/*
- * Atrac 1 compatible decoder
+ * ATRAC1 compatible decoder
* Copyright (c) 2009 Maxim Poliakovski
* Copyright (c) 2009 Benjamin Larsson
*
/**
* @file
- * Atrac 1 compatible decoder.
+ * ATRAC1 compatible decoder.
* This decoder handles raw ATRAC1 data and probably SDDS data.
*/
#include <stddef.h>
#include <stdio.h>
+#include "libavutil/float_dsp.h"
#include "avcodec.h"
#include "get_bits.h"
-#include "dsputil.h"
#include "fft.h"
-#include "fmtconvert.h"
+#include "internal.h"
#include "sinewin.h"
#include "atrac.h"
/**
* Sound unit struct, one unit is used per channel
*/
-typedef struct {
+typedef struct AT1SUCtx {
int log2_block_count[AT1_QMF_BANDS]; ///< log2 number of blocks in a band
int num_bfus; ///< number of Block Floating Units
float* spectrum[2];
/**
* The atrac1 context, holds all needed parameters for decoding
*/
-typedef struct {
- AVFrame frame;
+typedef struct AT1Ctx {
AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit
DECLARE_ALIGNED(32, float, spec)[AT1_SU_SAMPLES]; ///< the mdct spectrum buffer
DECLARE_ALIGNED(32, float, mid)[256];
DECLARE_ALIGNED(32, float, high)[512];
float* bands[3];
- float *out_samples[AT1_MAX_CHANNELS];
FFTContext mdct_ctx[3];
- int channels;
- DSPContext dsp;
- FmtConvertContext fmt_conv;
+ AVFloatDSPContext fdsp;
} AT1Ctx;
/** size of the transform in samples in the long mode for each QMF band */
at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos + start_pos], nbits, band_num);
/* overlap and window */
- q->dsp.vector_fmul_window(&q->bands[band_num][start_pos], prev_buf,
- &su->spectrum[0][ref_pos + start_pos], ff_sine_32, 16);
+ q->fdsp.vector_fmul_window(&q->bands[band_num][start_pos], prev_buf,
+ &su->spectrum[0][ref_pos + start_pos], ff_sine_32, 16);
prev_buf = &su->spectrum[0][ref_pos+start_pos + 16];
start_pos += block_size;
float iqmf_temp[512 + 46];
/* combine low and middle bands */
- atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp);
+ ff_atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp);
/* delay the signal of the high band by 23 samples */
memcpy( su->last_qmf_delay, &su->last_qmf_delay[256], sizeof(float) * 23);
memcpy(&su->last_qmf_delay[23], q->bands[2], sizeof(float) * 256);
/* combine (low + middle) and high bands */
- atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp);
+ ff_atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp);
}
static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
+ AVFrame *frame = data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
AT1Ctx *q = avctx->priv_data;
int ch, ret;
GetBitContext gb;
- float *samples;
- if (buf_size < 212 * q->channels) {
+ if (buf_size < 212 * avctx->channels) {
av_log(avctx, AV_LOG_ERROR, "Not enough data to decode!\n");
return AVERROR_INVALIDDATA;
}
/* get output buffer */
- q->frame.nb_samples = AT1_SU_SAMPLES;
- if ((ret = avctx->get_buffer(avctx, &q->frame)) < 0) {
+ frame->nb_samples = AT1_SU_SAMPLES;
+ if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
- samples = (float *)q->frame.data[0];
- for (ch = 0; ch < q->channels; ch++) {
+ for (ch = 0; ch < avctx->channels; ch++) {
AT1SUCtx* su = &q->SUs[ch];
init_get_bits(&gb, &buf[212 * ch], 212 * 8);
ret = at1_imdct_block(su, q);
if (ret < 0)
return ret;
- at1_subband_synthesis(q, su, q->channels == 1 ? samples : q->out_samples[ch]);
+ at1_subband_synthesis(q, su, (float *)frame->extended_data[ch]);
}
- /* interleave */
- if (q->channels == 2) {
- q->fmt_conv.float_interleave(samples, (const float **)q->out_samples,
- AT1_SU_SAMPLES, 2);
- }
-
- *got_frame_ptr = 1;
- *(AVFrame *)data = q->frame;
+ *got_frame_ptr = 1;
return avctx->block_align;
}
{
AT1Ctx *q = avctx->priv_data;
- av_freep(&q->out_samples[0]);
-
ff_mdct_end(&q->mdct_ctx[0]);
ff_mdct_end(&q->mdct_ctx[1]);
ff_mdct_end(&q->mdct_ctx[2]);
AT1Ctx *q = avctx->priv_data;
int ret;
- avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
if (avctx->channels < 1 || avctx->channels > AT1_MAX_CHANNELS) {
av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n",
avctx->channels);
return AVERROR(EINVAL);
}
- q->channels = avctx->channels;
-
- if (avctx->channels == 2) {
- q->out_samples[0] = av_malloc(2 * AT1_SU_SAMPLES * sizeof(*q->out_samples[0]));
- q->out_samples[1] = q->out_samples[0] + AT1_SU_SAMPLES;
- if (!q->out_samples[0]) {
- av_freep(&q->out_samples[0]);
- return AVERROR(ENOMEM);
- }
- }
/* Init the mdct transforms */
if ((ret = ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15))) ||
ff_init_ff_sine_windows(5);
- atrac_generate_tables();
+ ff_atrac_generate_tables();
- dsputil_init(&q->dsp, avctx);
- ff_fmt_convert_init(&q->fmt_conv, avctx);
+ avpriv_float_dsp_init(&q->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
q->bands[0] = q->low;
q->bands[1] = q->mid;
q->SUs[1].spectrum[0] = q->SUs[1].spec1;
q->SUs[1].spectrum[1] = q->SUs[1].spec2;
- avcodec_get_frame_defaults(&q->frame);
- avctx->coded_frame = &q->frame;
-
return 0;
}
AVCodec ff_atrac1_decoder = {
- .name = "atrac1",
- .type = AVMEDIA_TYPE_AUDIO,
- .id = CODEC_ID_ATRAC1,
+ .name = "atrac1",
+ .long_name = NULL_IF_CONFIG_SMALL("ATRAC1 (Adaptive TRansform Acoustic Coding)"),
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_CODEC_ID_ATRAC1,
.priv_data_size = sizeof(AT1Ctx),
- .init = atrac1_decode_init,
- .close = atrac1_decode_end,
- .decode = atrac1_decode_frame,
- .capabilities = CODEC_CAP_DR1,
- .long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),
+ .init = atrac1_decode_init,
+ .close = atrac1_decode_end,
+ .decode = atrac1_decode_frame,
+ .capabilities = CODEC_CAP_DR1,
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_NONE },
};
projects / ffmpeg / blobdiff