static av_cold int ac3_decode_init(AVCodecContext *avctx)
{
AC3DecodeContext *s = avctx->priv_data;
+ int i;
+
s->avctx = avctx;
ff_ac3_common_init();
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);
+ avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
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 */
- 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;
- }
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
/* allow downmixing to stereo or mono */
if (avctx->channels > 0 && avctx->request_channels > 0 &&
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
+ for (i = 0; i < AC3_MAX_CHANNELS; i++) {
+ s->xcfptr[i] = s->transform_coeffs[i];
+ s->dlyptr[i] = s->delay[i];
+ }
+
return 0;
}
int mantissa;
switch (bap) {
case 0:
+ /* random noise with approximate range of -0.707 to 0.707 */
if (dither)
- mantissa = (av_lfg_get(&s->dith_state) & 0x7FFFFF) - 0x400000;
+ mantissa = (av_lfg_get(&s->dith_state) / 362) - 5932275;
else
mantissa = 0;
break;
for (ch = 1; ch <= s->channels; ch++) {
/* transform coefficients for full-bandwidth channel */
decode_transform_coeffs_ch(s, blk, ch, &m);
- /* tranform coefficients for coupling channel come right after the
+ /* transform coefficients for coupling channel come right after the
coefficients for the first coupled channel*/
if (s->channel_in_cpl[ch]) {
if (!got_cplchan) {
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->dsp.vector_fmul_window(s->output[ch - 1], s->delay[ch - 1],
- s->tmp_output, s->window, 128);
+ s->fdsp.vector_fmul_window(s->outptr[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];
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->dsp.vector_fmul_window(s->output[ch - 1], s->delay[ch - 1],
- s->tmp_output, s->window, 128);
+ 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));
}
}
if (s->eac3 && get_bits1(gbc)) {
/* TODO: parse enhanced coupling strategy info */
av_log_missing_feature(s->avctx, "Enhanced coupling", 1);
- return -1;
+ return AVERROR_PATCHWELCOME;
}
/* determine which channels are coupled */
/* apply scaling to coefficients (headroom, dynrng) */
for (ch = 1; ch <= s->channels; ch++) {
- float gain = s->mul_bias / 4194304.0f;
+ float gain = 1.0 / 4194304.0f;
if (s->channel_mode == AC3_CHMODE_DUALMONO) {
gain *= s->dynamic_range[2 - ch];
} else {
do_imdct(s, s->channels);
if (downmix_output) {
- s->ac3dsp.downmix(s->output, s->downmix_coeffs,
+ s->ac3dsp.downmix(s->outptr, s->downmix_coeffs,
s->out_channels, s->fbw_channels, 256);
}
} else {
if (downmix_output) {
- s->ac3dsp.downmix(s->transform_coeffs + 1, s->downmix_coeffs,
+ s->ac3dsp.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->delay, s->downmix_coeffs, s->out_channels,
+ s->ac3dsp.downmix(s->dlyptr, s->downmix_coeffs, s->out_channels,
s->fbw_channels, 128);
}
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
AC3DecodeContext *s = avctx->priv_data;
- float *out_samples_flt;
- int16_t *out_samples_s16;
int blk, ch, err, ret;
const uint8_t *channel_map;
const float *output[AC3_MAX_CHANNELS];
avctx->audio_service_type = AV_AUDIO_SERVICE_TYPE_KARAOKE;
/* get output buffer */
+ avctx->channels = s->out_channels;
s->frame.nb_samples = s->num_blocks * 256;
- if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
+ if ((ret = ff_get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
- out_samples_flt = (float *)s->frame.data[0];
- out_samples_s16 = (int16_t *)s->frame.data[0];
/* decode the audio blocks */
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]];
+ for (ch = 0; ch < s->channels; ch++) {
+ if (ch < s->out_channels)
+ s->outptr[channel_map[ch]] = (float *)s->frame.data[ch];
+ else
+ s->outptr[ch] = s->output[ch];
+ output[ch] = s->output[ch];
+ }
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) {
- 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;
+ if (err)
+ for (ch = 0; ch < s->out_channels; ch++)
+ memcpy(s->outptr[channel_map[ch]], output[ch], 1024);
+ for (ch = 0; ch < s->out_channels; ch++) {
+ output[ch] = s->outptr[channel_map[ch]];
+ s->outptr[channel_map[ch]] += AC3_BLOCK_SIZE;
}
}
+ /* keep last block for error concealment in next frame */
+ for (ch = 0; ch < s->out_channels; ch++)
+ memcpy(s->output[ch], output[ch], 1024);
+
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
.decode = ac3_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
- .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,
- AV_SAMPLE_FMT_S16,
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
.priv_class = &ac3_decoder_class,
};
.decode = ac3_decode_frame,
.capabilities = CODEC_CAP_DR1,
.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,
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
.priv_class = &eac3_decoder_class,
};
projects / ffmpeg / blobdiff