mirror of
https://git.ffmpeg.org/ffmpeg.git
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418 lines
13 KiB
C
418 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 "dcamath.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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static uint16_t crc16(const uint8_t *data, int size)
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{
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static const uint16_t crctab[16] = {
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0x0000, 0x1021, 0x2042, 0x3063, 0x4084, 0x50a5, 0x60c6, 0x70e7,
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0x8108, 0x9129, 0xa14a, 0xb16b, 0xc18c, 0xd1ad, 0xe1ce, 0xf1ef,
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};
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uint16_t res = 0xffff;
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int i;
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for (i = 0; i < size; i++) {
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res = (res << 4) ^ crctab[(data[i] >> 4) ^ (res >> 12)];
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res = (res << 4) ^ crctab[(data[i] & 15) ^ (res >> 12)];
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}
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return res;
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}
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int ff_dca_check_crc(GetBitContext *s, int p1, int p2)
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{
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if (((p1 | p2) & 7) || p1 < 0 || p2 > s->size_in_bits || p2 - p1 < 16)
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return -1;
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if (crc16(s->buffer + p1 / 8, (p2 - p1) / 8))
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return -1;
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return 0;
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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 convert_bitstream(const uint8_t *src, int src_size, uint8_t *dst, int max_size)
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{
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switch (AV_RB32(src)) {
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case DCA_SYNCWORD_CORE_BE:
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case DCA_SYNCWORD_SUBSTREAM:
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memcpy(dst, src, src_size);
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return src_size;
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case DCA_SYNCWORD_CORE_LE:
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case DCA_SYNCWORD_CORE_14B_BE:
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case DCA_SYNCWORD_CORE_14B_LE:
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return avpriv_dca_convert_bitstream(src, src_size, dst, max_size);
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default:
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return AVERROR_INVALIDDATA;
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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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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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av_fast_malloc(&s->buffer, &s->buffer_size,
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FFALIGN(input_size, 4096) + DCA_BUFFER_PADDING_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 = convert_bitstream(input + i, input_size - i, s->buffer, s->buffer_size);
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if (ret < 0)
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return ret;
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input = s->buffer;
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input_size = ret;
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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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s->core_residual_valid = 0;
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return ret;
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}
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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 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_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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s->core_residual_valid = 0;
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return ret;
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}
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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 (!s->core_residual_valid) {
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if (s->xll.nreschsets > 0 && s->xll.nchsets > 1)
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s->packet |= DCA_PACKET_RECOVERY;
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s->core_residual_valid = 1;
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}
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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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s->core_residual_valid = 0;
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return ret;
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}
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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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s->core_residual_valid = 0;
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return ret;
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}
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s->core_residual_valid = !!(s->core.filter_mode & DCA_FILTER_MODE_FIXED);
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} else {
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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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s->core_residual_valid = 0;
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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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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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if (ff_dca_core_init(&s->core) < 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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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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avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
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avctx->bits_per_raw_sample = 24;
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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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