* Copyright (c) 2006-2008 Maxim Poliakovski
* Copyright (c) 2006-2008 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/atrac3.c
+ * @file
* Atrac 3 compatible decoder.
* This decoder handles Sony's ATRAC3 data.
*
#include "get_bits.h"
#include "dsputil.h"
#include "bytestream.h"
+#include "fft.h"
#include "atrac.h"
#include "atrac3data.h"
int gcBlkSwitch;
gain_block gainBlock[2];
- DECLARE_ALIGNED(16, float, spectrum)[1024];
- DECLARE_ALIGNED(16, float, IMDCT_buf)[1024];
+ DECLARE_ALIGNED(32, float, spectrum)[1024];
+ DECLARE_ALIGNED(32, float, IMDCT_buf)[1024];
float delayBuf1[46]; ///<qmf delay buffers
float delayBuf2[46];
int scrambled_stream;
int frame_factor;
//@}
+
+ FFTContext mdct_ctx;
} ATRAC3Context;
-static DECLARE_ALIGNED(16, float,mdct_window)[512];
+static DECLARE_ALIGNED(32, float, mdct_window)[512];
static VLC spectral_coeff_tab[7];
static float gain_tab1[16];
static float gain_tab2[31];
-static FFTContext mdct_ctx;
static DSPContext dsp;
* @param odd_band 1 if the band is an odd band
*/
-static void IMLT(float *pInput, float *pOutput, int odd_band)
+static void IMLT(ATRAC3Context *q, float *pInput, float *pOutput, int odd_band)
{
int i;
/**
* Reverse the odd bands before IMDCT, this is an effect of the QMF transform
* or it gives better compression to do it this way.
- * FIXME: It should be possible to handle this in ff_imdct_calc
+ * FIXME: It should be possible to handle this in imdct_calc
* for that to happen a modification of the prerotation step of
* all SIMD code and C code is needed.
* Or fix the functions before so they generate a pre reversed spectrum.
FFSWAP(float, pInput[i], pInput[255-i]);
}
- ff_imdct_calc(&mdct_ctx,pOutput,pInput);
+ q->mdct_ctx.imdct_calc(&q->mdct_ctx,pOutput,pInput);
/* Perform windowing on the output. */
- dsp.vector_fmul(pOutput,mdct_window,512);
+ dsp.vector_fmul(pOutput, pOutput, mdct_window, 512);
}
/**
* Atrac 3 indata descrambling, only used for data coming from the rm container
*
- * @param in pointer to 8 bit array of indata
- * @param bits amount of bits
+ * @param inbuffer pointer to 8 bit array of indata
* @param out pointer to 8 bit array of outdata
+ * @param bytes amount of bytes
*/
static int decode_bytes(const uint8_t* inbuffer, uint8_t* out, int bytes){
off = (intptr_t)inbuffer & 3;
buf = (const uint32_t*) (inbuffer - off);
- c = be2me_32((0x537F6103 >> (off*8)) | (0x537F6103 << (32-(off*8))));
+ c = av_be2ne32((0x537F6103 >> (off*8)) | (0x537F6103 << (32-(off*8))));
bytes += 3 + off;
for (i = 0; i < bytes/4; i++)
obuf[i] = c ^ buf[i];
if (off)
- av_log(NULL,AV_LOG_DEBUG,"Offset of %d not handled, post sample on ffmpeg-dev.\n",off);
+ av_log_ask_for_sample(NULL, "Offset of %d not handled.\n", off);
return off;
}
}
/* Initialize the MDCT transform. */
- ff_mdct_init(&mdct_ctx, 9, 1, 1.0);
+ ff_mdct_init(&q->mdct_ctx, 9, 1, 1.0);
}
/**
av_free(q->pUnits);
av_free(q->decoded_bytes_buffer);
+ ff_mdct_end(&q->mdct_ctx);
return 0;
}
readQuantSpectralCoeffs (gb, subband_vlc_index[cnt], codingMode, mantissas, subbWidth);
/* Decode the scale factor for this subband. */
- SF = sf_table[SF_idxs[cnt]] * iMaxQuant[subband_vlc_index[cnt]];
+ SF = ff_atrac_sf_table[SF_idxs[cnt]] * iMaxQuant[subband_vlc_index[cnt]];
/* Inverse quantize the coefficients. */
for (pIn=mantissas ; first<last; first++, pIn++)
coded_values = coded_values_per_component + 1;
coded_values = FFMIN(max_coded_values,coded_values);
- scalefactor = sf_table[sfIndx] * iMaxQuant[quant_step_index];
+ scalefactor = ff_atrac_sf_table[sfIndx] * iMaxQuant[quant_step_index];
readQuantSpectralCoeffs(gb, quant_step_index, coding_mode, mantissa, coded_values);
for (band=0; band<4; band++) {
/* Perform the IMDCT step without overlapping. */
if (band <= numBands) {
- IMLT(&(pSnd->spectrum[band*256]), pSnd->IMDCT_buf, band&1);
+ IMLT(q, &(pSnd->spectrum[band*256]), pSnd->IMDCT_buf, band&1);
} else
memset(pSnd->IMDCT_buf, 0, 512 * sizeof(float));
const uint8_t* databuf;
int16_t* samples = data;
- if (buf_size < avctx->block_align)
+ if (buf_size < avctx->block_align) {
+ av_log(avctx, AV_LOG_ERROR,
+ "Frame too small (%d bytes). Truncated file?\n", buf_size);
+ *data_size = 0;
return buf_size;
+ }
/* Check if we need to descramble and what buffer to pass on. */
if (q->scrambled_stream) {
return AVERROR(ENOMEM);
}
- avctx->sample_fmt = SAMPLE_FMT_S16;
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16;
return 0;
}
-AVCodec atrac3_decoder =
+AVCodec ff_atrac3_decoder =
{
.name = "atrac3",
- .type = CODEC_TYPE_AUDIO,
+ .type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_ATRAC3,
.priv_data_size = sizeof(ATRAC3Context),
.init = atrac3_decode_init,
projects / ffmpeg / blobdiff