/*
- * 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 "bitstream.h"
#include "fft.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 {
+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];
- 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;
* Parse the block size mode byte
*/
-static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS])
+static int at1_parse_bsm(BitstreamContext *bc,
+ int log2_block_cnt[AT1_QMF_BANDS])
{
int log2_block_count_tmp, i;
for (i = 0; i < 2; i++) {
/* low and mid band */
- log2_block_count_tmp = get_bits(gb, 2);
+ log2_block_count_tmp = bitstream_read(bc, 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);
+ log2_block_count_tmp = bitstream_read(bc, 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);
+ bitstream_skip(bc, 2);
return 0;
}
-static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su,
+static int at1_unpack_dequant(BitstreamContext *bc, AT1SUCtx *su,
float spec[AT1_SU_SAMPLES])
{
int bits_used, band_num, bfu_num, i;
uint8_t idsfs[AT1_MAX_BFU]; ///< the scalefactor indexes for each BFU
/* parse the info byte (2nd byte) telling how much BFUs were coded */
- su->num_bfus = bfu_amount_tab1[get_bits(gb, 3)];
+ su->num_bfus = bfu_amount_tab1[bitstream_read(bc, 3)];
/* calc number of consumed bits:
num_BFUs * (idwl(4bits) + idsf(6bits)) + log2_block_count(8bits) + info_byte(8bits)
+ info_byte_copy(8bits) + log2_block_count_copy(8bits) */
bits_used = su->num_bfus * 10 + 32 +
- bfu_amount_tab2[get_bits(gb, 2)] +
- (bfu_amount_tab3[get_bits(gb, 3)] << 1);
+ bfu_amount_tab2[bitstream_read(bc, 2)] +
+ (bfu_amount_tab3[bitstream_read(bc, 3)] << 1);
/* get word length index (idwl) for each BFU */
for (i = 0; i < su->num_bfus; i++)
- idwls[i] = get_bits(gb, 4);
+ idwls[i] = bitstream_read(bc, 4);
/* get scalefactor index (idsf) for each BFU */
for (i = 0; i < su->num_bfus; i++)
- idsfs[i] = get_bits(gb, 6);
+ idsfs[i] = bitstream_read(bc, 6);
/* zero idwl/idsf for empty BFUs */
for (i = su->num_bfus; i < AT1_MAX_BFU; i++)
/* 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];
/* read in a quantized spec and convert it to
* signed int and then inverse quantization
*/
- spec[pos+i] = get_sbits(gb, word_len) * scale_factor * max_quant;
+ spec[pos+i] = bitstream_read_signed(bc, word_len) * scale_factor * max_quant;
}
} else { /* word_len = 0 -> empty BFU, zero all specs in the emty BFU */
memset(&spec[pos], 0, num_specs * sizeof(float));
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;
- GetBitContext gb;
- float* samples = data;
+ int ch, ret;
+ BitstreamContext bc;
+
+ if (buf_size < 212 * avctx->channels) {
+ av_log(avctx, AV_LOG_ERROR, "Not enough data to decode!\n");
+ return AVERROR_INVALIDDATA;
+ }
- if (buf_size < 212 * q->channels) {
- av_log(q,AV_LOG_ERROR,"Not enought data to decode!\n");
- return -1;
+ /* 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);
+ bitstream_init8(&bc, &buf[212 * ch], 212);
/* parse block_size_mode, 1st byte */
- ret = at1_parse_bsm(&gb, su->log2_block_count);
+ ret = at1_parse_bsm(&bc, su->log2_block_count);
if (ret < 0)
return ret;
- ret = at1_unpack_dequant(&gb, su, q->spec);
+ ret = at1_unpack_dequant(&bc, su, q->spec);
if (ret < 0)
return ret;
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 = q->channels * AT1_SU_SAMPLES * sizeof(*samples);
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 & AV_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 = AV_CODEC_CAP_DR1,
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_NONE },
};
projects / ffmpeg / blobdiff