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Diffstat (limited to 'ffmpeg1/libavcodec/dcaenc.c')
-rw-r--r--ffmpeg1/libavcodec/dcaenc.c602
1 files changed, 0 insertions, 602 deletions
diff --git a/ffmpeg1/libavcodec/dcaenc.c b/ffmpeg1/libavcodec/dcaenc.c
deleted file mode 100644
index 4799ef4..0000000
--- a/ffmpeg1/libavcodec/dcaenc.c
+++ /dev/null
@@ -1,602 +0,0 @@
-/*
- * DCA encoder
- * Copyright (C) 2008 Alexander E. Patrakov
- * 2010 Benjamin Larsson
- * 2011 Xiang Wang
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include "libavutil/channel_layout.h"
-#include "libavutil/common.h"
-#include "libavutil/avassert.h"
-#include "avcodec.h"
-#include "internal.h"
-#include "put_bits.h"
-#include "dcaenc.h"
-#include "dcadata.h"
-#include "dca.h"
-
-#undef NDEBUG
-
-#define MAX_CHANNELS 6
-#define DCA_SUBBANDS_32 32
-#define DCA_MAX_FRAME_SIZE 16383
-#define DCA_HEADER_SIZE 13
-
-#define DCA_SUBBANDS 32 ///< Subband activity count
-#define QUANTIZER_BITS 16
-#define SUBFRAMES 1
-#define SUBSUBFRAMES 4
-#define PCM_SAMPLES (SUBFRAMES*SUBSUBFRAMES*8)
-#define LFE_BITS 8
-#define LFE_INTERPOLATION 64
-#define LFE_PRESENT 2
-#define LFE_MISSING 0
-
-static const int8_t dca_lfe_index[] = {
- 1,2,2,2,2,3,2,3,2,3,2,3,1,3,2,3
-};
-
-static const int8_t dca_channel_reorder_lfe[][9] = {
- { 0, -1, -1, -1, -1, -1, -1, -1, -1 },
- { 0, 1, -1, -1, -1, -1, -1, -1, -1 },
- { 0, 1, -1, -1, -1, -1, -1, -1, -1 },
- { 0, 1, -1, -1, -1, -1, -1, -1, -1 },
- { 0, 1, -1, -1, -1, -1, -1, -1, -1 },
- { 1, 2, 0, -1, -1, -1, -1, -1, -1 },
- { 0, 1, -1, 2, -1, -1, -1, -1, -1 },
- { 1, 2, 0, -1, 3, -1, -1, -1, -1 },
- { 0, 1, -1, 2, 3, -1, -1, -1, -1 },
- { 1, 2, 0, -1, 3, 4, -1, -1, -1 },
- { 2, 3, -1, 0, 1, 4, 5, -1, -1 },
- { 1, 2, 0, -1, 3, 4, 5, -1, -1 },
- { 0, -1, 4, 5, 2, 3, 1, -1, -1 },
- { 3, 4, 1, -1, 0, 2, 5, 6, -1 },
- { 2, 3, -1, 5, 7, 0, 1, 4, 6 },
- { 3, 4, 1, -1, 0, 2, 5, 7, 6 },
-};
-
-static const int8_t dca_channel_reorder_nolfe[][9] = {
- { 0, -1, -1, -1, -1, -1, -1, -1, -1 },
- { 0, 1, -1, -1, -1, -1, -1, -1, -1 },
- { 0, 1, -1, -1, -1, -1, -1, -1, -1 },
- { 0, 1, -1, -1, -1, -1, -1, -1, -1 },
- { 0, 1, -1, -1, -1, -1, -1, -1, -1 },
- { 1, 2, 0, -1, -1, -1, -1, -1, -1 },
- { 0, 1, 2, -1, -1, -1, -1, -1, -1 },
- { 1, 2, 0, 3, -1, -1, -1, -1, -1 },
- { 0, 1, 2, 3, -1, -1, -1, -1, -1 },
- { 1, 2, 0, 3, 4, -1, -1, -1, -1 },
- { 2, 3, 0, 1, 4, 5, -1, -1, -1 },
- { 1, 2, 0, 3, 4, 5, -1, -1, -1 },
- { 0, 4, 5, 2, 3, 1, -1, -1, -1 },
- { 3, 4, 1, 0, 2, 5, 6, -1, -1 },
- { 2, 3, 5, 7, 0, 1, 4, 6, -1 },
- { 3, 4, 1, 0, 2, 5, 7, 6, -1 },
-};
-
-typedef struct {
- PutBitContext pb;
- int32_t history[MAX_CHANNELS][512]; /* This is a circular buffer */
- int start[MAX_CHANNELS];
- int frame_size;
- int prim_channels;
- int lfe_channel;
- int sample_rate_code;
- int scale_factor[MAX_CHANNELS][DCA_SUBBANDS_32];
- int lfe_scale_factor;
- int lfe_data[SUBFRAMES*SUBSUBFRAMES*4];
-
- int a_mode; ///< audio channels arrangement
- int num_channel;
- int lfe_state;
- int lfe_offset;
- const int8_t *channel_order_tab; ///< channel reordering table, lfe and non lfe
-
- int32_t pcm[FFMAX(LFE_INTERPOLATION, DCA_SUBBANDS_32)];
- int32_t subband[PCM_SAMPLES][MAX_CHANNELS][DCA_SUBBANDS_32]; /* [sample][channel][subband] */
-} DCAContext;
-
-static int32_t cos_table[128];
-
-static inline int32_t mul32(int32_t a, int32_t b)
-{
- int64_t r = (int64_t) a * b;
- /* round the result before truncating - improves accuracy */
- return (r + 0x80000000) >> 32;
-}
-
-/* Integer version of the cosine modulated Pseudo QMF */
-
-static void qmf_init(void)
-{
- int i;
- int32_t c[17], s[17];
- s[0] = 0; /* sin(index * PI / 64) * 0x7fffffff */
- c[0] = 0x7fffffff; /* cos(index * PI / 64) * 0x7fffffff */
-
- for (i = 1; i <= 16; i++) {
- s[i] = 2 * (mul32(c[i - 1], 105372028) + mul32(s[i - 1], 2144896908));
- c[i] = 2 * (mul32(c[i - 1], 2144896908) - mul32(s[i - 1], 105372028));
- }
-
- for (i = 0; i < 16; i++) {
- cos_table[i ] = c[i] >> 3; /* avoid output overflow */
- cos_table[i + 16] = s[16 - i] >> 3;
- cos_table[i + 32] = -s[i] >> 3;
- cos_table[i + 48] = -c[16 - i] >> 3;
- cos_table[i + 64] = -c[i] >> 3;
- cos_table[i + 80] = -s[16 - i] >> 3;
- cos_table[i + 96] = s[i] >> 3;
- cos_table[i + 112] = c[16 - i] >> 3;
- }
-}
-
-static int32_t band_delta_factor(int band, int sample_num)
-{
- int index = band * (2 * sample_num + 1);
- if (band == 0)
- return 0x07ffffff;
- else
- return cos_table[index & 127];
-}
-
-static void add_new_samples(DCAContext *c, const int32_t *in,
- int count, int channel)
-{
- int i;
-
- /* Place new samples into the history buffer */
- for (i = 0; i < count; i++) {
- c->history[channel][c->start[channel] + i] = in[i];
- av_assert0(c->start[channel] + i < 512);
- }
- c->start[channel] += count;
- if (c->start[channel] == 512)
- c->start[channel] = 0;
- av_assert0(c->start[channel] < 512);
-}
-
-static void qmf_decompose(DCAContext *c, int32_t in[32], int32_t out[32],
- int channel)
-{
- int band, i, j, k;
- int32_t resp;
- int32_t accum[DCA_SUBBANDS_32] = {0};
-
- add_new_samples(c, in, DCA_SUBBANDS_32, channel);
-
- /* Calculate the dot product of the signal with the (possibly inverted)
- reference decoder's response to this vector:
- (0.0, 0.0, ..., 0.0, -1.0, 1.0, 0.0, ..., 0.0)
- so that -1.0 cancels 1.0 from the previous step */
-
- for (k = 48, j = 0, i = c->start[channel]; i < 512; k++, j++, i++)
- accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] += mul32(c->history[channel][i], UnQMF[j]);
- for (i = 0; i < c->start[channel]; k++, j++, i++)
- accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] += mul32(c->history[channel][i], UnQMF[j]);
-
- resp = 0;
- /* TODO: implement FFT instead of this naive calculation */
- for (band = 0; band < DCA_SUBBANDS_32; band++) {
- for (j = 0; j < 32; j++)
- resp += mul32(accum[j], band_delta_factor(band, j));
-
- out[band] = (band & 2) ? (-resp) : resp;
- }
-}
-
-static int32_t lfe_fir_64i[512];
-static int lfe_downsample(DCAContext *c, int32_t in[LFE_INTERPOLATION])
-{
- int i, j;
- int channel = c->prim_channels;
- int32_t accum = 0;
-
- add_new_samples(c, in, LFE_INTERPOLATION, channel);
- for (i = c->start[channel], j = 0; i < 512; i++, j++)
- accum += mul32(c->history[channel][i], lfe_fir_64i[j]);
- for (i = 0; i < c->start[channel]; i++, j++)
- accum += mul32(c->history[channel][i], lfe_fir_64i[j]);
- return accum;
-}
-
-static void init_lfe_fir(void)
-{
- static int initialized = 0;
- int i;
- if (initialized)
- return;
-
- for (i = 0; i < 512; i++)
- lfe_fir_64i[i] = lfe_fir_64[i] * (1 << 25); //float -> int32_t
- initialized = 1;
-}
-
-static void put_frame_header(DCAContext *c)
-{
- /* SYNC */
- put_bits(&c->pb, 16, 0x7ffe);
- put_bits(&c->pb, 16, 0x8001);
-
- /* Frame type: normal */
- put_bits(&c->pb, 1, 1);
-
- /* Deficit sample count: none */
- put_bits(&c->pb, 5, 31);
-
- /* CRC is not present */
- put_bits(&c->pb, 1, 0);
-
- /* Number of PCM sample blocks */
- put_bits(&c->pb, 7, PCM_SAMPLES-1);
-
- /* Primary frame byte size */
- put_bits(&c->pb, 14, c->frame_size-1);
-
- /* Audio channel arrangement: L + R (stereo) */
- put_bits(&c->pb, 6, c->num_channel);
-
- /* Core audio sampling frequency */
- put_bits(&c->pb, 4, c->sample_rate_code);
-
- /* Transmission bit rate: 1411.2 kbps */
- put_bits(&c->pb, 5, 0x16); /* FIXME: magic number */
-
- /* Embedded down mix: disabled */
- put_bits(&c->pb, 1, 0);
-
- /* Embedded dynamic range flag: not present */
- put_bits(&c->pb, 1, 0);
-
- /* Embedded time stamp flag: not present */
- put_bits(&c->pb, 1, 0);
-
- /* Auxiliary data flag: not present */
- put_bits(&c->pb, 1, 0);
-
- /* HDCD source: no */
- put_bits(&c->pb, 1, 0);
-
- /* Extension audio ID: N/A */
- put_bits(&c->pb, 3, 0);
-
- /* Extended audio data: not present */
- put_bits(&c->pb, 1, 0);
-
- /* Audio sync word insertion flag: after each sub-frame */
- put_bits(&c->pb, 1, 0);
-
- /* Low frequency effects flag: not present or interpolation factor=64 */
- put_bits(&c->pb, 2, c->lfe_state);
-
- /* Predictor history switch flag: on */
- put_bits(&c->pb, 1, 1);
-
- /* No CRC */
- /* Multirate interpolator switch: non-perfect reconstruction */
- put_bits(&c->pb, 1, 0);
-
- /* Encoder software revision: 7 */
- put_bits(&c->pb, 4, 7);
-
- /* Copy history: 0 */
- put_bits(&c->pb, 2, 0);
-
- /* Source PCM resolution: 16 bits, not DTS ES */
- put_bits(&c->pb, 3, 0);
-
- /* Front sum/difference coding: no */
- put_bits(&c->pb, 1, 0);
-
- /* Surrounds sum/difference coding: no */
- put_bits(&c->pb, 1, 0);
-
- /* Dialog normalization: 0 dB */
- put_bits(&c->pb, 4, 0);
-}
-
-static void put_primary_audio_header(DCAContext *c)
-{
- static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
- static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
-
- int ch, i;
- /* Number of subframes */
- put_bits(&c->pb, 4, SUBFRAMES - 1);
-
- /* Number of primary audio channels */
- put_bits(&c->pb, 3, c->prim_channels - 1);
-
- /* Subband activity count */
- for (ch = 0; ch < c->prim_channels; ch++)
- put_bits(&c->pb, 5, DCA_SUBBANDS - 2);
-
- /* High frequency VQ start subband */
- for (ch = 0; ch < c->prim_channels; ch++)
- put_bits(&c->pb, 5, DCA_SUBBANDS - 1);
-
- /* Joint intensity coding index: 0, 0 */
- for (ch = 0; ch < c->prim_channels; ch++)
- put_bits(&c->pb, 3, 0);
-
- /* Transient mode codebook: A4, A4 (arbitrary) */
- for (ch = 0; ch < c->prim_channels; ch++)
- put_bits(&c->pb, 2, 0);
-
- /* Scale factor code book: 7 bit linear, 7-bit sqrt table (for each channel) */
- for (ch = 0; ch < c->prim_channels; ch++)
- put_bits(&c->pb, 3, 6);
-
- /* Bit allocation quantizer select: linear 5-bit */
- for (ch = 0; ch < c->prim_channels; ch++)
- put_bits(&c->pb, 3, 6);
-
- /* Quantization index codebook select: dummy data
- to avoid transmission of scale factor adjustment */
-
- for (i = 1; i < 11; i++)
- for (ch = 0; ch < c->prim_channels; ch++)
- put_bits(&c->pb, bitlen[i], thr[i]);
-
- /* Scale factor adjustment index: not transmitted */
-}
-
-/**
- * 8-23 bits quantization
- * @param sample
- * @param bits
- */
-static inline uint32_t quantize(int32_t sample, int bits)
-{
- av_assert0(sample < 1 << (bits - 1));
- av_assert0(sample >= -(1 << (bits - 1)));
- return sample & ((1 << bits) - 1);
-}
-
-static inline int find_scale_factor7(int64_t max_value, int bits)
-{
- int i = 0, j = 128, q;
- max_value = ((max_value << 15) / lossy_quant[bits + 3]) >> (bits - 1);
- while (i < j) {
- q = (i + j) >> 1;
- if (max_value < scale_factor_quant7[q])
- j = q;
- else
- i = q + 1;
- }
- av_assert1(i < 128);
- return i;
-}
-
-static inline void put_sample7(DCAContext *c, int64_t sample, int bits,
- int scale_factor)
-{
- sample = (sample << 15) / ((int64_t) lossy_quant[bits + 3] * scale_factor_quant7[scale_factor]);
- put_bits(&c->pb, bits, quantize((int) sample, bits));
-}
-
-static void put_subframe(DCAContext *c,
- int32_t subband_data[8 * SUBSUBFRAMES][MAX_CHANNELS][32],
- int subframe)
-{
- int i, sub, ss, ch, max_value;
- int32_t *lfe_data = c->lfe_data + 4 * SUBSUBFRAMES * subframe;
-
- /* Subsubframes count */
- put_bits(&c->pb, 2, SUBSUBFRAMES -1);
-
- /* Partial subsubframe sample count: dummy */
- put_bits(&c->pb, 3, 0);
-
- /* Prediction mode: no ADPCM, in each channel and subband */
- for (ch = 0; ch < c->prim_channels; ch++)
- for (sub = 0; sub < DCA_SUBBANDS; sub++)
- put_bits(&c->pb, 1, 0);
-
- /* Prediction VQ addres: not transmitted */
- /* Bit allocation index */
- for (ch = 0; ch < c->prim_channels; ch++)
- for (sub = 0; sub < DCA_SUBBANDS; sub++)
- put_bits(&c->pb, 5, QUANTIZER_BITS+3);
-
- if (SUBSUBFRAMES > 1) {
- /* Transition mode: none for each channel and subband */
- for (ch = 0; ch < c->prim_channels; ch++)
- for (sub = 0; sub < DCA_SUBBANDS; sub++)
- put_bits(&c->pb, 1, 0); /* codebook A4 */
- }
-
- /* Determine scale_factor */
- for (ch = 0; ch < c->prim_channels; ch++)
- for (sub = 0; sub < DCA_SUBBANDS; sub++) {
- max_value = 0;
- for (i = 0; i < 8 * SUBSUBFRAMES; i++)
- max_value = FFMAX(max_value, FFABS(subband_data[i][ch][sub]));
- c->scale_factor[ch][sub] = find_scale_factor7(max_value, QUANTIZER_BITS);
- }
-
- if (c->lfe_channel) {
- max_value = 0;
- for (i = 0; i < 4 * SUBSUBFRAMES; i++)
- max_value = FFMAX(max_value, FFABS(lfe_data[i]));
- c->lfe_scale_factor = find_scale_factor7(max_value, LFE_BITS);
- }
-
- /* Scale factors: the same for each channel and subband,
- encoded according to Table D.1.2 */
- for (ch = 0; ch < c->prim_channels; ch++)
- for (sub = 0; sub < DCA_SUBBANDS; sub++)
- put_bits(&c->pb, 7, c->scale_factor[ch][sub]);
-
- /* Joint subband scale factor codebook select: not transmitted */
- /* Scale factors for joint subband coding: not transmitted */
- /* Stereo down-mix coefficients: not transmitted */
- /* Dynamic range coefficient: not transmitted */
- /* Stde information CRC check word: not transmitted */
- /* VQ encoded high frequency subbands: not transmitted */
-
- /* LFE data */
- if (c->lfe_channel) {
- for (i = 0; i < 4 * SUBSUBFRAMES; i++)
- put_sample7(c, lfe_data[i], LFE_BITS, c->lfe_scale_factor);
- put_bits(&c->pb, 8, c->lfe_scale_factor);
- }
-
- /* Audio data (subsubframes) */
-
- for (ss = 0; ss < SUBSUBFRAMES ; ss++)
- for (ch = 0; ch < c->prim_channels; ch++)
- for (sub = 0; sub < DCA_SUBBANDS; sub++)
- for (i = 0; i < 8; i++)
- put_sample7(c, subband_data[ss * 8 + i][ch][sub], QUANTIZER_BITS, c->scale_factor[ch][sub]);
-
- /* DSYNC */
- put_bits(&c->pb, 16, 0xffff);
-}
-
-static void put_frame(DCAContext *c,
- int32_t subband_data[PCM_SAMPLES][MAX_CHANNELS][32],
- uint8_t *frame)
-{
- int i;
- init_put_bits(&c->pb, frame + DCA_HEADER_SIZE, DCA_MAX_FRAME_SIZE-DCA_HEADER_SIZE);
-
- put_primary_audio_header(c);
- for (i = 0; i < SUBFRAMES; i++)
- put_subframe(c, &subband_data[SUBSUBFRAMES * 8 * i], i);
-
- flush_put_bits(&c->pb);
- c->frame_size = (put_bits_count(&c->pb) >> 3) + DCA_HEADER_SIZE;
-
- init_put_bits(&c->pb, frame, DCA_HEADER_SIZE);
- put_frame_header(c);
- flush_put_bits(&c->pb);
-}
-
-static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
- const AVFrame *frame, int *got_packet_ptr)
-{
- int i, k, channel;
- DCAContext *c = avctx->priv_data;
- const int16_t *samples;
- int ret, real_channel = 0;
-
- if ((ret = ff_alloc_packet2(avctx, avpkt, DCA_MAX_FRAME_SIZE + DCA_HEADER_SIZE)) < 0)
- return ret;
-
- samples = (const int16_t *)frame->data[0];
- for (i = 0; i < PCM_SAMPLES; i ++) { /* i is the decimated sample number */
- for (channel = 0; channel < c->prim_channels + 1; channel++) {
- real_channel = c->channel_order_tab[channel];
- if (real_channel >= 0) {
- /* Get 32 PCM samples */
- for (k = 0; k < 32; k++) { /* k is the sample number in a 32-sample block */
- c->pcm[k] = samples[avctx->channels * (32 * i + k) + channel] << 16;
- }
- /* Put subband samples into the proper place */
- qmf_decompose(c, c->pcm, &c->subband[i][real_channel][0], real_channel);
- }
- }
- }
-
- if (c->lfe_channel) {
- for (i = 0; i < PCM_SAMPLES / 2; i++) {
- for (k = 0; k < LFE_INTERPOLATION; k++) /* k is the sample number in a 32-sample block */
- c->pcm[k] = samples[avctx->channels * (LFE_INTERPOLATION*i+k) + c->lfe_offset] << 16;
- c->lfe_data[i] = lfe_downsample(c, c->pcm);
- }
- }
-
- put_frame(c, c->subband, avpkt->data);
-
- avpkt->size = c->frame_size;
- *got_packet_ptr = 1;
- return 0;
-}
-
-static int encode_init(AVCodecContext *avctx)
-{
- DCAContext *c = avctx->priv_data;
- int i;
- uint64_t layout = avctx->channel_layout;
-
- c->prim_channels = avctx->channels;
- c->lfe_channel = (avctx->channels == 3 || avctx->channels == 6);
-
- if (!layout) {
- av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The "
- "encoder will guess the layout, but it "
- "might be incorrect.\n");
- layout = av_get_default_channel_layout(avctx->channels);
- }
- switch (layout) {
- case AV_CH_LAYOUT_STEREO: c->a_mode = 2; c->num_channel = 2; break;
- case AV_CH_LAYOUT_5POINT0: c->a_mode = 9; c->num_channel = 9; break;
- case AV_CH_LAYOUT_5POINT1: c->a_mode = 9; c->num_channel = 9; break;
- case AV_CH_LAYOUT_5POINT0_BACK: c->a_mode = 9; c->num_channel = 9; break;
- case AV_CH_LAYOUT_5POINT1_BACK: c->a_mode = 9; c->num_channel = 9; break;
- default:
- av_log(avctx, AV_LOG_ERROR,
- "Only stereo, 5.0, 5.1 channel layouts supported at the moment!\n");
- return AVERROR_PATCHWELCOME;
- }
-
- if (c->lfe_channel) {
- init_lfe_fir();
- c->prim_channels--;
- c->channel_order_tab = dca_channel_reorder_lfe[c->a_mode];
- c->lfe_state = LFE_PRESENT;
- c->lfe_offset = dca_lfe_index[c->a_mode];
- } else {
- c->channel_order_tab = dca_channel_reorder_nolfe[c->a_mode];
- c->lfe_state = LFE_MISSING;
- }
-
- for (i = 0; i < 16; i++) {
- if (avpriv_dca_sample_rates[i] && (avpriv_dca_sample_rates[i] == avctx->sample_rate))
- break;
- }
- if (i == 16) {
- av_log(avctx, AV_LOG_ERROR, "Sample rate %iHz not supported, only ", avctx->sample_rate);
- for (i = 0; i < 16; i++)
- av_log(avctx, AV_LOG_ERROR, "%d, ", avpriv_dca_sample_rates[i]);
- av_log(avctx, AV_LOG_ERROR, "supported.\n");
- return -1;
- }
- c->sample_rate_code = i;
-
- avctx->frame_size = 32 * PCM_SAMPLES;
-
- if (!cos_table[127])
- qmf_init();
- return 0;
-}
-
-AVCodec ff_dca_encoder = {
- .name = "dca",
- .type = AVMEDIA_TYPE_AUDIO,
- .id = AV_CODEC_ID_DTS,
- .priv_data_size = sizeof(DCAContext),
- .init = encode_init,
- .encode2 = encode_frame,
- .capabilities = CODEC_CAP_EXPERIMENTAL,
- .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
- AV_SAMPLE_FMT_NONE },
- .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
-};
generated by cgit v1.2.3 (git 2.39.1) at 2025-12-06 17:25:51 +0000