* @author Paul B Mahol
*/
+#include "libavutil/internal.h"
#include "libavutil/samplefmt.h"
#include "tak.h"
+#include "audiodsp.h"
#include "avcodec.h"
-#include "dsputil.h"
#include "internal.h"
#include "unary.h"
typedef struct TAKDecContext {
AVCodecContext *avctx; // parent AVCodecContext
- AVFrame frame; // AVFrame for decoded output
- DSPContext dsp;
+ AudioDSPContext adsp;
TAKStreamInfo ti;
GetBitContext gb; // bitstream reader initialized to start at the current frame
{
TAKDecContext *s = avctx->priv_data;
- ff_dsputil_init(&s->dsp, avctx);
+ ff_audiodsp_init(&s->adsp);
s->avctx = avctx;
- avcodec_get_frame_defaults(&s->frame);
- avctx->coded_frame = &s->frame;
set_sample_rate_params(avctx);
for (i = 0; i < subframe_size - filter_order; i++) {
int v = 1 << (filter_quant - 1);
- v += s->dsp.scalarproduct_int16(&s->residues[i], filter,
- FFALIGN(filter_order, 16));
+ v += s->adsp.scalarproduct_int16(&s->residues[i], filter,
+ FFALIGN(filter_order, 16));
v = (av_clip(v >> filter_quant, -8192, 8191) << dshift) - *decoded;
*decoded++ = v;
for (i = 0; i < length2; i++) {
int v = 1 << 9;
- v += s->dsp.scalarproduct_int16(&s->residues[i], filter,
- FFALIGN(filter_order, 16));
+ v += s->adsp.scalarproduct_int16(&s->residues[i], filter,
+ FFALIGN(filter_order, 16));
p1[i] = (av_clip(v >> 10, -8192, 8191) << dshift) - p1[i];
}
int *got_frame_ptr, AVPacket *pkt)
{
TAKDecContext *s = avctx->priv_data;
+ AVFrame *frame = data;
GetBitContext *gb = &s->gb;
int chan, i, ret, hsize;
return ret;
if (s->ti.flags & TAK_FRAME_FLAG_HAS_METADATA) {
- av_log_missing_feature(avctx, "frame metadata", 1);
+ avpriv_request_sample(avctx, "Frame metadata");
return AVERROR_PATCHWELCOME;
}
if (avctx->err_recognition & AV_EF_CRCCHECK) {
if (ff_tak_check_crc(pkt->data, hsize)) {
av_log(avctx, AV_LOG_ERROR, "CRC error\n");
- return AVERROR_INVALIDDATA;
+ if (avctx->err_recognition & AV_EF_EXPLODE)
+ return AVERROR_INVALIDDATA;
}
}
s->nb_samples = s->ti.last_frame_samples ? s->ti.last_frame_samples
: s->ti.frame_samples;
- s->frame.nb_samples = s->nb_samples;
- if ((ret = ff_get_buffer(avctx, &s->frame)) < 0)
+ frame->nb_samples = s->nb_samples;
+ if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
if (avctx->bits_per_coded_sample <= 16) {
return ret;
} else {
for (chan = 0; chan < avctx->channels; chan++)
- s->decoded[chan] = (int32_t *)s->frame.extended_data[chan];
+ s->decoded[chan] = (int32_t *)frame->extended_data[chan];
}
if (s->nb_samples < 16) {
if (ff_tak_check_crc(pkt->data + hsize,
get_bits_count(gb) / 8 - hsize)) {
av_log(avctx, AV_LOG_ERROR, "CRC error\n");
- return AVERROR_INVALIDDATA;
+ if (avctx->err_recognition & AV_EF_EXPLODE)
+ return AVERROR_INVALIDDATA;
}
}
switch (avctx->sample_fmt) {
case AV_SAMPLE_FMT_U8P:
for (chan = 0; chan < avctx->channels; chan++) {
- uint8_t *samples = (uint8_t *)s->frame.extended_data[chan];
+ uint8_t *samples = (uint8_t *)frame->extended_data[chan];
int32_t *decoded = s->decoded[chan];
for (i = 0; i < s->nb_samples; i++)
samples[i] = decoded[i] + 0x80;
break;
case AV_SAMPLE_FMT_S16P:
for (chan = 0; chan < avctx->channels; chan++) {
- int16_t *samples = (int16_t *)s->frame.extended_data[chan];
+ int16_t *samples = (int16_t *)frame->extended_data[chan];
int32_t *decoded = s->decoded[chan];
for (i = 0; i < s->nb_samples; i++)
samples[i] = decoded[i];
break;
case AV_SAMPLE_FMT_S32P:
for (chan = 0; chan < avctx->channels; chan++) {
- int32_t *samples = (int32_t *)s->frame.extended_data[chan];
+ int32_t *samples = (int32_t *)frame->extended_data[chan];
for (i = 0; i < s->nb_samples; i++)
samples[i] <<= 8;
}
break;
}
- *got_frame_ptr = 1;
- *(AVFrame *)data = s->frame;
+ *got_frame_ptr = 1;
return pkt->size;
}
AVCodec ff_tak_decoder = {
.name = "tak",
+ .long_name = NULL_IF_CONFIG_SMALL("TAK (Tom's lossless Audio Kompressor)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_TAK,
.priv_data_size = sizeof(TAKDecContext),
.close = tak_decode_close,
.decode = tak_decode_frame,
.capabilities = CODEC_CAP_DR1,
- .long_name = NULL_IF_CONFIG_SMALL("TAK (Tom's lossless Audio Kompressor)"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_S32P,