* Copyright (c) 2009 Maxim Poliakovski
* Copyright (c) 2009 Benjamin Larsson
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
- * @file libavcodec/atrac1.c
+ * @file
* Atrac 1 compatible decoder.
* This decoder handles raw ATRAC1 data and probably SDDS data.
*/
#include "avcodec.h"
#include "get_bits.h"
#include "dsputil.h"
+#include "fft.h"
+#include "fmtconvert.h"
+#include "sinewin.h"
#include "atrac.h"
#include "atrac1data.h"
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];
- DECLARE_ALIGNED_16(float, spec1[AT1_SU_SAMPLES]); ///< mdct buffer
- DECLARE_ALIGNED_16(float, spec2[AT1_SU_SAMPLES]); ///< mdct buffer
- DECLARE_ALIGNED_16(float, fst_qmf_delay[46]); ///< delay line for the 1st stacked QMF filter
- DECLARE_ALIGNED_16(float, snd_qmf_delay[46]); ///< delay line for the 2nd stacked QMF filter
- DECLARE_ALIGNED_16(float, last_qmf_delay[256+23]); ///< delay line for the last stacked QMF filter
+ DECLARE_ALIGNED(32, float, spec1)[AT1_SU_SAMPLES]; ///< mdct buffer
+ DECLARE_ALIGNED(32, float, spec2)[AT1_SU_SAMPLES]; ///< mdct buffer
+ DECLARE_ALIGNED(32, float, fst_qmf_delay)[46]; ///< delay line for the 1st stacked QMF filter
+ DECLARE_ALIGNED(32, float, snd_qmf_delay)[46]; ///< delay line for the 2nd stacked QMF filter
+ DECLARE_ALIGNED(32, float, last_qmf_delay)[256+23]; ///< delay line for the last stacked QMF filter
} AT1SUCtx;
/**
* The atrac1 context, holds all needed parameters for decoding
*/
typedef struct {
+ AVFrame frame;
AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit
- DECLARE_ALIGNED_16(float, spec[AT1_SU_SAMPLES]); ///< the mdct spectrum buffer
+ DECLARE_ALIGNED(32, float, spec)[AT1_SU_SAMPLES]; ///< the mdct spectrum buffer
- DECLARE_ALIGNED_16(float, low[256]);
- DECLARE_ALIGNED_16(float, mid[256]);
- DECLARE_ALIGNED_16(float, high[512]);
+ DECLARE_ALIGNED(32, float, low)[256];
+ DECLARE_ALIGNED(32, float, mid)[256];
+ DECLARE_ALIGNED(32, float, high)[512];
float* bands[3];
- DECLARE_ALIGNED_16(float, out_samples[AT1_MAX_CHANNELS][AT1_SU_SAMPLES]);
+ float *out_samples[AT1_MAX_CHANNELS];
FFTContext mdct_ctx[3];
int channels;
DSPContext dsp;
+ FmtConvertContext fmt_conv;
} AT1Ctx;
/** size of the transform in samples in the long mode for each QMF band */
for (i = 0; i < transf_size / 2; i++)
FFSWAP(float, spec[i], spec[transf_size - 1 - i]);
}
- ff_imdct_half(mdct_context, out, spec);
+ mdct_context->imdct_half(mdct_context, out, spec);
}
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;
/* 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, 0, 16);
+ &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);
int num_specs = specs_per_bfu[bfu_num];
int word_len = !!idwls[bfu_num] + idwls[bfu_num];
- float scale_factor = sf_table[idsfs[bfu_num]];
+ float scale_factor = ff_atrac_sf_table[idsfs[bfu_num]];
bits_used += word_len * num_specs; /* add number of bits consumed by current BFU */
/* 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];
}
-void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut)
+static void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut)
{
float temp[256];
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)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
AT1Ctx *q = avctx->priv_data;
- int ch, ret, i;
+ int ch, ret;
GetBitContext gb;
- float* samples = data;
+ float *samples;
if (buf_size < 212 * q->channels) {
- av_log(q,AV_LOG_ERROR,"Not enought data to decode!\n");
- return -1;
+ 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) {
+ 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++) {
AT1SUCtx* su = &q->SUs[ch];
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, q->channels == 1 ? samples : q->out_samples[ch]);
}
- /* round, convert to 16bit and interleave */
- if (q->channels == 1) {
- /* mono */
- q->dsp.vector_clipf(samples, q->out_samples[0], -32700.0 / (1 << 15),
- 32700.0 / (1 << 15), AT1_SU_SAMPLES);
- } else {
- /* stereo */
- for (i = 0; i < AT1_SU_SAMPLES; i++) {
- samples[i * 2] = av_clipf(q->out_samples[0][i],
- -32700.0 / (1 << 15),
- 32700.0 / (1 << 15));
- samples[i * 2 + 1] = av_clipf(q->out_samples[1][i],
- -32700.0 / (1 << 15),
- 32700.0 / (1 << 15));
- }
+ /* interleave */
+ if (q->channels == 2) {
+ q->fmt_conv.float_interleave(samples, (const float **)q->out_samples,
+ AT1_SU_SAMPLES, 2);
}
- *data_size = q->channels * AT1_SU_SAMPLES * sizeof(*samples);
+ *got_frame_ptr = 1;
+ *(AVFrame *)data = q->frame;
+
return avctx->block_align;
}
+static av_cold int atrac1_decode_end(AVCodecContext * avctx)
+{
+ 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]);
+
+ return 0;
+}
+
+
static av_cold int atrac1_decode_init(AVCodecContext *avctx)
{
AT1Ctx *q = avctx->priv_data;
+ int ret;
- avctx->sample_fmt = SAMPLE_FMT_FLT;
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+ 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 */
- 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);
+ ff_dsputil_init(&q->dsp, avctx);
+ ff_fmt_convert_init(&q->fmt_conv, avctx);
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;
- return 0;
-}
-
-
-static av_cold int atrac1_decode_end(AVCodecContext * avctx) {
- AT1Ctx *q = avctx->priv_data;
+ avcodec_get_frame_defaults(&q->frame);
+ avctx->coded_frame = &q->frame;
- ff_mdct_end(&q->mdct_ctx[0]);
- ff_mdct_end(&q->mdct_ctx[1]);
- ff_mdct_end(&q->mdct_ctx[2]);
return 0;
}
-AVCodec atrac1_decoder = {
- .name = "atrac1",
- .type = CODEC_TYPE_AUDIO,
- .id = CODEC_ID_ATRAC1,
+AVCodec ff_atrac1_decoder = {
+ .name = "atrac1",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = 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,
+ .long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),
};
projects / ffmpeg / blobdiff