/**
* Init IMDCT and windowing tables
*/
static av_cold int init_mdct_win(TwinVQContext *tctx)
{
int i, j, ret;
const TwinVQModeTab *mtab = tctx->mtab;
int size_s = mtab->size / mtab->fmode[TWINVQ_FT_SHORT].sub;
int size_m = mtab->size / mtab->fmode[TWINVQ_FT_MEDIUM].sub;
int channels = tctx->avctx->channels;
float norm = channels == 1 ? 2.0 : 1.0;
for (i = 0; i < 3; i++) {
int bsize = tctx->mtab->size / tctx->mtab->fmode[i].sub;
if ((ret = ff_mdct_init(&tctx->mdct_ctx[i], av_log2(bsize) + 1, 1,
-sqrt(norm / bsize) / (1 << 15))))
return ret;
}
FF_ALLOC_OR_GOTO(tctx->avctx, tctx->tmp_buf,
mtab->size * sizeof(*tctx->tmp_buf), alloc_fail);
FF_ALLOC_OR_GOTO(tctx->avctx, tctx->spectrum,
2 * mtab->size * channels * sizeof(*tctx->spectrum),
alloc_fail);
FF_ALLOC_OR_GOTO(tctx->avctx, tctx->curr_frame,
2 * mtab->size * channels * sizeof(*tctx->curr_frame),
alloc_fail);
FF_ALLOC_OR_GOTO(tctx->avctx, tctx->prev_frame,
2 * mtab->size * channels * sizeof(*tctx->prev_frame),
alloc_fail);
for (i = 0; i < 3; i++) {
int m = 4 * mtab->size / mtab->fmode[i].sub;
double freq = 2 * M_PI / m;
FF_ALLOC_OR_GOTO(tctx->avctx, tctx->cos_tabs[i],
(m / 4) * sizeof(*tctx->cos_tabs[i]), alloc_fail);
for (j = 0; j <= m / 8; j++)
tctx->cos_tabs[i][j] = cos((2 * j + 1) * freq);
for (j = 1; j < m / 8; j++)
tctx->cos_tabs[i][m / 4 - j] = tctx->cos_tabs[i][j];
}
ff_init_ff_sine_windows(av_log2(size_m));
ff_init_ff_sine_windows(av_log2(size_s / 2));
ff_init_ff_sine_windows(av_log2(mtab->size));
return 0;
alloc_fail:
return AVERROR(ENOMEM);
}
int AC3_NAME(allocate_sample_buffers)(AC3EncodeContext *s)
{
int ch;
FF_ALLOC_OR_GOTO(s->avctx, s->windowed_samples, AC3_WINDOW_SIZE *
sizeof(*s->windowed_samples), alloc_fail);
FF_ALLOC_OR_GOTO(s->avctx, s->planar_samples, s->channels * sizeof(*s->planar_samples),
alloc_fail);
for (ch = 0; ch < s->channels; ch++) {
FF_ALLOCZ_OR_GOTO(s->avctx, s->planar_samples[ch],
(AC3_FRAME_SIZE+AC3_BLOCK_SIZE) * sizeof(**s->planar_samples),
alloc_fail);
}
return 0;
alloc_fail:
return AVERROR(ENOMEM);
}
static av_cold int adpcm_encode_init(AVCodecContext *avctx)
{
ADPCMEncodeContext *s = avctx->priv_data;
uint8_t *extradata;
int i;
int ret = AVERROR(ENOMEM);
if (avctx->channels > 2) {
av_log(avctx, AV_LOG_ERROR, "only stereo or mono is supported\n");
return AVERROR(EINVAL);
}
if (avctx->trellis && (unsigned)avctx->trellis > 16U) {
av_log(avctx, AV_LOG_ERROR, "invalid trellis size\n");
return AVERROR(EINVAL);
}
if (avctx->trellis) {
int frontier = 1 << avctx->trellis;
int max_paths = frontier * FREEZE_INTERVAL;
FF_ALLOC_OR_GOTO(avctx, s->paths,
max_paths * sizeof(*s->paths), error);
FF_ALLOC_OR_GOTO(avctx, s->node_buf,
2 * frontier * sizeof(*s->node_buf), error);
FF_ALLOC_OR_GOTO(avctx, s->nodep_buf,
2 * frontier * sizeof(*s->nodep_buf), error);
FF_ALLOC_OR_GOTO(avctx, s->trellis_hash,
65536 * sizeof(*s->trellis_hash), error);
}
avctx->bits_per_coded_sample = av_get_bits_per_sample(avctx->codec->id);
switch (avctx->codec->id) {
case AV_CODEC_ID_ADPCM_IMA_WAV:
/* each 16 bits sample gives one nibble
and we have 4 bytes per channel overhead */
avctx->frame_size = (BLKSIZE - 4 * avctx->channels) * 8 /
(4 * avctx->channels) + 1;
/* seems frame_size isn't taken into account...
have to buffer the samples :-( */
avctx->block_align = BLKSIZE;
break;
case AV_CODEC_ID_ADPCM_IMA_QT:
avctx->frame_size = 64;
avctx->block_align = 34 * avctx->channels;
break;
case AV_CODEC_ID_ADPCM_MS:
/* each 16 bits sample gives one nibble
and we have 7 bytes per channel overhead */
avctx->frame_size = (BLKSIZE - 7 * avctx->channels) * 2 /
avctx->channels + 2;
avctx->block_align = BLKSIZE;
if (!(avctx->extradata = av_malloc(32 + FF_INPUT_BUFFER_PADDING_SIZE)))
goto error;
avctx->extradata_size = 32;
extradata = avctx->extradata;
bytestream_put_le16(&extradata, avctx->frame_size);
bytestream_put_le16(&extradata, 7); /* wNumCoef */
for (i = 0; i < 7; i++) {
bytestream_put_le16(&extradata, ff_adpcm_AdaptCoeff1[i] * 4);
bytestream_put_le16(&extradata, ff_adpcm_AdaptCoeff2[i] * 4);
}
break;
case AV_CODEC_ID_ADPCM_YAMAHA:
avctx->frame_size = BLKSIZE * 2 / avctx->channels;
avctx->block_align = BLKSIZE;
break;
case AV_CODEC_ID_ADPCM_SWF:
if (avctx->sample_rate != 11025 &&
avctx->sample_rate != 22050 &&
avctx->sample_rate != 44100) {
av_log(avctx, AV_LOG_ERROR, "Sample rate must be 11025, "
"22050 or 44100\n");
ret = AVERROR(EINVAL);
goto error;
}
avctx->frame_size = 512 * (avctx->sample_rate / 11025);
break;
default:
ret = AVERROR(EINVAL);
goto error;
}
#if FF_API_OLD_ENCODE_AUDIO
if (!(avctx->coded_frame = avcodec_alloc_frame()))
goto error;
#endif
return 0;
error:
av_freep(&s->paths);
av_freep(&s->node_buf);
av_freep(&s->nodep_buf);
av_freep(&s->trellis_hash);
return ret;
}
