/*
- * 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 "internal.h"
#include "sinewin.h"
#include "atrac.h"
DECLARE_ALIGNED(32, float, mid)[256];
DECLARE_ALIGNED(32, float, high)[512];
float* bands[3];
- DECLARE_ALIGNED(32, float, out_samples)[AT1_MAX_CHANNELS][AT1_SU_SAMPLES];
FFTContext mdct_ctx[3];
- int channels;
- DSPContext dsp;
+ AVFloatDSPContext fdsp;
} AT1Ctx;
/** size of the transform in samples in the long mode for each QMF band */
nbits = mdct_long_nbits[band_num] - log2_block_count;
if (nbits != 5 && nbits != 7 && nbits != 8)
- return -1;
+ return AVERROR_INVALIDDATA;
} else {
block_size = 32;
nbits = 5;
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;
/* low and mid band */
log2_block_count_tmp = get_bits(gb, 2);
if (log2_block_count_tmp & 1)
- return -1;
+ return AVERROR_INVALIDDATA;
log2_block_cnt[i] = 2 - log2_block_count_tmp;
}
/* high band */
log2_block_count_tmp = get_bits(gb, 2);
if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3)
- return -1;
+ return AVERROR_INVALIDDATA;
log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp;
skip_bits(gb, 2);
/* check for bitstream overflow */
if (bits_used > AT1_SU_MAX_BITS)
- return -1;
+ return AVERROR_INVALIDDATA;
/* get the position of the 1st spec according to the block size mode */
pos = su->log2_block_count[band_num] ? bfu_start_short[bfu_num] : bfu_start_long[bfu_num];
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 *data_size, AVPacket *avpkt)
+ 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, i, out_size;
+ int ch, ret;
GetBitContext gb;
- float* samples = data;
- if (buf_size < 212 * q->channels) {
- av_log(q,AV_LOG_ERROR,"Not enought data to decode!\n");
- return -1;
+ if (buf_size < 212 * avctx->channels) {
+ av_log(avctx, AV_LOG_ERROR, "Not enough data to decode!\n");
+ return AVERROR_INVALIDDATA;
}
- out_size = q->channels * AT1_SU_SAMPLES *
- 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 */
+ 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;
}
- 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->out_samples[ch]);
+ at1_subband_synthesis(q, su, (float *)frame->extended_data[ch]);
}
- /* interleave; FIXME, should create/use a DSP function */
- if (q->channels == 1) {
- /* mono */
- memcpy(samples, q->out_samples[0], AT1_SU_SAMPLES * 4);
- } else {
- /* stereo */
- for (i = 0; i < AT1_SU_SAMPLES; i++) {
- samples[i * 2] = q->out_samples[0][i];
- samples[i * 2 + 1] = q->out_samples[1][i];
- }
- }
+ *got_frame_ptr = 1;
- *data_size = out_size;
return avctx->block_align;
}
+static av_cold int atrac1_decode_end(AVCodecContext * avctx)
+{
+ AT1Ctx *q = avctx->priv_data;
+
+ ff_mdct_end(&q->mdct_ctx[0]);
+ ff_mdct_end(&q->mdct_ctx[1]);
+ ff_mdct_end(&q->mdct_ctx[2]);
+
+ return 0;
+}
+
+
static av_cold int atrac1_decode_init(AVCodecContext *avctx)
{
AT1Ctx *q = avctx->priv_data;
+ int ret;
- avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
- q->channels = avctx->channels;
+ 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);
+ }
/* Init the mdct transforms */
- ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15));
- ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15));
- ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15));
+ if ((ret = ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15))) ||
+ (ret = ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15))) ||
+ (ret = ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15)))) {
+ av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
+ atrac1_decode_end(avctx);
+ return ret;
+ }
ff_init_ff_sine_windows(5);
- atrac_generate_tables();
+ ff_atrac_generate_tables();
- dsputil_init(&q->dsp, avctx);
+ avpriv_float_dsp_init(&q->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
q->bands[0] = q->low;
q->bands[1] = q->mid;
}
-static av_cold int atrac1_decode_end(AVCodecContext * avctx) {
- AT1Ctx *q = avctx->priv_data;
-
- ff_mdct_end(&q->mdct_ctx[0]);
- ff_mdct_end(&q->mdct_ctx[1]);
- ff_mdct_end(&q->mdct_ctx[2]);
- 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,
- .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