* Copyright (c) 2007-2008 Bartlomiej Wolowiec <bartek.wolowiec@gmail.com>
* Copyright (c) 2007 Justin Ruggles <justin.ruggles@gmail.com>
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav 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,
+ * Libav 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
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <string.h>
#include "libavutil/crc.h"
+#include "libavutil/opt.h"
#include "internal.h"
#include "aac_ac3_parser.h"
#include "ac3_parser.h"
#include "ac3dec.h"
#include "ac3dec_data.h"
-
-/** Large enough for maximum possible frame size when the specification limit is ignored */
-#define AC3_FRAME_BUFFER_SIZE 32768
+#include "kbdwin.h"
/**
* table for ungrouping 3 values in 7 bits.
static float dynamic_range_tab[256];
/** Adjustments in dB gain */
-#define LEVEL_PLUS_3DB 1.4142135623730950
-#define LEVEL_PLUS_1POINT5DB 1.1892071150027209
-#define LEVEL_MINUS_1POINT5DB 0.8408964152537145
-#define LEVEL_MINUS_3DB 0.7071067811865476
-#define LEVEL_MINUS_4POINT5DB 0.5946035575013605
-#define LEVEL_MINUS_6DB 0.5000000000000000
-#define LEVEL_MINUS_9DB 0.3535533905932738
-#define LEVEL_ZERO 0.0000000000000000
-#define LEVEL_ONE 1.0000000000000000
-
static const float gain_levels[9] = {
LEVEL_PLUS_3DB,
LEVEL_PLUS_1POINT5DB,
AC3DecodeContext *s = avctx->priv_data;
s->avctx = avctx;
- ac3_common_init();
+#if FF_API_DRC_SCALE
+ if (avctx->drc_scale)
+ s->drc_scale = avctx->drc_scale;
+#endif
+
+ ff_ac3_common_init();
ac3_tables_init();
ff_mdct_init(&s->imdct_256, 8, 1, 1.0);
ff_mdct_init(&s->imdct_512, 9, 1, 1.0);
ff_kbd_window_init(s->window, 5.0, 256);
dsputil_init(&s->dsp, avctx);
+ ff_ac3dsp_init(&s->ac3dsp, avctx->flags & CODEC_FLAG_BITEXACT);
+ ff_fmt_convert_init(&s->fmt_conv, avctx);
av_lfg_init(&s->dith_state, 0);
- /* set bias values for float to int16 conversion */
- if(s->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) {
- s->add_bias = 385.0f;
+ /* set scale value for float to int16 conversion */
+ if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
s->mul_bias = 1.0f;
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
} else {
- s->add_bias = 0.0f;
s->mul_bias = 32767.0f;
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16;
}
/* allow downmixing to stereo or mono */
}
s->downmixed = 1;
- /* allocate context input buffer */
- if (avctx->error_recognition >= FF_ER_CAREFUL) {
- s->input_buffer = av_mallocz(AC3_FRAME_BUFFER_SIZE + FF_INPUT_BUFFER_PADDING_SIZE);
- if (!s->input_buffer)
- return AVERROR_NOMEM;
- }
-
- avctx->sample_fmt = SAMPLE_FMT_S16;
return 0;
}
/* get decoding parameters from header info */
s->bit_alloc_params.sr_code = hdr.sr_code;
+ s->bitstream_mode = hdr.bitstream_mode;
s->channel_mode = hdr.channel_mode;
s->channel_layout = hdr.channel_layout;
s->lfe_on = hdr.lfe_on;
bin = s->start_freq[CPL_CH];
for (band = 0; band < s->num_cpl_bands; band++) {
+ int band_start = bin;
int band_end = bin + s->cpl_band_sizes[band];
- for (; bin < band_end; bin++) {
- for (ch = 1; ch <= s->fbw_channels; ch++) {
- if (s->channel_in_cpl[ch]) {
- s->fixed_coeffs[ch][bin] = ((int64_t)s->fixed_coeffs[CPL_CH][bin] *
- (int64_t)s->cpl_coords[ch][band]) >> 23;
- if (ch == 2 && s->phase_flags[band])
- s->fixed_coeffs[ch][bin] = -s->fixed_coeffs[ch][bin];
+ for (ch = 1; ch <= s->fbw_channels; ch++) {
+ if (s->channel_in_cpl[ch]) {
+ int cpl_coord = s->cpl_coords[ch][band] << 5;
+ for (bin = band_start; bin < band_end; bin++) {
+ s->fixed_coeffs[ch][bin] = MULH(s->fixed_coeffs[CPL_CH][bin] << 4, cpl_coord);
+ }
+ if (ch == 2 && s->phase_flags[band]) {
+ for (bin = band_start; bin < band_end; bin++)
+ s->fixed_coeffs[2][bin] = -s->fixed_coeffs[2][bin];
}
}
}
+ bin = band_end;
}
}
mantissa = b5_mantissas[get_bits(gbc, 4)];
break;
default: /* 6 to 15 */
- mantissa = get_bits(gbc, quantization_tab[bap]);
/* Shift mantissa and sign-extend it. */
- mantissa = (mantissa << (32-quantization_tab[bap]))>>8;
+ mantissa = get_sbits(gbc, quantization_tab[bap]);
+ mantissa <<= 24 - quantization_tab[bap];
break;
}
coeffs[freq] = mantissa >> exps[freq];
{
int bnd, i;
int end, bndend;
- int tmp0, tmp1;
end = FFMIN(s->end_freq[1], s->end_freq[2]);
if(s->rematrixing_flags[bnd]) {
bndend = FFMIN(end, ff_ac3_rematrix_band_tab[bnd+1]);
for(i=ff_ac3_rematrix_band_tab[bnd]; i<bndend; i++) {
- tmp0 = s->fixed_coeffs[1][i];
- tmp1 = s->fixed_coeffs[2][i];
- s->fixed_coeffs[1][i] = tmp0 + tmp1;
- s->fixed_coeffs[2][i] = tmp0 - tmp1;
+ int tmp0 = s->fixed_coeffs[1][i];
+ s->fixed_coeffs[1][i] += s->fixed_coeffs[2][i];
+ s->fixed_coeffs[2][i] = tmp0 - s->fixed_coeffs[2][i];
}
}
}
static inline void do_imdct(AC3DecodeContext *s, int channels)
{
int ch;
- float add_bias = s->add_bias;
- if(s->out_channels==1 && channels>1)
- add_bias *= LEVEL_MINUS_3DB; // compensate for the gain in downmix
for (ch=1; ch<=channels; ch++) {
if (s->block_switch[ch]) {
float *x = s->tmp_output+128;
for(i=0; i<128; i++)
x[i] = s->transform_coeffs[ch][2*i];
- ff_imdct_half(&s->imdct_256, s->tmp_output, x);
- s->dsp.vector_fmul_window(s->output[ch-1], s->delay[ch-1], s->tmp_output, s->window, add_bias, 128);
+ s->imdct_256.imdct_half(&s->imdct_256, s->tmp_output, x);
+ s->dsp.vector_fmul_window(s->output[ch-1], s->delay[ch-1], s->tmp_output, s->window, 128);
for(i=0; i<128; i++)
x[i] = s->transform_coeffs[ch][2*i+1];
- ff_imdct_half(&s->imdct_256, s->delay[ch-1], x);
+ s->imdct_256.imdct_half(&s->imdct_256, s->delay[ch-1], x);
} else {
- ff_imdct_half(&s->imdct_512, s->tmp_output, s->transform_coeffs[ch]);
- s->dsp.vector_fmul_window(s->output[ch-1], s->delay[ch-1], s->tmp_output, s->window, add_bias, 128);
+ s->imdct_512.imdct_half(&s->imdct_512, s->tmp_output, s->transform_coeffs[ch]);
+ s->dsp.vector_fmul_window(s->output[ch-1], s->delay[ch-1], s->tmp_output, s->window, 128);
memcpy(s->delay[ch-1], s->tmp_output+128, 128*sizeof(float));
}
}
do {
if(get_bits1(gbc)) {
s->dynamic_range[i] = ((dynamic_range_tab[get_bits(gbc, 8)]-1.0) *
- s->avctx->drc_scale)+1.0;
+ s->drc_scale)+1.0;
} else if(blk == 0) {
s->dynamic_range[i] = 1.0f;
}
/* spectral extension strategy */
if (s->eac3 && (!blk || get_bits1(gbc))) {
- if (get_bits1(gbc)) {
- av_log_missing_feature(s->avctx, "Spectral extension", 1);
- return -1;
+ s->spx_in_use = get_bits1(gbc);
+ if (s->spx_in_use) {
+ int dst_start_freq, dst_end_freq, src_start_freq,
+ start_subband, end_subband;
+
+ /* determine which channels use spx */
+ if (s->channel_mode == AC3_CHMODE_MONO) {
+ s->channel_uses_spx[1] = 1;
+ } else {
+ for (ch = 1; ch <= fbw_channels; ch++)
+ s->channel_uses_spx[ch] = get_bits1(gbc);
+ }
+
+ /* get the frequency bins of the spx copy region and the spx start
+ and end subbands */
+ dst_start_freq = get_bits(gbc, 2);
+ start_subband = get_bits(gbc, 3) + 2;
+ if (start_subband > 7)
+ start_subband += start_subband - 7;
+ end_subband = get_bits(gbc, 3) + 5;
+ if (end_subband > 7)
+ end_subband += end_subband - 7;
+ dst_start_freq = dst_start_freq * 12 + 25;
+ src_start_freq = start_subband * 12 + 25;
+ dst_end_freq = end_subband * 12 + 25;
+
+ /* check validity of spx ranges */
+ if (start_subband >= end_subband) {
+ av_log(s->avctx, AV_LOG_ERROR, "invalid spectral extension "
+ "range (%d >= %d)\n", start_subband, end_subband);
+ return -1;
+ }
+ if (dst_start_freq >= src_start_freq) {
+ av_log(s->avctx, AV_LOG_ERROR, "invalid spectral extension "
+ "copy start bin (%d >= %d)\n", dst_start_freq, src_start_freq);
+ return -1;
+ }
+
+ s->spx_dst_start_freq = dst_start_freq;
+ s->spx_src_start_freq = src_start_freq;
+ s->spx_dst_end_freq = dst_end_freq;
+
+ decode_band_structure(gbc, blk, s->eac3, 0,
+ start_subband, end_subband,
+ ff_eac3_default_spx_band_struct,
+ &s->num_spx_bands,
+ s->spx_band_sizes);
+ } else {
+ for (ch = 1; ch <= fbw_channels; ch++) {
+ s->channel_uses_spx[ch] = 0;
+ s->first_spx_coords[ch] = 1;
+ }
}
- /* TODO: parse spectral extension strategy info */
}
- /* TODO: spectral extension coordinates */
+ /* spectral extension coordinates */
+ if (s->spx_in_use) {
+ for (ch = 1; ch <= fbw_channels; ch++) {
+ if (s->channel_uses_spx[ch]) {
+ if (s->first_spx_coords[ch] || get_bits1(gbc)) {
+ float spx_blend;
+ int bin, master_spx_coord;
+
+ s->first_spx_coords[ch] = 0;
+ spx_blend = get_bits(gbc, 5) * (1.0f/32);
+ master_spx_coord = get_bits(gbc, 2) * 3;
+
+ bin = s->spx_src_start_freq;
+ for (bnd = 0; bnd < s->num_spx_bands; bnd++) {
+ int bandsize;
+ int spx_coord_exp, spx_coord_mant;
+ float nratio, sblend, nblend, spx_coord;
+
+ /* calculate blending factors */
+ bandsize = s->spx_band_sizes[bnd];
+ nratio = ((float)((bin + (bandsize >> 1))) / s->spx_dst_end_freq) - spx_blend;
+ nratio = av_clipf(nratio, 0.0f, 1.0f);
+ nblend = sqrtf(3.0f * nratio); // noise is scaled by sqrt(3) to give unity variance
+ sblend = sqrtf(1.0f - nratio);
+ bin += bandsize;
+
+ /* decode spx coordinates */
+ spx_coord_exp = get_bits(gbc, 4);
+ spx_coord_mant = get_bits(gbc, 2);
+ if (spx_coord_exp == 15) spx_coord_mant <<= 1;
+ else spx_coord_mant += 4;
+ spx_coord_mant <<= (25 - spx_coord_exp - master_spx_coord);
+ spx_coord = spx_coord_mant * (1.0f/(1<<23));
+
+ /* multiply noise and signal blending factors by spx coordinate */
+ s->spx_noise_blend [ch][bnd] = nblend * spx_coord;
+ s->spx_signal_blend[ch][bnd] = sblend * spx_coord;
+ }
+ }
+ } else {
+ s->first_spx_coords[ch] = 1;
+ }
+ }
+ }
/* coupling strategy */
if (s->eac3 ? s->cpl_strategy_exists[blk] : get_bits1(gbc)) {
s->phase_flags_in_use = get_bits1(gbc);
/* coupling frequency range */
- /* TODO: modify coupling end freq if spectral extension is used */
cpl_start_subband = get_bits(gbc, 4);
- cpl_end_subband = get_bits(gbc, 4) + 3;
+ cpl_end_subband = s->spx_in_use ? (s->spx_src_start_freq - 37) / 12 :
+ get_bits(gbc, 4) + 3;
if (cpl_start_subband >= cpl_end_subband) {
av_log(s->avctx, AV_LOG_ERROR, "invalid coupling range (%d >= %d)\n",
cpl_start_subband, cpl_end_subband);
if (channel_mode == AC3_CHMODE_STEREO) {
if ((s->eac3 && !blk) || get_bits1(gbc)) {
s->num_rematrixing_bands = 4;
- if(cpl_in_use && s->start_freq[CPL_CH] <= 61)
+ if (cpl_in_use && s->start_freq[CPL_CH] <= 61) {
s->num_rematrixing_bands -= 1 + (s->start_freq[CPL_CH] == 37);
+ } else if (s->spx_in_use && s->spx_src_start_freq <= 61) {
+ s->num_rematrixing_bands--;
+ }
for(bnd=0; bnd<s->num_rematrixing_bands; bnd++)
s->rematrixing_flags[bnd] = get_bits1(gbc);
} else if (!blk) {
- av_log(s->avctx, AV_LOG_ERROR, "new rematrixing strategy must be present in block 0\n");
- return -1;
+ av_log(s->avctx, AV_LOG_WARNING, "Warning: new rematrixing strategy not present in block 0\n");
+ s->num_rematrixing_bands = 0;
}
}
int prev = s->end_freq[ch];
if (s->channel_in_cpl[ch])
s->end_freq[ch] = s->start_freq[CPL_CH];
+ else if (s->channel_uses_spx[ch])
+ s->end_freq[ch] = s->spx_src_start_freq;
else {
int bandwidth_code = get_bits(gbc, 6);
if (bandwidth_code > 60) {
/* channel delta offset, len and bit allocation */
for (ch = !cpl_in_use; ch <= fbw_channels; ch++) {
if (s->dba_mode[ch] == DBA_NEW) {
- s->dba_nsegs[ch] = get_bits(gbc, 3);
- for (seg = 0; seg <= s->dba_nsegs[ch]; seg++) {
+ s->dba_nsegs[ch] = get_bits(gbc, 3) + 1;
+ for (seg = 0; seg < s->dba_nsegs[ch]; seg++) {
s->dba_offsets[ch][seg] = get_bits(gbc, 5);
s->dba_lengths[ch][seg] = get_bits(gbc, 4);
s->dba_values[ch][seg] = get_bits(gbc, 3);
/* Compute bit allocation */
const uint8_t *bap_tab = s->channel_uses_aht[ch] ?
ff_eac3_hebap_tab : ff_ac3_bap_tab;
- ff_ac3_bit_alloc_calc_bap(s->mask[ch], s->psd[ch],
+ s->ac3dsp.bit_alloc_calc_bap(s->mask[ch], s->psd[ch],
s->start_freq[ch], s->end_freq[ch],
s->snr_offset[ch],
s->bit_alloc_params.floor,
/* TODO: generate enhanced coupling coordinates and uncouple */
- /* TODO: apply spectral extension */
-
/* recover coefficients if rematrixing is in use */
if(s->channel_mode == AC3_CHMODE_STEREO)
do_rematrixing(s);
for(ch=1; ch<=s->channels; ch++) {
float gain = s->mul_bias / 4194304.0f;
if(s->channel_mode == AC3_CHMODE_DUALMONO) {
- gain *= s->dynamic_range[ch-1];
+ gain *= s->dynamic_range[2-ch];
} else {
gain *= s->dynamic_range[0];
}
- s->dsp.int32_to_float_fmul_scalar(s->transform_coeffs[ch], s->fixed_coeffs[ch], gain, 256);
+ s->fmt_conv.int32_to_float_fmul_scalar(s->transform_coeffs[ch], s->fixed_coeffs[ch], gain, 256);
+ }
+
+ /* apply spectral extension to high frequency bins */
+ if (s->spx_in_use && CONFIG_EAC3_DECODER) {
+ ff_eac3_apply_spectral_extension(s);
}
/* downmix and MDCT. order depends on whether block switching is used for
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
AC3DecodeContext *s = avctx->priv_data;
- int16_t *out_samples = (int16_t *)data;
+ float *out_samples_flt = data;
+ int16_t *out_samples_s16 = data;
int blk, ch, err;
const uint8_t *channel_map;
const float *output[AC3_MAX_CHANNELS];
- /* initialize the GetBitContext with the start of valid AC-3 Frame */
- if (s->input_buffer) {
- /* copy input buffer to decoder context to avoid reading past the end
- of the buffer, which can be caused by a damaged input stream. */
+ /* copy input buffer to decoder context to avoid reading past the end
+ of the buffer, which can be caused by a damaged input stream. */
+ if (buf_size >= 2 && AV_RB16(buf) == 0x770B) {
+ // seems to be byte-swapped AC-3
+ int cnt = FFMIN(buf_size, AC3_FRAME_BUFFER_SIZE) >> 1;
+ s->dsp.bswap16_buf((uint16_t *)s->input_buffer, (const uint16_t *)buf, cnt);
+ } else
memcpy(s->input_buffer, buf, FFMIN(buf_size, AC3_FRAME_BUFFER_SIZE));
- init_get_bits(&s->gbc, s->input_buffer, buf_size * 8);
- } else {
- init_get_bits(&s->gbc, buf, buf_size * 8);
- }
+ buf = s->input_buffer;
+ /* initialize the GetBitContext with the start of valid AC-3 Frame */
+ init_get_bits(&s->gbc, buf, buf_size * 8);
/* parse the syncinfo */
*data_size = 0;
err = parse_frame_header(s);
- /* check that reported frame size fits in input buffer */
- if(s->frame_size > buf_size) {
- av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
- err = AAC_AC3_PARSE_ERROR_FRAME_SIZE;
- }
-
- /* check for crc mismatch */
- if(err != AAC_AC3_PARSE_ERROR_FRAME_SIZE && avctx->error_recognition >= FF_ER_CAREFUL) {
- if(av_crc(av_crc_get_table(AV_CRC_16_ANSI), 0, &buf[2], s->frame_size-2)) {
- av_log(avctx, AV_LOG_ERROR, "frame CRC mismatch\n");
- err = AAC_AC3_PARSE_ERROR_CRC;
- }
- }
-
- if(err && err != AAC_AC3_PARSE_ERROR_CRC) {
+ if (err) {
switch(err) {
case AAC_AC3_PARSE_ERROR_SYNC:
av_log(avctx, AV_LOG_ERROR, "frame sync error\n");
av_log(avctx, AV_LOG_ERROR, "invalid header\n");
break;
}
+ } else {
+ /* check that reported frame size fits in input buffer */
+ if (s->frame_size > buf_size) {
+ av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
+ err = AAC_AC3_PARSE_ERROR_FRAME_SIZE;
+ } else if (avctx->error_recognition >= FF_ER_CAREFUL) {
+ /* check for crc mismatch */
+ if (av_crc(av_crc_get_table(AV_CRC_16_ANSI), 0, &buf[2], s->frame_size-2)) {
+ av_log(avctx, AV_LOG_ERROR, "frame CRC mismatch\n");
+ err = AAC_AC3_PARSE_ERROR_CRC;
+ }
+ }
}
/* if frame is ok, set audio parameters */
if(s->out_channels < s->channels)
s->output_mode = s->out_channels == 1 ? AC3_CHMODE_MONO : AC3_CHMODE_STEREO;
}
+ /* set audio service type based on bitstream mode for AC-3 */
+ avctx->audio_service_type = s->bitstream_mode;
+ if (s->bitstream_mode == 0x7 && s->channels > 1)
+ avctx->audio_service_type = AV_AUDIO_SERVICE_TYPE_KARAOKE;
/* decode the audio blocks */
channel_map = ff_ac3_dec_channel_map[s->output_mode & ~AC3_OUTPUT_LFEON][s->lfe_on];
av_log(avctx, AV_LOG_ERROR, "error decoding the audio block\n");
err = 1;
}
- s->dsp.float_to_int16_interleave(out_samples, output, 256, s->out_channels);
- out_samples += 256 * s->out_channels;
+ if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
+ s->fmt_conv.float_interleave(out_samples_flt, output, 256,
+ s->out_channels);
+ out_samples_flt += 256 * s->out_channels;
+ } else {
+ s->fmt_conv.float_to_int16_interleave(out_samples_s16, output, 256,
+ s->out_channels);
+ out_samples_s16 += 256 * s->out_channels;
+ }
}
- *data_size = s->num_blocks * 256 * avctx->channels * sizeof (int16_t);
- return s->frame_size;
+ *data_size = s->num_blocks * 256 * avctx->channels *
+ av_get_bytes_per_sample(avctx->sample_fmt);
+ return FFMIN(buf_size, s->frame_size);
}
/**
ff_mdct_end(&s->imdct_512);
ff_mdct_end(&s->imdct_256);
- av_freep(&s->input_buffer);
-
return 0;
}
-AVCodec ac3_decoder = {
+#define OFFSET(x) offsetof(AC3DecodeContext, x)
+#define PAR (AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM)
+static const AVOption options[] = {
+ { "drc_scale", "percentage of dynamic range compression to apply", OFFSET(drc_scale), FF_OPT_TYPE_FLOAT, {1.0}, 0.0, 1.0, PAR },
+ { NULL},
+};
+
+static const AVClass ac3_decoder_class = {
+ .class_name = "AC3 decoder",
+ .item_name = av_default_item_name,
+ .option = options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
+
+AVCodec ff_ac3_decoder = {
.name = "ac3",
- .type = CODEC_TYPE_AUDIO,
+ .type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_AC3,
.priv_data_size = sizeof (AC3DecodeContext),
.init = ac3_decode_init,
.close = ac3_decode_end,
.decode = ac3_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
+ .sample_fmts = (const enum AVSampleFormat[]) {
+ AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
+ },
+ .priv_class = &ac3_decoder_class,
};
#if CONFIG_EAC3_DECODER
-AVCodec eac3_decoder = {
+static const AVClass eac3_decoder_class = {
+ .class_name = "E-AC3 decoder",
+ .item_name = av_default_item_name,
+ .option = options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
+AVCodec ff_eac3_decoder = {
.name = "eac3",
- .type = CODEC_TYPE_AUDIO,
+ .type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_EAC3,
.priv_data_size = sizeof (AC3DecodeContext),
.init = ac3_decode_init,
.close = ac3_decode_end,
.decode = ac3_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52B (AC-3, E-AC-3)"),
+ .sample_fmts = (const enum AVSampleFormat[]) {
+ AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
+ },
+ .priv_class = &eac3_decoder_class,
};
#endif
projects / ffmpeg / blobdiff