mirror of
https://github.com/nyanmisaka/ffmpeg-rockchip.git
synced 2025-10-26 18:30:52 +08:00
1be3d8a0cb
Also include channel_layout.h directly wherever used. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2448 lines
84 KiB
C
2448 lines
84 KiB
C
/*
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* Copyright (C) 2016 foo86
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/channel_layout.h"
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#include "dcaadpcm.h"
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#include "dcadec.h"
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#include "dcadata.h"
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#include "dcahuff.h"
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#include "dcamath.h"
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#include "dca_syncwords.h"
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#if ARCH_ARM
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#include "arm/dca.h"
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#endif
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enum HeaderType {
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HEADER_CORE,
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HEADER_XCH,
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HEADER_XXCH
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};
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static const int8_t prm_ch_to_spkr_map[DCA_AMODE_COUNT][5] = {
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{ DCA_SPEAKER_C, -1, -1, -1, -1 },
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{ DCA_SPEAKER_L, DCA_SPEAKER_R, -1, -1, -1 },
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{ DCA_SPEAKER_L, DCA_SPEAKER_R, -1, -1, -1 },
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{ DCA_SPEAKER_L, DCA_SPEAKER_R, -1, -1, -1 },
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{ DCA_SPEAKER_L, DCA_SPEAKER_R, -1, -1, -1 },
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{ DCA_SPEAKER_C, DCA_SPEAKER_L, DCA_SPEAKER_R , -1, -1 },
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{ DCA_SPEAKER_L, DCA_SPEAKER_R, DCA_SPEAKER_Cs, -1, -1 },
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{ DCA_SPEAKER_C, DCA_SPEAKER_L, DCA_SPEAKER_R , DCA_SPEAKER_Cs, -1 },
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{ DCA_SPEAKER_L, DCA_SPEAKER_R, DCA_SPEAKER_Ls, DCA_SPEAKER_Rs, -1 },
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{ DCA_SPEAKER_C, DCA_SPEAKER_L, DCA_SPEAKER_R, DCA_SPEAKER_Ls, DCA_SPEAKER_Rs }
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};
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static const uint8_t audio_mode_ch_mask[DCA_AMODE_COUNT] = {
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DCA_SPEAKER_LAYOUT_MONO,
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DCA_SPEAKER_LAYOUT_STEREO,
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DCA_SPEAKER_LAYOUT_STEREO,
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DCA_SPEAKER_LAYOUT_STEREO,
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DCA_SPEAKER_LAYOUT_STEREO,
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DCA_SPEAKER_LAYOUT_3_0,
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DCA_SPEAKER_LAYOUT_2_1,
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DCA_SPEAKER_LAYOUT_3_1,
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DCA_SPEAKER_LAYOUT_2_2,
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DCA_SPEAKER_LAYOUT_5POINT0
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};
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static const uint8_t block_code_nbits[7] = {
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7, 10, 12, 13, 15, 17, 19
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};
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static int dca_get_vlc(GetBitContext *s, DCAVLC *v, int i)
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{
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return get_vlc2(s, v->vlc[i].table, v->vlc[i].bits, v->max_depth) + v->offset;
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}
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static void get_array(GetBitContext *s, int32_t *array, int size, int n)
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{
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int i;
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for (i = 0; i < size; i++)
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array[i] = get_sbits(s, n);
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}
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// 5.3.1 - Bit stream header
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static int parse_frame_header(DCACoreDecoder *s)
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{
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DCACoreFrameHeader h = { 0 };
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int err = ff_dca_parse_core_frame_header(&h, &s->gb);
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if (err < 0) {
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switch (err) {
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case DCA_PARSE_ERROR_DEFICIT_SAMPLES:
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av_log(s->avctx, AV_LOG_ERROR, "Deficit samples are not supported\n");
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return h.normal_frame ? AVERROR_INVALIDDATA : AVERROR_PATCHWELCOME;
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case DCA_PARSE_ERROR_PCM_BLOCKS:
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av_log(s->avctx, AV_LOG_ERROR, "Unsupported number of PCM sample blocks (%d)\n", h.npcmblocks);
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return (h.npcmblocks < 6 || h.normal_frame) ? AVERROR_INVALIDDATA : AVERROR_PATCHWELCOME;
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case DCA_PARSE_ERROR_FRAME_SIZE:
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av_log(s->avctx, AV_LOG_ERROR, "Invalid core frame size (%d bytes)\n", h.frame_size);
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return AVERROR_INVALIDDATA;
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case DCA_PARSE_ERROR_AMODE:
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av_log(s->avctx, AV_LOG_ERROR, "Unsupported audio channel arrangement (%d)\n", h.audio_mode);
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return AVERROR_PATCHWELCOME;
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case DCA_PARSE_ERROR_SAMPLE_RATE:
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av_log(s->avctx, AV_LOG_ERROR, "Invalid core audio sampling frequency\n");
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return AVERROR_INVALIDDATA;
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case DCA_PARSE_ERROR_RESERVED_BIT:
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av_log(s->avctx, AV_LOG_ERROR, "Reserved bit set\n");
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return AVERROR_INVALIDDATA;
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case DCA_PARSE_ERROR_LFE_FLAG:
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av_log(s->avctx, AV_LOG_ERROR, "Invalid low frequency effects flag\n");
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return AVERROR_INVALIDDATA;
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case DCA_PARSE_ERROR_PCM_RES:
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av_log(s->avctx, AV_LOG_ERROR, "Invalid source PCM resolution\n");
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return AVERROR_INVALIDDATA;
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default:
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av_log(s->avctx, AV_LOG_ERROR, "Unknown core frame header error\n");
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return AVERROR_INVALIDDATA;
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}
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}
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s->crc_present = h.crc_present;
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s->npcmblocks = h.npcmblocks;
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s->frame_size = h.frame_size;
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s->audio_mode = h.audio_mode;
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s->sample_rate = avpriv_dca_sample_rates[h.sr_code];
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s->bit_rate = ff_dca_bit_rates[h.br_code];
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s->drc_present = h.drc_present;
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s->ts_present = h.ts_present;
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s->aux_present = h.aux_present;
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s->ext_audio_type = h.ext_audio_type;
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s->ext_audio_present = h.ext_audio_present;
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s->sync_ssf = h.sync_ssf;
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s->lfe_present = h.lfe_present;
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s->predictor_history = h.predictor_history;
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s->filter_perfect = h.filter_perfect;
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s->source_pcm_res = ff_dca_bits_per_sample[h.pcmr_code];
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s->es_format = h.pcmr_code & 1;
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s->sumdiff_front = h.sumdiff_front;
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s->sumdiff_surround = h.sumdiff_surround;
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return 0;
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}
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// 5.3.2 - Primary audio coding header
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static int parse_coding_header(DCACoreDecoder *s, enum HeaderType header, int xch_base)
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{
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int n, ch, nchannels, header_size = 0, header_pos = get_bits_count(&s->gb);
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unsigned int mask, index;
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if (get_bits_left(&s->gb) < 0)
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return AVERROR_INVALIDDATA;
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switch (header) {
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case HEADER_CORE:
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// Number of subframes
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s->nsubframes = get_bits(&s->gb, 4) + 1;
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// Number of primary audio channels
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s->nchannels = get_bits(&s->gb, 3) + 1;
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if (s->nchannels != ff_dca_channels[s->audio_mode]) {
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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);
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return AVERROR_INVALIDDATA;
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}
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av_assert1(s->nchannels <= DCA_CHANNELS - 2);
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s->ch_mask = audio_mode_ch_mask[s->audio_mode];
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// Add LFE channel if present
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if (s->lfe_present)
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s->ch_mask |= DCA_SPEAKER_MASK_LFE1;
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break;
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case HEADER_XCH:
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s->nchannels = ff_dca_channels[s->audio_mode] + 1;
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av_assert1(s->nchannels <= DCA_CHANNELS - 1);
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s->ch_mask |= DCA_SPEAKER_MASK_Cs;
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break;
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case HEADER_XXCH:
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// Channel set header length
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header_size = get_bits(&s->gb, 7) + 1;
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// Check CRC
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if (s->xxch_crc_present
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&& ff_dca_check_crc(s->avctx, &s->gb, header_pos, header_pos + header_size * 8)) {
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av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH channel set header checksum\n");
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return AVERROR_INVALIDDATA;
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}
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// Number of channels in a channel set
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nchannels = get_bits(&s->gb, 3) + 1;
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if (nchannels > DCA_XXCH_CHANNELS_MAX) {
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avpriv_request_sample(s->avctx, "%d XXCH channels", nchannels);
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return AVERROR_PATCHWELCOME;
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}
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s->nchannels = ff_dca_channels[s->audio_mode] + nchannels;
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av_assert1(s->nchannels <= DCA_CHANNELS);
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// Loudspeaker layout mask
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mask = get_bits_long(&s->gb, s->xxch_mask_nbits - DCA_SPEAKER_Cs);
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s->xxch_spkr_mask = mask << DCA_SPEAKER_Cs;
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if (av_popcount(s->xxch_spkr_mask) != nchannels) {
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av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH speaker layout mask (%#x)\n", s->xxch_spkr_mask);
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return AVERROR_INVALIDDATA;
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}
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if (s->xxch_core_mask & s->xxch_spkr_mask) {
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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);
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return AVERROR_INVALIDDATA;
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}
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// Combine core and XXCH masks together
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s->ch_mask = s->xxch_core_mask | s->xxch_spkr_mask;
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// Downmix coefficients present in stream
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if (get_bits1(&s->gb)) {
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int *coeff_ptr = s->xxch_dmix_coeff;
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// Downmix already performed by encoder
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s->xxch_dmix_embedded = get_bits1(&s->gb);
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// Downmix scale factor
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index = get_bits(&s->gb, 6) * 4 - FF_DCA_DMIXTABLE_OFFSET - 3;
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if (index >= FF_DCA_INV_DMIXTABLE_SIZE) {
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av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH downmix scale index (%d)\n", index);
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return AVERROR_INVALIDDATA;
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}
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s->xxch_dmix_scale_inv = ff_dca_inv_dmixtable[index];
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// Downmix channel mapping mask
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for (ch = 0; ch < nchannels; ch++) {
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mask = get_bits_long(&s->gb, s->xxch_mask_nbits);
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if ((mask & s->xxch_core_mask) != mask) {
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av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH downmix channel mapping mask (%#x)\n", mask);
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return AVERROR_INVALIDDATA;
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}
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s->xxch_dmix_mask[ch] = mask;
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}
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// Downmix coefficients
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for (ch = 0; ch < nchannels; ch++) {
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for (n = 0; n < s->xxch_mask_nbits; n++) {
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if (s->xxch_dmix_mask[ch] & (1U << n)) {
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int code = get_bits(&s->gb, 7);
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int sign = (code >> 6) - 1;
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if (code &= 63) {
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index = code * 4 - 3;
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if (index >= FF_DCA_DMIXTABLE_SIZE) {
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av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH downmix coefficient index (%d)\n", index);
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return AVERROR_INVALIDDATA;
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}
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*coeff_ptr++ = (ff_dca_dmixtable[index] ^ sign) - sign;
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} else {
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*coeff_ptr++ = 0;
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}
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}
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}
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}
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} else {
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s->xxch_dmix_embedded = 0;
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}
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break;
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}
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// Subband activity count
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for (ch = xch_base; ch < s->nchannels; ch++) {
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s->nsubbands[ch] = get_bits(&s->gb, 5) + 2;
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if (s->nsubbands[ch] > DCA_SUBBANDS) {
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av_log(s->avctx, AV_LOG_ERROR, "Invalid subband activity count\n");
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return AVERROR_INVALIDDATA;
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}
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}
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// High frequency VQ start subband
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for (ch = xch_base; ch < s->nchannels; ch++)
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s->subband_vq_start[ch] = get_bits(&s->gb, 5) + 1;
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// Joint intensity coding index
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for (ch = xch_base; ch < s->nchannels; ch++) {
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if ((n = get_bits(&s->gb, 3)) && header == HEADER_XXCH)
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n += xch_base - 1;
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if (n > s->nchannels) {
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av_log(s->avctx, AV_LOG_ERROR, "Invalid joint intensity coding index\n");
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return AVERROR_INVALIDDATA;
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}
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s->joint_intensity_index[ch] = n;
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}
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// Transient mode code book
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for (ch = xch_base; ch < s->nchannels; ch++)
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s->transition_mode_sel[ch] = get_bits(&s->gb, 2);
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// Scale factor code book
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for (ch = xch_base; ch < s->nchannels; ch++) {
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s->scale_factor_sel[ch] = get_bits(&s->gb, 3);
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if (s->scale_factor_sel[ch] == 7) {
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av_log(s->avctx, AV_LOG_ERROR, "Invalid scale factor code book\n");
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return AVERROR_INVALIDDATA;
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}
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}
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// Bit allocation quantizer select
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for (ch = xch_base; ch < s->nchannels; ch++) {
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s->bit_allocation_sel[ch] = get_bits(&s->gb, 3);
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if (s->bit_allocation_sel[ch] == 7) {
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av_log(s->avctx, AV_LOG_ERROR, "Invalid bit allocation quantizer select\n");
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return AVERROR_INVALIDDATA;
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}
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}
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// Quantization index codebook select
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for (n = 0; n < DCA_CODE_BOOKS; n++)
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for (ch = xch_base; ch < s->nchannels; ch++)
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s->quant_index_sel[ch][n] = get_bits(&s->gb, ff_dca_quant_index_sel_nbits[n]);
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// Scale factor adjustment index
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for (n = 0; n < DCA_CODE_BOOKS; n++)
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for (ch = xch_base; ch < s->nchannels; ch++)
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if (s->quant_index_sel[ch][n] < ff_dca_quant_index_group_size[n])
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s->scale_factor_adj[ch][n] = ff_dca_scale_factor_adj[get_bits(&s->gb, 2)];
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if (header == HEADER_XXCH) {
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// Reserved
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// Byte align
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// CRC16 of channel set header
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if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8)) {
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av_log(s->avctx, AV_LOG_ERROR, "Read past end of XXCH channel set header\n");
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return AVERROR_INVALIDDATA;
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}
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} else {
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// Audio header CRC check word
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if (s->crc_present)
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skip_bits(&s->gb, 16);
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}
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return 0;
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}
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static inline int parse_scale(DCACoreDecoder *s, int *scale_index, int sel)
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{
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const uint32_t *scale_table;
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unsigned int scale_size;
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// Select the root square table
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if (sel > 5) {
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scale_table = ff_dca_scale_factor_quant7;
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scale_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant7);
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} else {
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scale_table = ff_dca_scale_factor_quant6;
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scale_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant6);
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}
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// If Huffman code was used, the difference of scales was encoded
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if (sel < 5)
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*scale_index += dca_get_vlc(&s->gb, &ff_dca_vlc_scale_factor, sel);
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else
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*scale_index = get_bits(&s->gb, sel + 1);
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// Look up scale factor from the root square table
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if ((unsigned int)*scale_index >= scale_size) {
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av_log(s->avctx, AV_LOG_ERROR, "Invalid scale factor index\n");
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return AVERROR_INVALIDDATA;
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}
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return scale_table[*scale_index];
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}
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static inline int parse_joint_scale(DCACoreDecoder *s, int sel)
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{
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int scale_index;
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// Absolute value was encoded even when Huffman code was used
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if (sel < 5)
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scale_index = dca_get_vlc(&s->gb, &ff_dca_vlc_scale_factor, sel);
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else
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scale_index = get_bits(&s->gb, sel + 1);
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// Bias by 64
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scale_index += 64;
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// Look up joint scale factor
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if ((unsigned int)scale_index >= FF_ARRAY_ELEMS(ff_dca_joint_scale_factors)) {
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av_log(s->avctx, AV_LOG_ERROR, "Invalid joint scale factor index\n");
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return AVERROR_INVALIDDATA;
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}
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return ff_dca_joint_scale_factors[scale_index];
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}
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// 5.4.1 - Primary audio coding side information
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static int parse_subframe_header(DCACoreDecoder *s, int sf,
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enum HeaderType header, int xch_base)
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{
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int ch, band, ret;
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if (get_bits_left(&s->gb) < 0)
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return AVERROR_INVALIDDATA;
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if (header == HEADER_CORE) {
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// Subsubframe count
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s->nsubsubframes[sf] = get_bits(&s->gb, 2) + 1;
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// Partial subsubframe sample count
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skip_bits(&s->gb, 3);
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}
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// Prediction mode
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for (ch = xch_base; ch < s->nchannels; ch++)
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for (band = 0; band < s->nsubbands[ch]; band++)
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s->prediction_mode[ch][band] = get_bits1(&s->gb);
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// Prediction coefficients VQ address
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for (ch = xch_base; ch < s->nchannels; ch++)
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for (band = 0; band < s->nsubbands[ch]; band++)
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if (s->prediction_mode[ch][band])
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s->prediction_vq_index[ch][band] = get_bits(&s->gb, 12);
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// Bit allocation index
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for (ch = xch_base; ch < s->nchannels; ch++) {
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int sel = s->bit_allocation_sel[ch];
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for (band = 0; band < s->subband_vq_start[ch]; band++) {
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int abits;
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if (sel < 5)
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abits = dca_get_vlc(&s->gb, &ff_dca_vlc_bit_allocation, sel);
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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 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;
|
|
|
|
for (i = sb_start; i < sb_end; i++) {
|
|
if (prediction_mode[i]) {
|
|
const int pred_id = vq_index[i];
|
|
int32_t *ptr = subband_samples[i] + ofs;
|
|
for (j = 0; j < len; j++) {
|
|
int32_t x = ff_dcaadpcm_predict(pred_id, ptr + j - DCA_ADPCM_COEFFS);
|
|
ptr[j] = clip23(ptr[j] + x);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
// 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);
|
|
}
|
|
|
|
ff_dca_core_dequantize(s->subband_samples[ch][band] + ofs,
|
|
audio, step_size, scale, 0, DCA_SUBBAND_SAMPLES);
|
|
}
|
|
}
|
|
|
|
// 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];
|
|
|
|
ff_dca_core_dequantize(s->subband_samples[ch][band] + ofs,
|
|
audio, step_size, scale, 1, DCA_SUBBAND_SAMPLES);
|
|
}
|
|
}
|
|
|
|
// 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];
|
|
|
|
ff_dca_core_dequantize(s->x96_subband_samples[ch][band] + ofs,
|
|
audio, step_size, scale, 0, DCA_SUBBAND_SAMPLES);
|
|
}
|
|
}
|
|
|
|
// 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;
|
|
uint32_t w1, w2 = 0;
|
|
|
|
// 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 DCA_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--, w2 = w1) {
|
|
w1 = AV_RB32(s->gb.buffer + sync_pos * 4);
|
|
if (w1 == DCA_SYNCWORD_XCH) {
|
|
size = (w2 >> 22) + 1;
|
|
dist = s->frame_size - sync_pos * 4;
|
|
if (size >= 96
|
|
&& (size == dist || size - 1 == dist)
|
|
&& (w2 >> 15 & 0x7f) == 0x08) {
|
|
s->xch_pos = sync_pos * 32 + 49;
|
|
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 DCA_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--, w2 = w1) {
|
|
w1 = AV_RB32(s->gb.buffer + sync_pos * 4);
|
|
if (w1 == DCA_SYNCWORD_X96) {
|
|
size = (w2 >> 20) + 1;
|
|
dist = s->frame_size - sync_pos * 4;
|
|
if (size >= 96 && size == dist) {
|
|
s->x96_pos = sync_pos * 32 + 44;
|
|
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 DCA_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--, w2 = w1) {
|
|
w1 = AV_RB32(s->gb.buffer + sync_pos * 4);
|
|
if (w1 == DCA_SYNCWORD_XXCH) {
|
|
size = (w2 >> 26) + 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;
|
|
s->gb_in = s->gb;
|
|
|
|
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;
|
|
|
|
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;
|
|
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 = s->gb_in;
|
|
skip_bits_long(&s->gb, s->xxch_pos);
|
|
ret = parse_xxch_frame(s);
|
|
ext = DCA_CSS_XXCH;
|
|
} else if (s->xch_pos) {
|
|
s->gb = s->gb_in;
|
|
skip_bits_long(&s->gb, 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 = s->gb_in;
|
|
skip_bits_long(&s->gb, 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 == DCA_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 >= DCA_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 > DCA_AMODE_MONO)
|
|
|| s->audio_mode == DCA_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 >= DCA_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 == DCA_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 >= DCA_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 > DCA_AMODE_MONO)
|
|
|| s->audio_mode == DCA_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 >= DCA_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 > DCA_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 == DCA_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;
|
|
}
|