_XXCH) {
             if ((ret = init_get_bits8(&s->gb, data + asset->xxch_offset, asset->xxch_size)) < 0)
                 return ret;
             ret = parse_xxch_frame(s);
             ext = DCA_EXSS_XXCH;
         } else if (s->xxch_pos) {
             s->gb.index = s->xxch_pos;
             ret = parse_xxch_frame(s);
             ext = DCA_CSS_XXCH;
         } else if (s->xch_pos) {
             s->gb.index = s->xch_pos;
             ret = parse_xch_frame(s);
             ext = DCA_CSS_XCH;
         }
 
         // Revert to primary channel set in case (X)XCH parsing fails
         if (ret < 0) {
             if (avctx->err_recognition & AV_EF_EXPLODE)
                 return ret;
             s->nchannels = ff_dca_channels[s->audio_mode];
             s->ch_mask = audio_mode_ch_mask[s->audio_mode];
             if (s->lfe_present)
                 s->ch_mask |= DCA_SPEAKER_MASK_LFE1;
         } else {
             s->ext_audio_mask |= ext;
         }
     }
 
     // Parse XBR
     if (exss_mask & DCA_EXSS_XBR) {
         if ((ret = init_get_bits8(&s->gb, data + asset->xbr_offset, asset->xbr_size)) < 0)
             return ret;
         if ((ret = parse_xbr_frame(s)) < 0) {
             if (avctx->err_recognition & AV_EF_EXPLODE)
                 return ret;
         } else {
             s->ext_audio_mask |= DCA_EXSS_XBR;
         }
     }
 
     // Parse X96 unless decoding XLL
     if (!(dca->packet & DCA_PACKET_XLL)) {
         if (exss_mask & DCA_EXSS_X96) {
             if ((ret = init_get_bits8(&s->gb, data + asset->x96_offset, asset->x96_size)) < 0)
                 return ret;
             if ((ret = parse_x96_frame_exss(s)) < 0) {
                 if (ret == AVERROR(ENOMEM) || (avctx->err_recognition & AV_EF_EXPLODE))
                     return ret;
             } else {
                 s->ext_audio_mask |= DCA_EXSS_X96;
             }
         } else if (s->x96_pos) {
             s->gb = gb;
             s->gb.index = s->x96_pos;
             if ((ret = parse_x96_frame(s)) < 0) {
                 if (ret == AVERROR(ENOMEM) || (avctx->err_recognition & AV_EF_EXPLODE))
                     return ret;
             } else {
                 s->ext_audio_mask |= DCA_CSS_X96;
             }
         }
     }
 
     return 0;
 }
 
 static int map_prm_ch_to_spkr(DCACoreDecoder *s, int ch)
 {
     int pos, spkr;
 
     // Try to map this channel to core first
     pos = ff_dca_channels[s->audio_mode];
     if (ch < pos) {
         spkr = prm_ch_to_spkr_map[s->audio_mode][ch];
         if (s->ext_audio_mask & (DCA_CSS_XXCH | DCA_EXSS_XXCH)) {
             if (s->xxch_core_mask & (1U << spkr))
                 return spkr;
             if (spkr == DCA_SPEAKER_Ls && (s->xxch_core_mask & DCA_SPEAKER_MASK_Lss))
                 return DCA_SPEAKER_Lss;
             if (spkr == DCA_SPEAKER_Rs && (s->xxch_core_mask & DCA_SPEAKER_MASK_Rss))
                 return DCA_SPEAKER_Rss;
             return -1;
         }
         return spkr;
     }
 
     // Then XCH
     if ((s->ext_audio_mask & DCA_CSS_XCH) && ch == pos)
         return DCA_SPEAKER_Cs;
 
     // Then XXCH
     if (s->ext_audio_mask & (DCA_CSS_XXCH | DCA_EXSS_XXCH)) {
         for (spkr = DCA_SPEAKER_Cs; spkr < s->xxch_mask_nbits; spkr++)
             if (s->xxch_spkr_mask & (1U << spkr))
                 if (pos++ == ch)
                     return spkr;
     }
 
     // No mapping
     return -1;
 }
 
 static void erase_dsp_history(DCACoreDecoder *s)
 {
     memset(s->dcadsp_data, 0, sizeof(s->dcadsp_data));
     s->output_history_lfe_fixed = 0;
     s->output_history_lfe_float = 0;
 }
 
 static void set_filter_mode(DCACoreDecoder *s, int mode)
 {
     if (s->filter_mode != mode) {
         erase_dsp_history(s);
         s->filter_mode = mode;
     }
 }
 
 int ff_dca_core_filter_fixed(DCACoreDecoder *s, int x96_synth)
 {
     int n, ch, spkr, nsamples, x96_nchannels = 0;
     const int32_t *filter_coeff;
     int32_t *ptr;
 
     // Externally set x96_synth flag implies that X96 synthesis should be
     // enabled, yet actual X96 subband data should be discarded. This is a
     // special case for lossless residual decoder that ignores X96 data if
     // present.
     if (!x96_synth && (s->ext_audio_mask & (DCA_CSS_X96 | DCA_EXSS_X96))) {
         x96_nchannels = s->x96_nchannels;
         x96_synth = 1;
     }
     if (x96_synth < 0)
         x96_synth = 0;
 
     s->output_rate = s->sample_rate << x96_synth;
     s->npcmsamples = nsamples = (s->npcmblocks * DCA_PCMBLOCK_SAMPLES) << x96_synth;
 
     // Reallocate PCM output buffer
     av_fast_malloc(&s->output_buffer, &s->output_size,
                    nsamples * av_popcount(s->ch_mask) * sizeof(int32_t));
     if (!s->output_buffer)
         return AVERROR(ENOMEM);
 
     ptr = (int32_t *)s->output_buffer;
     for (spkr = 0; spkr < DCA_SPEAKER_COUNT; spkr++) {
         if (s->ch_mask & (1U << spkr)) {
             s->output_samples[spkr] = ptr;
             ptr += nsamples;
         } else {
             s->output_samples[spkr] = NULL;
         }
     }
 
     // Handle change of filtering mode
     set_filter_mode(s, x96_synth | DCA_FILTER_MODE_FIXED);
 
     // Select filter
     if (x96_synth)
         filter_coeff = ff_dca_fir_64bands_fixed;
     else if (s->filter_perfect)
         filter_coeff = ff_dca_fir_32bands_perfect_fixed;
     else
         filter_coeff = ff_dca_fir_32bands_nonperfect_fixed;
 
     // Filter primary channels
     for (ch = 0; ch < s->nchannels; ch++) {
         // Map this primary channel to speaker
         spkr = map_prm_ch_to_spkr(s, ch);
         if (spkr < 0)
             return AVERROR(EINVAL);
 
         // Filter bank reconstruction
         s->dcadsp->sub_qmf_fixed[x96_synth](
             &s->synth,
             &s->dcadct,
             s->output_samples[spkr],
             s->subband_samples[ch],
             ch < x96_nchannels ? s->x96_subband_samples[ch] : NULL,
             s->dcadsp_data[ch].u.fix.hist1,
             &s->dcadsp_data[ch].offset,
             s->dcadsp_data[ch].u.fix.hist2,
             filter_coeff,
             s->npcmblocks);
     }
 
     // Filter LFE channel
     if (s->lfe_present) {
         int32_t *samples = s->output_samples[DCA_SPEAKER_LFE1];
         int nlfesamples = s->npcmblocks >> 1;
 
         // Check LFF
         if (s->lfe_present == LFE_FLAG_128) {
             av_log(s->avctx, AV_LOG_ERROR, "Fixed point mode doesn't support LFF=1\n");
             return AVERROR(EINVAL);
         }
 
         // Offset intermediate buffer for X96
         if (x96_synth)
             samples += nsamples / 2;
 
         // Interpolate LFE channel
         s->dcadsp->lfe_fir_fixed(samples, s->lfe_samples + DCA_LFE_HISTORY,
                                  ff_dca_lfe_fir_64_fixed, s->npcmblocks);
 
         if (x96_synth) {
             // Filter 96 kHz oversampled LFE PCM to attenuate high frequency
             // (47.6 - 48.0 kHz) components of interpolation image
             s->dcadsp->lfe_x96_fixed(s->output_samples[DCA_SPEAKER_LFE1],
                                      samples, &s->output_history_lfe_fixed,
                                      nsamples / 2);
 
         }
 
         // Update LFE history
         for (n = DCA_LFE_HISTORY - 1; n >= 0; n--)
             s->lfe_samples[n] = s->lfe_samples[nlfesamples + n];
     }
 
     return 0;
 }
 
 static int filter_frame_fixed(DCACoreDecoder *s, AVFrame *frame)
 {
     AVCodecContext *avctx = s->avctx;
     DCAContext *dca = avctx->priv_data;
     int i, n, ch, ret, spkr, nsamples;
 
     // Don't filter twice when falling back from XLL
     if (!(dca->packet & DCA_PACKET_XLL) && (ret = ff_dca_core_filter_fixed(s, 0)) < 0)
         return ret;
 
     avctx->sample_rate = s->output_rate;
     avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
     avctx->bits_per_raw_sample = 24;
 
     frame->nb_samples = nsamples = s->npcmsamples;
     if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
         return ret;
 
     // Undo embedded XCH downmix
     if (s->es_format && (s->ext_audio_mask & DCA_CSS_XCH)
         && s->audio_mode >= AMODE_2F2R) {
         s->dcadsp->dmix_sub_xch(s->output_samples[DCA_SPEAKER_Ls],
                                 s->output_samples[DCA_SPEAKER_Rs],
                                 s->output_samples[DCA_SPEAKER_Cs],
                                 nsamples);
 
     }
 
     // Undo embedded XXCH downmix
     if ((s->ext_audio_mask & (DCA_CSS_XXCH | DCA_EXSS_XXCH))
         && s->xxch_dmix_embedded) {
         int scale_inv   = s->xxch_dmix_scale_inv;
         int *coeff_ptr  = s->xxch_dmix_coeff;
         int xch_base    = ff_dca_channels[s->audio_mode];
         av_assert1(s->nchannels - xch_base <= DCA_XXCH_CHANNELS_MAX);
 
         // Undo embedded core downmix pre-scaling
         for (spkr = 0; spkr < s->xxch_mask_nbits; spkr++) {
             if (s->xxch_core_mask & (1U << spkr)) {
                 s->dcadsp->dmix_scale_inv(s->output_samples[spkr],
                                           scale_inv, nsamples);
             }
         }
 
         // Undo downmix
         for (ch = xch_base; ch < s->nchannels; ch++) {
             int src_spkr = map_prm_ch_to_spkr(s, ch);
             if (src_spkr < 0)
                 return AVERROR(EINVAL);
             for (spkr = 0; spkr < s->xxch_mask_nbits; spkr++) {
                 if (s->xxch_dmix_mask[ch - xch_base] & (1U << spkr)) {
                     int coeff = mul16(*coeff_ptr++, scale_inv);
                     if (coeff) {
                         s->dcadsp->dmix_sub(s->output_samples[spkr    ],
                                             s->output_samples[src_spkr],
                                             coeff, nsamples);
                     }
                 }
             }
         }
     }
 
     if (!(s->ext_audio_mask & (DCA_CSS_XXCH | DCA_CSS_XCH | DCA_EXSS_XXCH))) {
         // Front sum/difference decoding
         if ((s->sumdiff_front && s->audio_mode > AMODE_MONO)
             || s->audio_mode == AMODE_STEREO_SUMDIFF) {
             s->fixed_dsp->butterflies_fixed(s->output_samples[DCA_SPEAKER_L],
                                             s->output_samples[DCA_SPEAKER_R],
                                             nsamples);
         }
 
         // Surround sum/difference decoding
         if (s->sumdiff_surround && s->audio_mode >= AMODE_2F2R) {
             s->fixed_dsp->butterflies_fixed(s->output_samples[DCA_SPEAKER_Ls],
                                             s->output_samples[DCA_SPEAKER_Rs],
                                             nsamples);
         }
     }
 
     // Downmix primary channel set to stereo
     if (s->request_mask != s->ch_mask) {
         ff_dca_downmix_to_stereo_fixed(s->dcadsp,
                                        s->output_samples,
                                        s->prim_dmix_coeff,
                                        nsamples, s->ch_mask);
     }
 
     for (i = 0; i < avctx->channels; i++) {
         int32_t *samples = s->output_samples[s->ch_remap[i]];
         int32_t *plane = (int32_t *)frame->extended_data[i];
         for (n = 0; n < nsamples; n++)
             plane[n] = clip23(samples[n]) * (1 << 8);
     }
 
     return 0;
 }
 
 static int filter_frame_float(DCACoreDecoder *s, AVFrame *frame)
 {
     AVCodecContext *avctx = s->avctx;
     int x96_nchannels = 0, x96_synth = 0;
     int i, n, ch, ret, spkr, nsamples, nchannels;
     float *output_samples[DCA_SPEAKER_COUNT] = { NULL }, *ptr;
     const float *filter_coeff;
 
     if (s->ext_audio_mask & (DCA_CSS_X96 | DCA_EXSS_X96)) {
         x96_nchannels = s->x96_nchannels;
         x96_synth = 1;
     }
 
     avctx->sample_rate = s->sample_rate << x96_synth;
     avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
     avctx->bits_per_raw_sample = 0;
 
     frame->nb_samples = nsamples = (s->npcmblocks * DCA_PCMBLOCK_SAMPLES) << x96_synth;
     if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
         return ret;
 
     // Build reverse speaker to channel mapping
     for (i = 0; i < avctx->channels; i++)
         output_samples[s->ch_remap[i]] = (float *)frame->extended_data[i];
 
     // Allocate space for extra channels
     nchannels = av_popcount(s->ch_mask) - avctx->channels;
     if (nchannels > 0) {
         av_fast_malloc(&s->output_buffer, &s->output_size,
                        nsamples * nchannels * sizeof(float));
         if (!s->output_buffer)
             return AVERROR(ENOMEM);
 
         ptr = (float *)s->output_buffer;
         for (spkr = 0; spkr < DCA_SPEAKER_COUNT; spkr++) {
             if (!(s->ch_mask & (1U << spkr)))
                 continue;
             if (output_samples[spkr])
                 continue;
             output_samples[spkr] = ptr;
             ptr += nsamples;
         }
     }
 
     // Handle change of filtering mode
     set_filter_mode(s, x96_synth);
 
     // Select filter
     if (x96_synth)
         filter_coeff = ff_dca_fir_64bands;
     else if (s->filter_perfect)
         filter_coeff = ff_dca_fir_32bands_perfect;
     else
         filter_coeff = ff_dca_fir_32bands_nonperfect;
 
     // Filter primary channels
     for (ch = 0; ch < s->nchannels; ch++) {
         // Map this primary channel to speaker
         spkr = map_prm_ch_to_spkr(s, ch);
         if (spkr < 0)
             return AVERROR(EINVAL);
 
         // Filter bank reconstruction
         s->dcadsp->sub_qmf_float[x96_synth](
             &s->synth,
             &s->imdct[x96_synth],
             output_samples[spkr],
             s->subband_samples[ch],
             ch < x96_nchannels ? s->x96_subband_samples[ch] : NULL,
             s->dcadsp_data[ch].u.flt.hist1,
             &s->dcadsp_data[ch].offset,
             s->dcadsp_data[ch].u.flt.hist2,
             filter_coeff,
             s->npcmblocks,
             1.0f / (1 << (17 - x96_synth)));
     }
 
     // Filter LFE channel
     if (s->lfe_present) {
         int dec_select = (s->lfe_present == LFE_FLAG_128);
         float *samples = output_samples[DCA_SPEAKER_LFE1];
         int nlfesamples = s->npcmblocks >> (dec_select + 1);
 
         // Offset intermediate buffer for X96
         if (x96_synth)
             samples += nsamples / 2;
 
         // Select filter
         if (dec_select)
             filter_coeff = ff_dca_lfe_fir_128;
         else
             filter_coeff = ff_dca_lfe_fir_64;
 
         // Interpolate LFE channel
         s->dcadsp->lfe_fir_float[dec_select](
             samples, s->lfe_samples + DCA_LFE_HISTORY,
             filter_coeff, s->npcmblocks);
 
         if (x96_synth) {
             // Filter 96 kHz oversampled LFE PCM to attenuate high frequency
             // (47.6 - 48.0 kHz) components of interpolation image
             s->dcadsp->lfe_x96_float(output_samples[DCA_SPEAKER_LFE1],
                                      samples, &s->output_history_lfe_float,
                                      nsamples / 2);
         }
 
         // Update LFE history
         for (n = DCA_LFE_HISTORY - 1; n >= 0; n--)
             s->lfe_samples[n] = s->lfe_samples[nlfesamples + n];
     }
 
     // Undo embedded XCH downmix
     if (s->es_format && (s->ext_audio_mask & DCA_CSS_XCH)
         && s->audio_mode >= AMODE_2F2R) {
         s->float_dsp->vector_fmac_scalar(output_samples[DCA_SPEAKER_Ls],
                                          output_samples[DCA_SPEAKER_Cs],
                                          -M_SQRT1_2, nsamples);
         s->float_dsp->vector_fmac_scalar(output_samples[DCA_SPEAKER_Rs],
                                          output_samples[DCA_SPEAKER_Cs],
                                          -M_SQRT1_2, nsamples);
     }
 
     // Undo embedded XXCH downmix
     if ((s->ext_audio_mask & (DCA_CSS_XXCH | DCA_EXSS_XXCH))
         && s->xxch_dmix_embedded) {
         float scale_inv = s->xxch_dmix_scale_inv * (1.0f / (1 << 16));
         int *coeff_ptr  = s->xxch_dmix_coeff;
         int xch_base    = ff_dca_channels[s->audio_mode];
         av_assert1(s->nchannels - xch_base <= DCA_XXCH_CHANNELS_MAX);
 
         // Undo downmix
         for (ch = xch_base; ch < s->nchannels; ch++) {
             int src_spkr = map_prm_ch_to_spkr(s, ch);
             if (src_spkr < 0)
                 return AVERROR(EINVAL);
             for (spkr = 0; spkr < s->xxch_mask_nbits; spkr++) {
                 if (s->xxch_dmix_mask[ch - xch_base] & (1U << spkr)) {
                     int coeff = *coeff_ptr++;
                     if (coeff) {
                         s->float_dsp->vector_fmac_scalar(output_samples[    spkr],
                                                          output_samples[src_spkr],
                                                          coeff * (-1.0f / (1 << 15)),
                                                          nsamples);
                     }
                 }
             }
         }
 
         // Undo embedded core downmix pre-scaling
         for (spkr = 0; spkr < s->xxch_mask_nbits; spkr++) {
             if (s->xxch_core_mask & (1U << spkr)) {
                 s->float_dsp->vector_fmul_scalar(output_samples[spkr],
                                                  output_samples[spkr],
                                                  scale_inv, nsamples);
             }
         }
     }
 
     if (!(s->ext_audio_mask & (DCA_CSS_XXCH | DCA_CSS_XCH | DCA_EXSS_XXCH))) {
         // Front sum/difference decoding
         if ((s->sumdiff_front && s->audio_mode > AMODE_MONO)
             || s->audio_mode == AMODE_STEREO_SUMDIFF) {
             s->float_dsp->butterflies_float(output_samples[DCA_SPEAKER_L],
                                             output_samples[DCA_SPEAKER_R],
                                             nsamples);
         }
 
         // Surround sum/difference decoding
         if (s->sumdiff_surround && s->audio_mode >= AMODE_2F2R) {
             s->float_dsp->butterflies_float(output_samples[DCA_SPEAKER_Ls],
                                             output_samples[DCA_SPEAKER_Rs],
                                             nsamples);
         }
     }
 
     // Downmix primary channel set to stereo
     if (s->request_mask != s->ch_mask) {
         ff_dca_downmix_to_stereo_float(s->float_dsp, output_samples,
                                        s->prim_dmix_coeff,
                                        nsamples, s->ch_mask);
     }
 
     return 0;
 }
 
 int ff_dca_core_filter_frame(DCACoreDecoder *s, AVFrame *frame)
 {
     AVCodecContext *avctx = s->avctx;
     DCAContext *dca = avctx->priv_data;
     DCAExssAsset *asset = &dca->exss.assets[0];
     enum AVMatrixEncoding matrix_encoding;
     int ret;
 
     // Handle downmixing to stereo request
     if (dca->request_channel_layout == DCA_SPEAKER_LAYOUT_STEREO
         && s->audio_mode > AMODE_MONO && s->prim_dmix_embedded
         && (s->prim_dmix_type == DCA_DMIX_TYPE_LoRo ||
             s->prim_dmix_type == DCA_DMIX_TYPE_LtRt))
         s->request_mask = DCA_SPEAKER_LAYOUT_STEREO;
     else
         s->request_mask = s->ch_mask;
     if (!ff_dca_set_channel_layout(avctx, s->ch_remap, s->request_mask))
         return AVERROR(EINVAL);
 
     // Force fixed point mode when falling back from XLL
     if ((avctx->flags & AV_CODEC_FLAG_BITEXACT) || ((dca->packet & DCA_PACKET_EXSS)
                                                     && (asset->extension_mask & DCA_EXSS_XLL)))
         ret = filter_frame_fixed(s, frame);
     else
         ret = filter_frame_float(s, frame);
     if (ret < 0)
         return ret;
 
     // Set profile, bit rate, etc
     if (s->ext_audio_mask & DCA_EXSS_MASK)
         avctx->profile = FF_PROFILE_DTS_HD_HRA;
     else if (s->ext_audio_mask & (DCA_CSS_XXCH | DCA_CSS_XCH))
         avctx->profile = FF_PROFILE_DTS_ES;
     else if (s->ext_audio_mask & DCA_CSS_X96)
         avctx->profile = FF_PROFILE_DTS_96_24;
     else
         avctx->profile = FF_PROFILE_DTS;
 
     if (s->bit_rate > 3 && !(s->ext_audio_mask & DCA_EXSS_MASK))
         avctx->bit_rate = s->bit_rate;
     else
         avctx->bit_rate = 0;
 
     if (s->audio_mode == AMODE_STEREO_TOTAL || (s->request_mask != s->ch_mask &&
                                                 s->prim_dmix_type == DCA_DMIX_TYPE_LtRt))
         matrix_encoding = AV_MATRIX_ENCODING_DOLBY;
     else
         matrix_encoding = AV_MATRIX_ENCODING_NONE;
     if ((ret = ff_side_data_update_matrix_encoding(frame, matrix_encoding)) < 0)
         return ret;
 
     return 0;
 }
 
 av_cold void ff_dca_core_flush(DCACoreDecoder *s)
 {
     if (s->subband_buffer) {
         erase_adpcm_history(s);
         memset(s->lfe_samples, 0, DCA_LFE_HISTORY * sizeof(int32_t));
     }
 
     if (s->x96_subband_buffer)
         erase_x96_adpcm_history(s);
 
     erase_dsp_history(s);
 }
 
 av_cold int ff_dca_core_init(DCACoreDecoder *s)
 {
     if (!(s->float_dsp = avpriv_float_dsp_alloc(0)))
         return -1;
     if (!(s->fixed_dsp = avpriv_alloc_fixed_dsp(0)))
         return -1;
 
     ff_dcadct_init(&s->dcadct);
     if (ff_mdct_init(&s->imdct[0], 6, 1, 1.0) < 0)
         return -1;
     if (ff_mdct_init(&s->imdct[1], 7, 1, 1.0) < 0)
         return -1;
     ff_synth_filter_init(&s->synth);
 
     s->x96_rand = 1;
     return 0;
 }
 
 av_cold void ff_dca_core_close(DCACoreDecoder *s)
 {
     av_freep(&s->float_dsp);
     av_freep(&s->fixed_dsp);
 
     ff_mdct_end(&s->imdct[0]);
     ff_mdct_end(&s->imdct[1]);
 
     av_freep(&s->subband_buffer);
     s->subband_size = 0;
 
     av_freep(&s->x96_subband_buffer);
     s->x96_subband_size = 0;
 
     av_freep(&s->output_buffer);
     s->output_size = 0;
 }