FFmpeg/libavcodec/mpegaudiodec_float.c
Michael Niedermayer 75a37b57a5 Merge remote-tracking branch 'qatar/master'
* qatar/master:
  APIchanges: fill in date and commit for request_sample_fmt
  Add floating-point sample format support to the ac3, eac3, dca, aac, and vorbis decoders.
  Add support for request_sample_format in ffmpeg and ffplay.
  Add APIchanges entry for request_sample_fmt.
  Add request_sample_fmt field to AVCodecContext.
  Add float_interleave() to FmtConvertContext with x86-optimized versions.
  Remove unused make variable SEEK_REFFILE
  fate: remove redundant aref and vref references
  fate: remove do_ffmpeg_nocheck function
  fate: do not collect -benchmark output
  mpegaudiodec: remove decode_end() function
  fate: run aref and vref as regular tests
  mpegaudio: sanitise compute_antialias_* names
  mpeg12: add slice-threading checks to slice-threading initializers.
  h264: copy pixel_shift between slice threading contexts.
  mdec: enable frame-level multithreading.
  mdec.c: fix overread.

Conflicts:
	libavcodec/aacdec.c
	libavcodec/ac3dec.c
	libavcodec/avcodec.h
	libavcodec/dca.c
	libavcodec/h264.c
	libavcodec/mdec.c
	libavcodec/mpeg12.c
	libavcodec/options.c
	libavcodec/version.h
	libavcodec/vorbisdec.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-05-19 06:00:31 +02:00

162 lines
4.0 KiB
C

/*
* Float MPEG Audio decoder
* Copyright (c) 2010 Michael Niedermayer
*
* This file is part of FFmpeg.
*
* FFmpeg 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,
* 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
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#define CONFIG_FLOAT 1
#include "mpegaudiodec.c"
void ff_mpa_synth_filter_float(MPADecodeContext *s, float *synth_buf_ptr,
int *synth_buf_offset,
float *window, int *dither_state,
float *samples, int incr,
float sb_samples[SBLIMIT])
{
float *synth_buf;
int offset;
offset = *synth_buf_offset;
synth_buf = synth_buf_ptr + offset;
s->dct.dct32(synth_buf, sb_samples);
s->apply_window_mp3(synth_buf, window, dither_state, samples, incr);
offset = (offset - 32) & 511;
*synth_buf_offset = offset;
}
static void compute_antialias_float(MPADecodeContext *s,
GranuleDef *g)
{
float *ptr;
int n, i;
/* we antialias only "long" bands */
if (g->block_type == 2) {
if (!g->switch_point)
return;
/* XXX: check this for 8000Hz case */
n = 1;
} else {
n = SBLIMIT - 1;
}
ptr = g->sb_hybrid + 18;
for(i = n;i > 0;i--) {
float tmp0, tmp1;
float *csa = &csa_table_float[0][0];
#define FLOAT_AA(j)\
tmp0= ptr[-1-j];\
tmp1= ptr[ j];\
ptr[-1-j] = tmp0 * csa[0+4*j] - tmp1 * csa[1+4*j];\
ptr[ j] = tmp0 * csa[1+4*j] + tmp1 * csa[0+4*j];
FLOAT_AA(0)
FLOAT_AA(1)
FLOAT_AA(2)
FLOAT_AA(3)
FLOAT_AA(4)
FLOAT_AA(5)
FLOAT_AA(6)
FLOAT_AA(7)
ptr += 18;
}
}
#if CONFIG_MP1FLOAT_DECODER
AVCodec ff_mp1float_decoder =
{
"mp1float",
AVMEDIA_TYPE_AUDIO,
CODEC_ID_MP1,
sizeof(MPADecodeContext),
decode_init,
NULL,
.close = NULL,
decode_frame,
CODEC_CAP_PARSE_ONLY,
.flush= flush,
.long_name= NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
};
#endif
#if CONFIG_MP2FLOAT_DECODER
AVCodec ff_mp2float_decoder =
{
"mp2float",
AVMEDIA_TYPE_AUDIO,
CODEC_ID_MP2,
sizeof(MPADecodeContext),
decode_init,
NULL,
.close = NULL,
decode_frame,
CODEC_CAP_PARSE_ONLY,
.flush= flush,
.long_name= NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
};
#endif
#if CONFIG_MP3FLOAT_DECODER
AVCodec ff_mp3float_decoder =
{
"mp3float",
AVMEDIA_TYPE_AUDIO,
CODEC_ID_MP3,
sizeof(MPADecodeContext),
decode_init,
NULL,
.close = NULL,
decode_frame,
CODEC_CAP_PARSE_ONLY,
.flush= flush,
.long_name= NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
};
#endif
#if CONFIG_MP3ADUFLOAT_DECODER
AVCodec ff_mp3adufloat_decoder =
{
"mp3adufloat",
AVMEDIA_TYPE_AUDIO,
CODEC_ID_MP3ADU,
sizeof(MPADecodeContext),
decode_init,
NULL,
.close = NULL,
decode_frame_adu,
CODEC_CAP_PARSE_ONLY,
.flush= flush,
.long_name= NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),
};
#endif
#if CONFIG_MP3ON4FLOAT_DECODER
AVCodec ff_mp3on4float_decoder =
{
"mp3on4float",
AVMEDIA_TYPE_AUDIO,
CODEC_ID_MP3ON4,
sizeof(MP3On4DecodeContext),
decode_init_mp3on4,
NULL,
decode_close_mp3on4,
decode_frame_mp3on4,
.flush= flush,
.long_name= NULL_IF_CONFIG_SMALL("MP3onMP4"),
};
#endif