* 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 "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;
/**
*/
typedef struct {
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];
+ DECLARE_ALIGNED(32, float, out_samples)[AT1_MAX_CHANNELS][AT1_SU_SAMPLES];
FFTContext mdct_ctx[3];
int channels;
DSPContext dsp;
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);
}
/* 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;
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 */
}
-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];
at1_subband_synthesis(q, su, q->out_samples[ch]);
}
- /* round, convert to 16bit and interleave */
+ /* interleave; FIXME, should create/use a DSP function */
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);
+ memcpy(samples, q->out_samples[0], AT1_SU_SAMPLES * 4);
} 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));
+ samples[i * 2] = q->out_samples[0][i];
+ samples[i * 2 + 1] = q->out_samples[1][i];
}
}
{
AT1Ctx *q = avctx->priv_data;
- avctx->sample_fmt = SAMPLE_FMT_FLT;
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
q->channels = avctx->channels;
}
-AVCodec atrac1_decoder = {
+AVCodec ff_atrac1_decoder = {
.name = "atrac1",
- .type = CODEC_TYPE_AUDIO,
+ .type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_ATRAC1,
.priv_data_size = sizeof(AT1Ctx),
.init = atrac1_decode_init,