* Copyright (c) 2007-2008 Bartlomiej Wolowiec <bartek.wolowiec@gmail.com>
* Copyright (c) 2007 Justin Ruggles <justin.ruggles@gmail.com>
*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * 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.
*
- * Libav is distributed in the hope that it will be useful,
+ * 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 Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/** dynamic range table. converts codes to scale factors. */
static float dynamic_range_tab[256];
+static float heavy_dynamic_range_tab[256];
/** Adjustments in dB gain */
static const float gain_levels[9] = {
static inline int
symmetric_dequant(int code, int levels)
{
- return ((code - (levels >> 1)) << 24) / levels;
+ return ((code - (levels >> 1)) * (1 << 24)) / levels;
}
/*
int v = (i >> 5) - ((i >> 7) << 3) - 5;
dynamic_range_tab[i] = powf(2.0f, v) * ((i & 0x1F) | 0x20);
}
+
+ /* generate compr dynamic range table
+ reference: Section 7.7.2 Heavy Compression */
+ for (i = 0; i < 256; i++) {
+ int v = (i >> 4) - ((i >> 7) << 4) - 4;
+ heavy_dynamic_range_tab[i] = powf(2.0f, v) * ((i & 0xF) | 0x10);
+ }
+
}
/**
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);
+ AC3_RENAME(ff_kbd_window_init)(s->window, 5.0, 256);
ff_bswapdsp_init(&s->bdsp);
- avpriv_float_dsp_init(&s->fdsp, avctx->flags & AV_CODEC_FLAG_BITEXACT);
- ff_ac3dsp_init(&s->ac3dsp, avctx->flags & AV_CODEC_FLAG_BITEXACT);
+
+#if (USE_FIXED)
+ s->fdsp = avpriv_alloc_fixed_dsp(avctx->flags & AV_CODEC_FLAG_BITEXACT);
+#else
+ s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
ff_fmt_convert_init(&s->fmt_conv, avctx);
+#endif
+
+ ff_ac3dsp_init(&s->ac3dsp, avctx->flags & AV_CODEC_FLAG_BITEXACT);
av_lfg_init(&s->dith_state, 0);
- avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
+ if (USE_FIXED)
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
+ else
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
/* allow downmixing to stereo or mono */
#if FF_API_REQUEST_CHANNELS
/* read the rest of the bsi. read twice for dual mono mode. */
i = !s->channel_mode;
do {
- skip_bits(gbc, 5); // skip dialog normalization
- if (get_bits1(gbc))
- skip_bits(gbc, 8); //skip compression
+ s->dialog_normalization[(!s->channel_mode)-i] = -get_bits(gbc, 5);
+ if (s->dialog_normalization[(!s->channel_mode)-i] == 0) {
+ s->dialog_normalization[(!s->channel_mode)-i] = -31;
+ }
+ if (s->target_level != 0) {
+ s->level_gain[(!s->channel_mode)-i] = powf(2.0f,
+ (float)(s->target_level -
+ s->dialog_normalization[(!s->channel_mode)-i])/6.0f);
+ }
+ if (s->compression_exists[(!s->channel_mode)-i] = get_bits1(gbc)) {
+ s->heavy_dynamic_range[(!s->channel_mode)-i] =
+ AC3_HEAVY_RANGE(get_bits(gbc, 8));
+ }
if (get_bits1(gbc))
skip_bits(gbc, 8); //skip language code
if (get_bits1(gbc))
*/
static int parse_frame_header(AC3DecodeContext *s)
{
- AC3HeaderInfo hdr;
+ AC3HeaderInfo hdr, *phdr=&hdr;
int err;
- err = avpriv_ac3_parse_header(&s->gbc, &hdr);
+ err = avpriv_ac3_parse_header2(&s->gbc, &phdr);
if (err)
return err;
float cmix = gain_levels[s-> center_mix_level];
float smix = gain_levels[s->surround_mix_level];
float norm0, norm1;
+ float downmix_coeffs[AC3_MAX_CHANNELS][2];
for (i = 0; i < s->fbw_channels; i++) {
- s->downmix_coeffs[i][0] = gain_levels[ac3_default_coeffs[s->channel_mode][i][0]];
- s->downmix_coeffs[i][1] = gain_levels[ac3_default_coeffs[s->channel_mode][i][1]];
+ downmix_coeffs[i][0] = gain_levels[ac3_default_coeffs[s->channel_mode][i][0]];
+ downmix_coeffs[i][1] = gain_levels[ac3_default_coeffs[s->channel_mode][i][1]];
}
if (s->channel_mode > 1 && s->channel_mode & 1) {
- s->downmix_coeffs[1][0] = s->downmix_coeffs[1][1] = cmix;
+ downmix_coeffs[1][0] = downmix_coeffs[1][1] = cmix;
}
if (s->channel_mode == AC3_CHMODE_2F1R || s->channel_mode == AC3_CHMODE_3F1R) {
int nf = s->channel_mode - 2;
- s->downmix_coeffs[nf][0] = s->downmix_coeffs[nf][1] = smix * LEVEL_MINUS_3DB;
+ downmix_coeffs[nf][0] = downmix_coeffs[nf][1] = smix * LEVEL_MINUS_3DB;
}
if (s->channel_mode == AC3_CHMODE_2F2R || s->channel_mode == AC3_CHMODE_3F2R) {
int nf = s->channel_mode - 4;
- s->downmix_coeffs[nf][0] = s->downmix_coeffs[nf+1][1] = smix;
+ downmix_coeffs[nf][0] = downmix_coeffs[nf+1][1] = smix;
}
/* renormalize */
norm0 = norm1 = 0.0;
for (i = 0; i < s->fbw_channels; i++) {
- norm0 += s->downmix_coeffs[i][0];
- norm1 += s->downmix_coeffs[i][1];
+ norm0 += downmix_coeffs[i][0];
+ norm1 += downmix_coeffs[i][1];
}
norm0 = 1.0f / norm0;
norm1 = 1.0f / norm1;
for (i = 0; i < s->fbw_channels; i++) {
- s->downmix_coeffs[i][0] *= norm0;
- s->downmix_coeffs[i][1] *= norm1;
+ downmix_coeffs[i][0] *= norm0;
+ downmix_coeffs[i][1] *= norm1;
}
if (s->output_mode == AC3_CHMODE_MONO) {
for (i = 0; i < s->fbw_channels; i++)
- s->downmix_coeffs[i][0] = (s->downmix_coeffs[i][0] +
- s->downmix_coeffs[i][1]) * LEVEL_MINUS_3DB;
+ downmix_coeffs[i][0] = (downmix_coeffs[i][0] +
+ downmix_coeffs[i][1]) * LEVEL_MINUS_3DB;
+ }
+ for (i = 0; i < s->fbw_channels; i++) {
+ s->downmix_coeffs[i][0] = FIXR12(downmix_coeffs[i][0]);
+ s->downmix_coeffs[i][1] = FIXR12(downmix_coeffs[i][1]);
}
}
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);
+ MULH(s->fixed_coeffs[CPL_CH][bin] * (1 << 4), cpl_coord);
}
if (ch == 2 && s->phase_flags[band]) {
for (bin = band_start; bin < band_end; bin++)
case 0:
/* random noise with approximate range of -0.707 to 0.707 */
if (dither)
- mantissa = (av_lfg_get(&s->dith_state) / 362) - 5932275;
+ mantissa = (((av_lfg_get(&s->dith_state)>>8)*181)>>8) - 5931008;
else
mantissa = 0;
break;
break;
default: /* 6 to 15 */
/* Shift mantissa and sign-extend it. */
- mantissa = get_sbits(gbc, quantization_tab[bap]);
- mantissa <<= 24 - quantization_tab[bap];
+ if (bap > 15) {
+ av_log(s->avctx, AV_LOG_ERROR, "bap %d is invalid in plain AC-3\n", bap);
+ bap = 15;
+ }
+ mantissa = (unsigned)get_sbits(gbc, quantization_tab[bap]) << (24 - quantization_tab[bap]);
break;
}
coeffs[freq] = mantissa >> exps[freq];
/* if AHT is used, mantissas for all blocks are encoded in the first
block of the frame. */
int bin;
- if (!blk && CONFIG_EAC3_DECODER)
+ if (CONFIG_EAC3_DECODER && !blk)
ff_eac3_decode_transform_coeffs_aht_ch(s, ch);
for (bin = s->start_freq[ch]; bin < s->end_freq[ch]; bin++) {
s->fixed_coeffs[ch][bin] = s->pre_mantissa[ch][bin][blk] >> s->dexps[ch][bin];
for (ch = 1; ch <= channels; ch++) {
if (s->block_switch[ch]) {
int i;
- float *x = s->tmp_output + 128;
+ FFTSample *x = s->tmp_output + 128;
for (i = 0; i < 128; i++)
x[i] = s->transform_coeffs[ch][2 * i];
s->imdct_256.imdct_half(&s->imdct_256, s->tmp_output, x);
- s->fdsp.vector_fmul_window(s->outptr[ch - 1], s->delay[ch - 1],
+#if USE_FIXED
+ s->fdsp->vector_fmul_window_scaled(s->outptr[ch - 1], s->delay[ch - 1],
+ s->tmp_output, s->window, 128, 8);
+#else
+ s->fdsp->vector_fmul_window(s->outptr[ch - 1], s->delay[ch - 1],
s->tmp_output, s->window, 128);
+#endif
for (i = 0; i < 128; i++)
x[i] = s->transform_coeffs[ch][2 * i + 1];
s->imdct_256.imdct_half(&s->imdct_256, s->delay[ch - 1], x);
} else {
s->imdct_512.imdct_half(&s->imdct_512, s->tmp_output, s->transform_coeffs[ch]);
- s->fdsp.vector_fmul_window(s->outptr[ch - 1], s->delay[ch - 1],
+#if USE_FIXED
+ s->fdsp->vector_fmul_window_scaled(s->outptr[ch - 1], s->delay[ch - 1],
+ s->tmp_output, s->window, 128, 8);
+#else
+ s->fdsp->vector_fmul_window(s->outptr[ch - 1], s->delay[ch - 1],
s->tmp_output, s->window, 128);
- memcpy(s->delay[ch - 1], s->tmp_output + 128, 128 * sizeof(float));
+#endif
+ memcpy(s->delay[ch - 1], s->tmp_output + 128, 128 * sizeof(FFTSample));
}
}
}
if (get_bits1(gbc)) {
/* Allow asymmetric application of DRC when drc_scale > 1.
Amplification of quiet sounds is enhanced */
- float range = dynamic_range_tab[get_bits(gbc, 8)];
- if (range > 1.0 || s->drc_scale <= 1.0)
- s->dynamic_range[i] = powf(range, s->drc_scale);
+ int range_bits = get_bits(gbc, 8);
+ INTFLOAT range = AC3_RANGE(range_bits);
+ if (range_bits <= 127 || s->drc_scale <= 1.0)
+ s->dynamic_range[i] = AC3_DYNAMIC_RANGE(range);
else
s->dynamic_range[i] = range;
} else if (blk == 0) {
- s->dynamic_range[i] = 1.0f;
+ s->dynamic_range[i] = AC3_DYNAMIC_RANGE1;
}
} while (i--);
if (start_subband > 7)
start_subband += start_subband - 7;
end_subband = get_bits(gbc, 3) + 5;
+#if USE_FIXED
+ s->spx_dst_end_freq = end_freq_inv_tab[end_subband-5];
+#endif
if (end_subband > 7)
end_subband += end_subband - 7;
dst_start_freq = dst_start_freq * 12 + 25;
s->spx_dst_start_freq = dst_start_freq;
s->spx_src_start_freq = src_start_freq;
- s->spx_dst_end_freq = dst_end_freq;
+ if (!USE_FIXED)
+ s->spx_dst_end_freq = dst_end_freq;
decode_band_structure(gbc, blk, s->eac3, 0,
start_subband, end_subband,
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;
+ INTFLOAT spx_blend;
int bin, master_spx_coord;
s->first_spx_coords[ch] = 0;
- spx_blend = get_bits(gbc, 5) * (1.0f/32);
+ spx_blend = AC3_SPX_BLEND(get_bits(gbc, 5));
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 bandsize = s->spx_band_sizes[bnd];
int spx_coord_exp, spx_coord_mant;
- float nratio, sblend, nblend, spx_coord;
+ INTFLOAT nratio, sblend, nblend;
+#if USE_FIXED
+ /* calculate blending factors */
+ int64_t accu = ((bin << 23) + (bandsize << 22))
+ * (int64_t)s->spx_dst_end_freq;
+ nratio = (int)(accu >> 32);
+ nratio -= spx_blend << 18;
+
+ if (nratio < 0) {
+ nblend = 0;
+ sblend = 0x800000;
+ } else if (nratio > 0x7fffff) {
+ nblend = 14529495; // sqrt(3) in FP.23
+ sblend = 0;
+ } else {
+ nblend = fixed_sqrt(nratio, 23);
+ accu = (int64_t)nblend * 1859775393;
+ nblend = (int)((accu + (1<<29)) >> 30);
+ sblend = fixed_sqrt(0x800000 - nratio, 23);
+ }
+#else
+ float 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);
+#endif
bin += bandsize;
/* decode spx coordinates */
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 */
+#if USE_FIXED
+ accu = (int64_t)nblend * spx_coord_mant;
+ s->spx_noise_blend[ch][bnd] = (int)((accu + (1<<22)) >> 23);
+ accu = (int64_t)sblend * spx_coord_mant;
+ s->spx_signal_blend[ch][bnd] = (int)((accu + (1<<22)) >> 23);
+#else
+ spx_coord = spx_coord_mant * (1.0f / (1 << 23));
s->spx_noise_blend [ch][bnd] = nblend * spx_coord;
s->spx_signal_blend[ch][bnd] = sblend * spx_coord;
+#endif
}
}
} else {
/* apply scaling to coefficients (headroom, dynrng) */
for (ch = 1; ch <= s->channels; ch++) {
- float gain = 1.0 / 4194304.0f;
- if (s->channel_mode == AC3_CHMODE_DUALMONO) {
- gain *= s->dynamic_range[2 - ch];
- } else {
- gain *= s->dynamic_range[0];
- }
+ int audio_channel = 0;
+ INTFLOAT gain;
+ if (s->channel_mode == AC3_CHMODE_DUALMONO)
+ audio_channel = 2-ch;
+ if (s->heavy_compression && s->compression_exists[audio_channel])
+ gain = s->heavy_dynamic_range[audio_channel];
+ else
+ gain = s->dynamic_range[audio_channel];
+
+#if USE_FIXED
+ scale_coefs(s->transform_coeffs[ch], s->fixed_coeffs[ch], gain, 256);
+#else
+ if (s->target_level != 0)
+ gain = gain * s->level_gain[audio_channel];
+ gain *= 1.0 / 4194304.0f;
s->fmt_conv.int32_to_float_fmul_scalar(s->transform_coeffs[ch],
s->fixed_coeffs[ch], gain, 256);
+#endif
}
/* apply spectral extension to high frequency bins */
- if (s->spx_in_use && CONFIG_EAC3_DECODER) {
+ if (CONFIG_EAC3_DECODER && s->spx_in_use) {
ff_eac3_apply_spectral_extension(s);
}
do_imdct(s, s->channels);
if (downmix_output) {
+#if USE_FIXED
+ ac3_downmix_c_fixed16(s->outptr, s->downmix_coeffs,
+ s->out_channels, s->fbw_channels, 256);
+#else
s->ac3dsp.downmix(s->outptr, s->downmix_coeffs,
s->out_channels, s->fbw_channels, 256);
+#endif
}
} else {
if (downmix_output) {
- s->ac3dsp.downmix(s->xcfptr + 1, s->downmix_coeffs,
- s->out_channels, s->fbw_channels, 256);
+ s->ac3dsp.AC3_RENAME(downmix)(s->xcfptr + 1, s->downmix_coeffs,
+ s->out_channels, s->fbw_channels, 256);
}
if (downmix_output && !s->downmixed) {
s->downmixed = 1;
- s->ac3dsp.downmix(s->dlyptr, s->downmix_coeffs, s->out_channels,
- s->fbw_channels, 128);
+ s->ac3dsp.AC3_RENAME(downmix)(s->dlyptr, s->downmix_coeffs,
+ s->out_channels, s->fbw_channels, 128);
}
do_imdct(s, s->out_channels);
AC3DecodeContext *s = avctx->priv_data;
int blk, ch, err, ret;
const uint8_t *channel_map;
- const float *output[AC3_MAX_CHANNELS];
+ const SHORTFLOAT *output[AC3_MAX_CHANNELS];
enum AVMatrixEncoding matrix_encoding;
AVDownmixInfo *downmix_info;
memcpy(s->input_buffer, buf, FFMIN(buf_size, AC3_FRAME_BUFFER_SIZE));
buf = s->input_buffer;
/* initialize the GetBitContext with the start of valid AC-3 Frame */
- init_get_bits(&s->gbc, buf, buf_size * 8);
+ if ((ret = init_get_bits8(&s->gbc, buf, buf_size)) < 0)
+ return ret;
/* parse the syncinfo */
err = parse_frame_header(s);
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->err_recognition & AV_EF_CRCCHECK) {
+ } else if (avctx->err_recognition & (AV_EF_CRCCHECK|AV_EF_CAREFUL)) {
/* check for crc mismatch */
if (av_crc(av_crc_get_table(AV_CRC_16_ANSI), 0, &buf[2],
s->frame_size - 2)) {
s->output_mode = AC3_CHMODE_STEREO;
}
+ s->loro_center_mix_level = gain_levels[s-> center_mix_level];
+ s->loro_surround_mix_level = gain_levels[s->surround_mix_level];
+ s->ltrt_center_mix_level = LEVEL_MINUS_3DB;
+ s->ltrt_surround_mix_level = LEVEL_MINUS_3DB;
/* set downmixing coefficients if needed */
if (s->channels != s->out_channels && !((s->output_mode & AC3_OUTPUT_LFEON) &&
s->fbw_channels == s->out_channels)) {
/* get output buffer */
frame->nb_samples = s->num_blocks * AC3_BLOCK_SIZE;
- if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
- av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
- }
/* decode the audio blocks */
channel_map = ff_ac3_dec_channel_map[s->output_mode & ~AC3_OUTPUT_LFEON][s->lfe_on];
+ for (ch = 0; ch < AC3_MAX_CHANNELS; ch++) {
+ output[ch] = s->output[ch];
+ s->outptr[ch] = s->output[ch];
+ }
for (ch = 0; ch < s->channels; ch++) {
if (ch < s->out_channels)
- s->outptr[channel_map[ch]] = (float *)frame->data[ch];
- else
- s->outptr[ch] = s->output[ch];
- output[ch] = s->output[ch];
+ s->outptr[channel_map[ch]] = (SHORTFLOAT *)frame->data[ch];
}
for (blk = 0; blk < s->num_blocks; blk++) {
if (!err && decode_audio_block(s, blk)) {
}
if (err)
for (ch = 0; ch < s->out_channels; ch++)
- memcpy(s->outptr[channel_map[ch]], output[ch], sizeof(**output) * AC3_BLOCK_SIZE);
+ memcpy(((SHORTFLOAT*)frame->data[ch]) + AC3_BLOCK_SIZE*blk, output[ch], AC3_BLOCK_SIZE*sizeof(SHORTFLOAT));
for (ch = 0; ch < s->out_channels; ch++)
output[ch] = s->outptr[channel_map[ch]];
- for (ch = 0; ch < s->out_channels; ch++)
- s->outptr[ch] += AC3_BLOCK_SIZE;
+ for (ch = 0; ch < s->out_channels; ch++) {
+ if (!ch || channel_map[ch])
+ s->outptr[channel_map[ch]] += AC3_BLOCK_SIZE;
+ }
}
+ av_frame_set_decode_error_flags(frame, err ? FF_DECODE_ERROR_INVALID_BITSTREAM : 0);
+
/* keep last block for error concealment in next frame */
for (ch = 0; ch < s->out_channels; ch++)
- memcpy(s->output[ch], output[ch], sizeof(**output) * AC3_BLOCK_SIZE);
+ memcpy(s->output[ch], output[ch], AC3_BLOCK_SIZE*sizeof(SHORTFLOAT));
/*
* AVMatrixEncoding
AC3DecodeContext *s = avctx->priv_data;
ff_mdct_end(&s->imdct_512);
ff_mdct_end(&s->imdct_256);
+ av_freep(&s->fdsp);
return 0;
}
#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), AV_OPT_TYPE_FLOAT, {.dbl = 1.0}, 0.0, 6.0, 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",
- .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
- .type = AVMEDIA_TYPE_AUDIO,
- .id = AV_CODEC_ID_AC3,
- .priv_data_size = sizeof (AC3DecodeContext),
- .init = ac3_decode_init,
- .close = ac3_decode_end,
- .decode = ac3_decode_frame,
- .capabilities = CODEC_CAP_DR1,
- .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
- AV_SAMPLE_FMT_NONE },
- .priv_class = &ac3_decoder_class,
-};
-
-#if CONFIG_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",
- .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52B (AC-3, E-AC-3)"),
- .type = AVMEDIA_TYPE_AUDIO,
- .id = AV_CODEC_ID_EAC3,
- .priv_data_size = sizeof (AC3DecodeContext),
- .init = ac3_decode_init,
- .close = ac3_decode_end,
- .decode = ac3_decode_frame,
- .capabilities = CODEC_CAP_DR1,
- .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
- AV_SAMPLE_FMT_NONE },
- .priv_class = &eac3_decoder_class,
-};
-#endif
projects / ffmpeg / blobdiff