ff_fmt_convert_init(&s->fmt_conv, avctx);
av_lfg_init(&s->dith_state, 0);
+ /* 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->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 &&
avctx->request_channels < avctx->channels &&
}
s->downmixed = 1;
- if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
- avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
- s->mul_bias = 1.0f;
- } else {
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
- /* set scale value for float to int16 conversion */
- s->mul_bias = 32767.0f;
- }
return 0;
}
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
AC3DecodeContext *s = avctx->priv_data;
- float *out_samples_flt = (float *)data;
- int16_t *out_samples = (int16_t *)data;
+ float *out_samples_flt = data;
+ int16_t *out_samples_s16 = data;
int blk, ch, err;
+ int data_size_orig, data_size_tmp;
const uint8_t *channel_map;
const float *output[AC3_MAX_CHANNELS];
init_get_bits(&s->gbc, buf, buf_size * 8);
/* parse the syncinfo */
+ data_size_orig = *data_size;
*data_size = 0;
err = parse_frame_header(s);
channel_map = ff_ac3_dec_channel_map[s->output_mode & ~AC3_OUTPUT_LFEON][s->lfe_on];
for (ch = 0; ch < s->out_channels; ch++)
output[ch] = s->output[channel_map[ch]];
+ data_size_tmp = s->num_blocks * 256 * avctx->channels;
+ data_size_tmp *= avctx->sample_fmt == AV_SAMPLE_FMT_FLT ? sizeof(*out_samples_flt) : sizeof(*out_samples_s16);
+ if (data_size_orig < data_size_tmp)
+ return -1;
+ *data_size = data_size_tmp;
for (blk = 0; blk < s->num_blocks; blk++) {
if (!err && decode_audio_block(s, blk)) {
av_log(avctx, AV_LOG_ERROR, "error decoding the audio block\n");
err = 1;
}
+
if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
- float_interleave_noscale(out_samples_flt, output, 256, s->out_channels);
+ 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, output, 256, s->out_channels);
- out_samples += 256 * s->out_channels;
+ 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;
- *data_size *= avctx->sample_fmt == AV_SAMPLE_FMT_FLT ? sizeof(*out_samples_flt) : sizeof(*out_samples);
+ *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