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 scale value for float to int16 conversion */
- s->mul_bias = 32767.0f;
+ if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
+ s->mul_bias = 1.0f;
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+ } else {
+ s->mul_bias = 32767.0f;
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ }
/* allow downmixing to stereo or mono */
if (avctx->channels > 0 && avctx->request_channels > 0 &&
if (!s->input_buffer)
return AVERROR(ENOMEM);
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
return 0;
}
/* 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,
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];
av_log(avctx, AV_LOG_ERROR, "error decoding the audio block\n");
err = 1;
}
- s->fmt_conv.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);
+ *data_size = s->num_blocks * 256 * avctx->channels *
+ (av_get_bits_per_sample_fmt(avctx->sample_fmt) / 8);
return FFMIN(buf_size, s->frame_size);
}
.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
+ },
};
#if CONFIG_EAC3_DECODER
.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
+ },
};
#endif