#include "libavutil/channel_layout.h"
#include "libavutil/crc.h"
+#include "libavutil/downmix_info.h"
#include "libavutil/opt.h"
+#include "bswapdsp.h"
#include "internal.h"
#include "aac_ac3_parser.h"
#include "ac3_parser.h"
/** 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] = {
LEVEL_MINUS_9DB
};
+/** Adjustments in dB gain (LFE, +10 to -21 dB) */
+static const float gain_levels_lfe[32] = {
+ 3.162275, 2.818382, 2.511886, 2.238719, 1.995261, 1.778278, 1.584893,
+ 1.412536, 1.258924, 1.122018, 1.000000, 0.891251, 0.794328, 0.707946,
+ 0.630957, 0.562341, 0.501187, 0.446683, 0.398107, 0.354813, 0.316227,
+ 0.281838, 0.251188, 0.223872, 0.199526, 0.177828, 0.158489, 0.141253,
+ 0.125892, 0.112201, 0.100000, 0.089125
+};
+
/**
* Table for default stereo downmixing coefficients
* reference: Section 7.8.2 Downmixing Into Two Channels
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);
- ff_dsputil_init(&s->dsp, avctx);
+ AC3_RENAME(ff_kbd_window_init)(s->window, 5.0, 256);
+ ff_bswapdsp_init(&s->bdsp);
+
+#if (USE_FIXED)
+ s->fdsp = avpriv_alloc_fixed_dsp(avctx->flags & CODEC_FLAG_BITEXACT);
+#else
avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
+#endif
+
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);
- 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))
skip_bits(gbc, 2); //skip copyright bit and original bitstream bit
- /* skip the timecodes (or extra bitstream information for Alternate Syntax)
- TODO: read & use the xbsi1 downmix levels */
- if (get_bits1(gbc))
- skip_bits(gbc, 14); //skip timecode1 / xbsi1
- if (get_bits1(gbc))
- skip_bits(gbc, 14); //skip timecode2 / xbsi2
+ /* skip the timecodes or parse the Alternate Bit Stream Syntax */
+ if (s->bitstream_id != 6) {
+ if (get_bits1(gbc))
+ skip_bits(gbc, 14); //skip timecode1
+ if (get_bits1(gbc))
+ skip_bits(gbc, 14); //skip timecode2
+ } else {
+ if (get_bits1(gbc)) {
+ s->preferred_downmix = get_bits(gbc, 2);
+ s->center_mix_level_ltrt = get_bits(gbc, 3);
+ s->surround_mix_level_ltrt = av_clip(get_bits(gbc, 3), 3, 7);
+ s->center_mix_level = get_bits(gbc, 3);
+ s->surround_mix_level = av_clip(get_bits(gbc, 3), 3, 7);
+ }
+ if (get_bits1(gbc)) {
+ s->dolby_surround_ex_mode = get_bits(gbc, 2);
+ s->dolby_headphone_mode = get_bits(gbc, 2);
+ skip_bits(gbc, 10); // skip adconvtyp (1), xbsi2 (8), encinfo (1)
+ }
+ }
/* skip additional bitstream info */
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;
/* get decoding parameters from header info */
s->bit_alloc_params.sr_code = hdr.sr_code;
+ s->bitstream_id = hdr.bitstream_id;
s->bitstream_mode = hdr.bitstream_mode;
s->channel_mode = hdr.channel_mode;
s->lfe_on = hdr.lfe_on;
s->fbw_channels = s->channels - s->lfe_on;
s->lfe_ch = s->fbw_channels + 1;
s->frame_size = hdr.frame_size;
+ s->preferred_downmix = AC3_DMIXMOD_NOTINDICATED;
s->center_mix_level = hdr.center_mix_level;
+ s->center_mix_level_ltrt = 4; // -3.0dB
s->surround_mix_level = hdr.surround_mix_level;
+ s->surround_mix_level_ltrt = 4; // -3.0dB
+ s->lfe_mix_level_exists = 0;
s->num_blocks = hdr.num_blocks;
s->frame_type = hdr.frame_type;
s->substreamid = hdr.substreamid;
+ s->dolby_surround_mode = hdr.dolby_surround_mode;
+ s->dolby_surround_ex_mode = AC3_DSUREXMOD_NOTINDICATED;
+ s->dolby_headphone_mode = AC3_DHEADPHONMOD_NOTINDICATED;
if (s->lfe_on) {
s->start_freq[s->lfe_ch] = 0;
s->channel_in_cpl[s->lfe_ch] = 0;
}
- if (hdr.bitstream_id <= 10) {
+ if (s->bitstream_id <= 10) {
s->eac3 = 0;
s->snr_offset_strategy = 2;
s->block_switch_syntax = 1;
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]);
}
}
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);
+#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]);
+#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));
}
}
}
i = !s->channel_mode;
do {
if (get_bits1(gbc)) {
- s->dynamic_range[i] = powf(dynamic_range_tab[get_bits(gbc, 8)],
- s->drc_scale);
+ /* Allow asymmetric application of DRC when drc_scale > 1.
+ Amplification of quiet sounds is enhanced */
+ 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];
+#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 spx_coord_exp, spx_coord_mant;
- float nratio, sblend, nblend, spx_coord;
+ INTFLOAT nratio, sblend, nblend;
+#if USE_FIXED
+ int64_t accu;
+ /* calculate blending factors */
+ bandsize = s->spx_band_sizes[bnd];
+ accu = (int64_t)((bin << 23) + (bandsize << 22)) * 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 = 0x800000;
+ 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];
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 */
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;
/* 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);
+ s->bdsp.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));
buf = s->input_buffer;
}
for (ch = 0; ch < s->channels; ch++) {
if (ch < s->out_channels)
- s->outptr[channel_map[ch]] = (float *)frame->data[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(((float*)frame->data[ch]) + AC3_BLOCK_SIZE*blk, 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++) {
/* 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
+ *
+ * Check whether the input layout is compatible, and make sure we're not
+ * downmixing (else the matrix encoding is no longer applicable).
+ */
+ matrix_encoding = AV_MATRIX_ENCODING_NONE;
+ if (s->channel_mode == AC3_CHMODE_STEREO &&
+ s->channel_mode == (s->output_mode & ~AC3_OUTPUT_LFEON)) {
+ if (s->dolby_surround_mode == AC3_DSURMOD_ON)
+ matrix_encoding = AV_MATRIX_ENCODING_DOLBY;
+ else if (s->dolby_headphone_mode == AC3_DHEADPHONMOD_ON)
+ matrix_encoding = AV_MATRIX_ENCODING_DOLBYHEADPHONE;
+ } else if (s->channel_mode >= AC3_CHMODE_2F2R &&
+ s->channel_mode == (s->output_mode & ~AC3_OUTPUT_LFEON)) {
+ switch (s->dolby_surround_ex_mode) {
+ case AC3_DSUREXMOD_ON: // EX or PLIIx
+ matrix_encoding = AV_MATRIX_ENCODING_DOLBYEX;
+ break;
+ case AC3_DSUREXMOD_PLIIZ:
+ matrix_encoding = AV_MATRIX_ENCODING_DPLIIZ;
+ break;
+ default: // not indicated or off
+ break;
+ }
+ }
+ if ((ret = ff_side_data_update_matrix_encoding(frame, matrix_encoding)) < 0)
+ return ret;
+
+ /* AVDownmixInfo */
+ if ((downmix_info = av_downmix_info_update_side_data(frame))) {
+ switch (s->preferred_downmix) {
+ case AC3_DMIXMOD_LTRT:
+ downmix_info->preferred_downmix_type = AV_DOWNMIX_TYPE_LTRT;
+ break;
+ case AC3_DMIXMOD_LORO:
+ downmix_info->preferred_downmix_type = AV_DOWNMIX_TYPE_LORO;
+ break;
+ case AC3_DMIXMOD_DPLII:
+ downmix_info->preferred_downmix_type = AV_DOWNMIX_TYPE_DPLII;
+ break;
+ default:
+ downmix_info->preferred_downmix_type = AV_DOWNMIX_TYPE_UNKNOWN;
+ break;
+ }
+ downmix_info->center_mix_level = gain_levels[s-> center_mix_level];
+ downmix_info->center_mix_level_ltrt = gain_levels[s-> center_mix_level_ltrt];
+ downmix_info->surround_mix_level = gain_levels[s-> surround_mix_level];
+ downmix_info->surround_mix_level_ltrt = gain_levels[s->surround_mix_level_ltrt];
+ if (s->lfe_mix_level_exists)
+ downmix_info->lfe_mix_level = gain_levels_lfe[s->lfe_mix_level];
+ else
+ downmix_info->lfe_mix_level = 0.0; // -inf dB
+ } else
+ return AVERROR(ENOMEM);
*got_frame_ptr = 1;
AC3DecodeContext *s = avctx->priv_data;
ff_mdct_end(&s->imdct_512);
ff_mdct_end(&s->imdct_256);
+#if (USE_FIXED)
+ av_freep(&s->fdsp);
+#endif
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, 1.0, PAR },
-
-{"dmix_mode", "Preferred Stereo Downmix Mode", OFFSET(preferred_stereo_downmix), AV_OPT_TYPE_INT, {.i64 = -1 }, -1, 2, 0, "dmix_mode"},
-{"ltrt_cmixlev", "Lt/Rt Center Mix Level", OFFSET(ltrt_center_mix_level), AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, 0},
-{"ltrt_surmixlev", "Lt/Rt Surround Mix Level", OFFSET(ltrt_surround_mix_level), AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, 0},
-{"loro_cmixlev", "Lo/Ro Center Mix Level", OFFSET(loro_center_mix_level), AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, 0},
-{"loro_surmixlev", "Lo/Ro Surround Mix Level", OFFSET(loro_surround_mix_level), AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, 0},
-
- { 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 / platform / framework / web / crosswalk.git / blobdiff