mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-11-21 10:55:51 +02:00
d1f558b362
Remove half-working attempt at supporting unchecked bitstream reader by always copying input data into intermediate buffer with large amount of padding at the end. Convert LBR decoder to checked bitstream reader. Convert dcadec_decode_frame() to parse input data directly if possible. Signed-off-by: James Almer <jamrial@gmail.com>
401 lines
13 KiB
C
401 lines
13 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/opt.h"
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#include "libavutil/channel_layout.h"
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#include "dcadec.h"
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#include "dcahuff.h"
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#include "dca_syncwords.h"
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#include "profiles.h"
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#define MIN_PACKET_SIZE 16
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#define MAX_PACKET_SIZE 0x104000
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int ff_dca_set_channel_layout(AVCodecContext *avctx, int *ch_remap, int dca_mask)
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{
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static const uint8_t dca2wav_norm[28] = {
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2, 0, 1, 9, 10, 3, 8, 4, 5, 9, 10, 6, 7, 12,
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13, 14, 3, 6, 7, 11, 12, 14, 16, 15, 17, 8, 4, 5,
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};
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static const uint8_t dca2wav_wide[28] = {
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2, 0, 1, 4, 5, 3, 8, 4, 5, 9, 10, 6, 7, 12,
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13, 14, 3, 9, 10, 11, 12, 14, 16, 15, 17, 8, 4, 5,
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};
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int dca_ch, wav_ch, nchannels = 0;
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if (avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE) {
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for (dca_ch = 0; dca_ch < DCA_SPEAKER_COUNT; dca_ch++)
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if (dca_mask & (1U << dca_ch))
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ch_remap[nchannels++] = dca_ch;
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avctx->channel_layout = dca_mask;
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} else {
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int wav_mask = 0;
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int wav_map[18];
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const uint8_t *dca2wav;
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if (dca_mask == DCA_SPEAKER_LAYOUT_7POINT0_WIDE ||
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dca_mask == DCA_SPEAKER_LAYOUT_7POINT1_WIDE)
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dca2wav = dca2wav_wide;
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else
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dca2wav = dca2wav_norm;
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for (dca_ch = 0; dca_ch < 28; dca_ch++) {
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if (dca_mask & (1 << dca_ch)) {
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wav_ch = dca2wav[dca_ch];
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if (!(wav_mask & (1 << wav_ch))) {
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wav_map[wav_ch] = dca_ch;
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wav_mask |= 1 << wav_ch;
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}
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}
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}
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for (wav_ch = 0; wav_ch < 18; wav_ch++)
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if (wav_mask & (1 << wav_ch))
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ch_remap[nchannels++] = wav_map[wav_ch];
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avctx->channel_layout = wav_mask;
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}
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avctx->channels = nchannels;
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return nchannels;
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}
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void ff_dca_downmix_to_stereo_fixed(DCADSPContext *dcadsp, int32_t **samples,
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int *coeff_l, int nsamples, int ch_mask)
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{
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int pos, spkr, max_spkr = av_log2(ch_mask);
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int *coeff_r = coeff_l + av_popcount(ch_mask);
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av_assert0(DCA_HAS_STEREO(ch_mask));
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// Scale left and right channels
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pos = (ch_mask & DCA_SPEAKER_MASK_C);
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dcadsp->dmix_scale(samples[DCA_SPEAKER_L], coeff_l[pos ], nsamples);
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dcadsp->dmix_scale(samples[DCA_SPEAKER_R], coeff_r[pos + 1], nsamples);
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// Downmix remaining channels
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for (spkr = 0; spkr <= max_spkr; spkr++) {
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if (!(ch_mask & (1U << spkr)))
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continue;
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if (*coeff_l && spkr != DCA_SPEAKER_L)
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dcadsp->dmix_add(samples[DCA_SPEAKER_L], samples[spkr],
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*coeff_l, nsamples);
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if (*coeff_r && spkr != DCA_SPEAKER_R)
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dcadsp->dmix_add(samples[DCA_SPEAKER_R], samples[spkr],
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*coeff_r, nsamples);
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coeff_l++;
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coeff_r++;
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}
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}
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void ff_dca_downmix_to_stereo_float(AVFloatDSPContext *fdsp, float **samples,
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int *coeff_l, int nsamples, int ch_mask)
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{
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int pos, spkr, max_spkr = av_log2(ch_mask);
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int *coeff_r = coeff_l + av_popcount(ch_mask);
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const float scale = 1.0f / (1 << 15);
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av_assert0(DCA_HAS_STEREO(ch_mask));
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// Scale left and right channels
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pos = (ch_mask & DCA_SPEAKER_MASK_C);
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fdsp->vector_fmul_scalar(samples[DCA_SPEAKER_L], samples[DCA_SPEAKER_L],
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coeff_l[pos ] * scale, nsamples);
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fdsp->vector_fmul_scalar(samples[DCA_SPEAKER_R], samples[DCA_SPEAKER_R],
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coeff_r[pos + 1] * scale, nsamples);
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// Downmix remaining channels
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for (spkr = 0; spkr <= max_spkr; spkr++) {
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if (!(ch_mask & (1U << spkr)))
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continue;
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if (*coeff_l && spkr != DCA_SPEAKER_L)
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fdsp->vector_fmac_scalar(samples[DCA_SPEAKER_L], samples[spkr],
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*coeff_l * scale, nsamples);
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if (*coeff_r && spkr != DCA_SPEAKER_R)
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fdsp->vector_fmac_scalar(samples[DCA_SPEAKER_R], samples[spkr],
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*coeff_r * scale, nsamples);
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coeff_l++;
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coeff_r++;
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}
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}
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static int dcadec_decode_frame(AVCodecContext *avctx, void *data,
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int *got_frame_ptr, AVPacket *avpkt)
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{
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DCAContext *s = avctx->priv_data;
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AVFrame *frame = data;
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uint8_t *input = avpkt->data;
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int input_size = avpkt->size;
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int i, ret, prev_packet = s->packet;
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uint32_t mrk;
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if (input_size < MIN_PACKET_SIZE || input_size > MAX_PACKET_SIZE) {
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av_log(avctx, AV_LOG_ERROR, "Invalid packet size\n");
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return AVERROR_INVALIDDATA;
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}
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// Convert input to BE format
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mrk = AV_RB32(input);
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if (mrk != DCA_SYNCWORD_CORE_BE && mrk != DCA_SYNCWORD_SUBSTREAM) {
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av_fast_padded_malloc(&s->buffer, &s->buffer_size, input_size);
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if (!s->buffer)
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return AVERROR(ENOMEM);
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for (i = 0, ret = AVERROR_INVALIDDATA; i < input_size - MIN_PACKET_SIZE + 1 && ret < 0; i++)
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ret = avpriv_dca_convert_bitstream(input + i, input_size - i, s->buffer, s->buffer_size);
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if (ret < 0) {
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av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");
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return ret;
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}
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input = s->buffer;
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input_size = ret;
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}
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s->packet = 0;
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// Parse backward compatible core sub-stream
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if (AV_RB32(input) == DCA_SYNCWORD_CORE_BE) {
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int frame_size;
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if ((ret = ff_dca_core_parse(&s->core, input, input_size)) < 0)
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return ret;
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s->packet |= DCA_PACKET_CORE;
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// EXXS data must be aligned on 4-byte boundary
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frame_size = FFALIGN(s->core.frame_size, 4);
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if (input_size - 4 > frame_size) {
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input += frame_size;
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input_size -= frame_size;
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}
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}
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if (!s->core_only) {
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DCAExssAsset *asset = NULL;
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// Parse extension sub-stream (EXSS)
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if (AV_RB32(input) == DCA_SYNCWORD_SUBSTREAM) {
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if ((ret = ff_dca_exss_parse(&s->exss, input, input_size)) < 0) {
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if (avctx->err_recognition & AV_EF_EXPLODE)
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return ret;
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} else {
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s->packet |= DCA_PACKET_EXSS;
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asset = &s->exss.assets[0];
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}
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}
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// Parse XLL component in EXSS
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if (asset && (asset->extension_mask & DCA_EXSS_XLL)) {
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if ((ret = ff_dca_xll_parse(&s->xll, input, asset)) < 0) {
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// Conceal XLL synchronization error
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if (ret == AVERROR(EAGAIN)
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&& (prev_packet & DCA_PACKET_XLL)
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&& (s->packet & DCA_PACKET_CORE))
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s->packet |= DCA_PACKET_XLL | DCA_PACKET_RECOVERY;
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else if (ret == AVERROR(ENOMEM) || (avctx->err_recognition & AV_EF_EXPLODE))
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return ret;
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} else {
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s->packet |= DCA_PACKET_XLL;
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}
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}
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// Parse LBR component in EXSS
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if (asset && (asset->extension_mask & DCA_EXSS_LBR)) {
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if ((ret = ff_dca_lbr_parse(&s->lbr, input, asset)) < 0) {
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if (ret == AVERROR(ENOMEM) || (avctx->err_recognition & AV_EF_EXPLODE))
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return ret;
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} else {
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s->packet |= DCA_PACKET_LBR;
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}
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}
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// Parse core extensions in EXSS or backward compatible core sub-stream
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if ((s->packet & DCA_PACKET_CORE)
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&& (ret = ff_dca_core_parse_exss(&s->core, input, asset)) < 0)
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return ret;
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}
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// Filter the frame
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if (s->packet & DCA_PACKET_LBR) {
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if ((ret = ff_dca_lbr_filter_frame(&s->lbr, frame)) < 0)
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return ret;
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} else if (s->packet & DCA_PACKET_XLL) {
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if (s->packet & DCA_PACKET_CORE) {
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int x96_synth = -1;
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// Enable X96 synthesis if needed
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if (s->xll.chset[0].freq == 96000 && s->core.sample_rate == 48000)
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x96_synth = 1;
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if ((ret = ff_dca_core_filter_fixed(&s->core, x96_synth)) < 0)
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return ret;
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// Force lossy downmixed output on the first core frame filtered.
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// This prevents audible clicks when seeking and is consistent with
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// what reference decoder does when there are multiple channel sets.
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if (!(prev_packet & DCA_PACKET_RESIDUAL) && s->xll.nreschsets > 0
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&& s->xll.nchsets > 1) {
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av_log(avctx, AV_LOG_VERBOSE, "Forcing XLL recovery mode\n");
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s->packet |= DCA_PACKET_RECOVERY;
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}
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// Set 'residual ok' flag for the next frame
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s->packet |= DCA_PACKET_RESIDUAL;
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}
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if ((ret = ff_dca_xll_filter_frame(&s->xll, frame)) < 0) {
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// Fall back to core unless hard error
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if (!(s->packet & DCA_PACKET_CORE))
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return ret;
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if (ret != AVERROR_INVALIDDATA || (avctx->err_recognition & AV_EF_EXPLODE))
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return ret;
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if ((ret = ff_dca_core_filter_frame(&s->core, frame)) < 0)
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return ret;
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}
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} else if (s->packet & DCA_PACKET_CORE) {
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if ((ret = ff_dca_core_filter_frame(&s->core, frame)) < 0)
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return ret;
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if (s->core.filter_mode & DCA_FILTER_MODE_FIXED)
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s->packet |= DCA_PACKET_RESIDUAL;
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} else {
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av_log(avctx, AV_LOG_ERROR, "No valid DCA sub-stream found\n");
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if (s->core_only)
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av_log(avctx, AV_LOG_WARNING, "Consider disabling 'core_only' option\n");
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return AVERROR_INVALIDDATA;
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}
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*got_frame_ptr = 1;
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return avpkt->size;
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}
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static av_cold void dcadec_flush(AVCodecContext *avctx)
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{
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DCAContext *s = avctx->priv_data;
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ff_dca_core_flush(&s->core);
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ff_dca_xll_flush(&s->xll);
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ff_dca_lbr_flush(&s->lbr);
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s->packet &= DCA_PACKET_MASK;
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}
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static av_cold int dcadec_close(AVCodecContext *avctx)
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{
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DCAContext *s = avctx->priv_data;
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ff_dca_core_close(&s->core);
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ff_dca_xll_close(&s->xll);
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ff_dca_lbr_close(&s->lbr);
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av_freep(&s->buffer);
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s->buffer_size = 0;
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return 0;
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}
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static av_cold int dcadec_init(AVCodecContext *avctx)
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{
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DCAContext *s = avctx->priv_data;
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s->avctx = avctx;
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s->core.avctx = avctx;
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s->exss.avctx = avctx;
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s->xll.avctx = avctx;
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s->lbr.avctx = avctx;
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ff_dca_init_vlcs();
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if (ff_dca_core_init(&s->core) < 0)
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return AVERROR(ENOMEM);
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if (ff_dca_lbr_init(&s->lbr) < 0)
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return AVERROR(ENOMEM);
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ff_dcadsp_init(&s->dcadsp);
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s->core.dcadsp = &s->dcadsp;
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s->xll.dcadsp = &s->dcadsp;
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s->lbr.dcadsp = &s->dcadsp;
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s->crctab = av_crc_get_table(AV_CRC_16_CCITT);
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switch (avctx->request_channel_layout & ~AV_CH_LAYOUT_NATIVE) {
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case 0:
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s->request_channel_layout = 0;
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break;
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case AV_CH_LAYOUT_STEREO:
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case AV_CH_LAYOUT_STEREO_DOWNMIX:
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s->request_channel_layout = DCA_SPEAKER_LAYOUT_STEREO;
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break;
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case AV_CH_LAYOUT_5POINT0:
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s->request_channel_layout = DCA_SPEAKER_LAYOUT_5POINT0;
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break;
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case AV_CH_LAYOUT_5POINT1:
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s->request_channel_layout = DCA_SPEAKER_LAYOUT_5POINT1;
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break;
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default:
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av_log(avctx, AV_LOG_WARNING, "Invalid request_channel_layout\n");
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break;
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}
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return 0;
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}
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#define OFFSET(x) offsetof(DCAContext, x)
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#define PARAM AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM
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static const AVOption dcadec_options[] = {
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{ "core_only", "Decode core only without extensions", OFFSET(core_only), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, PARAM },
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{ NULL }
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};
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static const AVClass dcadec_class = {
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.class_name = "DCA decoder",
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.item_name = av_default_item_name,
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.option = dcadec_options,
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.version = LIBAVUTIL_VERSION_INT,
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.category = AV_CLASS_CATEGORY_DECODER,
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};
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AVCodec ff_dca_decoder = {
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.name = "dca",
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.long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
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.type = AVMEDIA_TYPE_AUDIO,
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.id = AV_CODEC_ID_DTS,
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.priv_data_size = sizeof(DCAContext),
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.init = dcadec_init,
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.decode = dcadec_decode_frame,
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.close = dcadec_close,
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.flush = dcadec_flush,
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.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P,
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AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE },
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.priv_class = &dcadec_class,
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.profiles = NULL_IF_CONFIG_SMALL(ff_dca_profiles),
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.caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
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};
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