#include <stddef.h>
#include <stdio.h>
+#include "libavutil/float_dsp.h"
#include "avcodec.h"
#include "get_bits.h"
-#include "dsputil.h"
#include "bytestream.h"
#include "fft.h"
#include "fmtconvert.h"
} channel_unit;
typedef struct {
+ AVFrame frame;
GetBitContext gb;
//@{
/** stream data */
FFTContext mdct_ctx;
FmtConvertContext fmt_conv;
+ AVFloatDSPContext fdsp;
} ATRAC3Context;
static DECLARE_ALIGNED(32, float, mdct_window)[MDCT_SIZE];
static VLC spectral_coeff_tab[7];
static float gain_tab1[16];
static float gain_tab2[31];
-static DSPContext dsp;
/**
q->mdct_ctx.imdct_calc(&q->mdct_ctx,pOutput,pInput);
/* Perform windowing on the output. */
- dsp.vector_fmul(pOutput, pOutput, mdct_window, MDCT_SIZE);
+ q->fdsp.vector_fmul(pOutput, pOutput, mdct_window, MDCT_SIZE);
}
for (k=0; k<coded_components; k++) {
sfIndx = get_bits(gb,6);
+ if (component_count >= 64)
+ return AVERROR_INVALIDDATA;
pComponent[component_count].pos = j * 64 + (get_bits(gb,6));
max_coded_values = SAMPLES_PER_FRAME - pComponent[component_count].pos;
coded_values = coded_values_per_component + 1;
memset(pSnd->IMDCT_buf, 0, 512 * sizeof(float));
/* gain compensation and overlapping */
- gainCompensateAndOverlap (pSnd->IMDCT_buf, &(pSnd->prevFrame[band*256]), &(pOut[band*256]),
- &((pSnd->gainBlock[1 - (pSnd->gcBlkSwitch)]).gBlock[band]),
- &((pSnd->gainBlock[pSnd->gcBlkSwitch]).gBlock[band]));
+ gainCompensateAndOverlap(pSnd->IMDCT_buf, &pSnd->prevFrame[band * 256],
+ &pOut[band * 256],
+ &pSnd->gainBlock[1 - pSnd->gcBlkSwitch].gBlock[band],
+ &pSnd->gainBlock[ pSnd->gcBlkSwitch].gBlock[band]);
}
/* Swap the gain control buffers for the next frame. */
result = decodeChannelSoundUnit(q,&q->gb, q->pUnits, out_samples[0], 0, JOINT_STEREO);
if (result != 0)
- return (result);
+ return result;
/* Framedata of the su2 in the joint-stereo mode is encoded in
* reverse byte order so we need to swap it first. */
/* Decode Sound Unit 2. */
result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[1], out_samples[1], 1, JOINT_STEREO);
if (result != 0)
- return (result);
+ return result;
/* Reconstruct the channel coefficients. */
reverseMatrixing(out_samples[0], out_samples[1], q->matrix_coeff_index_prev, q->matrix_coeff_index_now);
for (i=0 ; i<q->channels ; i++) {
/* Set the bitstream reader at the start of a channel sound unit. */
- init_get_bits(&q->gb, databuf+((i*q->bytes_per_frame)/q->channels), (q->bits_per_frame)/q->channels);
+ init_get_bits(&q->gb,
+ databuf + i * q->bytes_per_frame / q->channels,
+ q->bits_per_frame / q->channels);
result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[i], out_samples[i], i, q->codingMode);
if (result != 0)
- return (result);
+ return result;
}
}
p2= p1+256;
p3= p2+256;
p4= p3+256;
- atrac_iqmf (p1, p2, 256, p1, q->pUnits[i].delayBuf1, q->tempBuf);
- atrac_iqmf (p4, p3, 256, p3, q->pUnits[i].delayBuf2, q->tempBuf);
- atrac_iqmf (p1, p3, 512, p1, q->pUnits[i].delayBuf3, q->tempBuf);
+ ff_atrac_iqmf (p1, p2, 256, p1, q->pUnits[i].delayBuf1, q->tempBuf);
+ ff_atrac_iqmf (p4, p3, 256, p3, q->pUnits[i].delayBuf2, q->tempBuf);
+ ff_atrac_iqmf (p1, p3, 512, p1, q->pUnits[i].delayBuf3, q->tempBuf);
}
return 0;
* @param avctx pointer to the AVCodecContext
*/
-static int atrac3_decode_frame(AVCodecContext *avctx,
- void *data, int *data_size,
- AVPacket *avpkt) {
+static int atrac3_decode_frame(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, AVPacket *avpkt)
+{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
ATRAC3Context *q = avctx->priv_data;
- int result = 0, out_size;
+ int result;
const uint8_t* databuf;
- float *samples_flt = data;
- int16_t *samples_s16 = data;
+ float *samples_flt;
+ int16_t *samples_s16;
if (buf_size < avctx->block_align) {
av_log(avctx, AV_LOG_ERROR,
return AVERROR_INVALIDDATA;
}
- out_size = SAMPLES_PER_FRAME * q->channels *
- av_get_bytes_per_sample(avctx->sample_fmt);
- if (*data_size < out_size) {
- av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
- return AVERROR(EINVAL);
+ /* get output buffer */
+ q->frame.nb_samples = SAMPLES_PER_FRAME;
+ if ((result = avctx->get_buffer(avctx, &q->frame)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ return result;
}
+ samples_flt = (float *)q->frame.data[0];
+ samples_s16 = (int16_t *)q->frame.data[0];
/* Check if we need to descramble and what buffer to pass on. */
if (q->scrambled_stream) {
(const float **)q->outSamples,
SAMPLES_PER_FRAME, q->channels);
}
- *data_size = out_size;
+
+ *got_frame_ptr = 1;
+ *(AVFrame *)data = q->frame;
return avctx->block_align;
}
return ret;
}
- atrac_generate_tables();
+ ff_atrac_generate_tables();
/* Generate gain tables. */
for (i=0 ; i<16 ; i++)
q->matrix_coeff_index_next[i] = 3;
}
- dsputil_init(&dsp, avctx);
+ avpriv_float_dsp_init(&q->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
ff_fmt_convert_init(&q->fmt_conv, avctx);
q->pUnits = av_mallocz(sizeof(channel_unit)*q->channels);
}
}
+ avcodec_get_frame_defaults(&q->frame);
+ avctx->coded_frame = &q->frame;
+
return 0;
}
AVCodec ff_atrac3_decoder =
{
- .name = "atrac3",
- .type = AVMEDIA_TYPE_AUDIO,
- .id = CODEC_ID_ATRAC3,
+ .name = "atrac3",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_CODEC_ID_ATRAC3,
.priv_data_size = sizeof(ATRAC3Context),
- .init = atrac3_decode_init,
- .close = atrac3_decode_close,
- .decode = atrac3_decode_frame,
- .capabilities = CODEC_CAP_SUBFRAMES,
- .long_name = NULL_IF_CONFIG_SMALL("Atrac 3 (Adaptive TRansform Acoustic Coding 3)"),
+ .init = atrac3_decode_init,
+ .close = atrac3_decode_close,
+ .decode = atrac3_decode_frame,
+ .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
+ .long_name = NULL_IF_CONFIG_SMALL("Atrac 3 (Adaptive TRansform Acoustic Coding 3)"),
};
projects / ffmpeg / blobdiff