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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-23 12:43:46 +02:00
FFmpeg/libavcodec/flac.c
Michael Niedermayer 039e9fe01c Merge remote-tracking branch 'qatar/master'
* qatar/master: (29 commits)
  lavfi: reclassify showfiltfmts as a TESTPROG
  graph2dot: fix printf format specifier
  swscale: yuv2planeX 8bit >=sse2 functions need aligned stack on x86-32.
  vp8: loopfilter >=sse2 functions need aligned stack on x86-32.
  amr: remove shift out of the AMR_BIT() macro.
  dsputilenc: group yasm and inline asm function pointer assignment.
  mov: use forward declaration of a function instead of a table.
  Clarify Doxygen comment for FF_API_* #defines.
  configure: simplify get_version()
  Create version.h headers for libraries that lack them
  gitignore: Use full path instead of relative path to specify patterns
  mpegvideo: remove VLAs
  Add XTEA encryption support in libavutil
  Add Blowfish encryption support in libavutil
  eval: Add the isinf() function and tests for it
  flacdec: move lpc filter to flacdsp
  flacdec: split off channel decorrelation as flacdsp
  avplay: Add an option for not limiting the input buffer size
  FATE: add a test for WMA cover art.
  FATE: add a test for apetag cover art
  ...

Conflicts:
	.gitignore
	configure
	ffplay.c
	libavcodec/Makefile
	libavcodec/error_resilience.c
	libavcodec/mpegvideo.c
	libavcodec/ratecontrol.c
	libavdevice/avdevice.h
	libavfilter/Makefile
	libavfilter/filtfmts.c
	libavfilter/version.h
	libavformat/mov.c
	libavformat/version.h
	libavutil/Makefile
	libavutil/avutil.h
	libavutil/version.h
	libswscale/swscale.h
	libswscale/x86/swscale_mmx.c
	tests/fate/libavutil.mak
	tests/lavfi-regression.sh
	tools/graph2dot.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-07-04 21:03:28 +02:00

153 lines
4.6 KiB
C

/*
* FLAC common code
* Copyright (c) 2009 Justin Ruggles
*
* 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
*/
#include "libavutil/crc.h"
#include "flac.h"
#include "flacdata.h"
static const int8_t sample_size_table[] = { 0, 8, 12, 0, 16, 20, 24, 0 };
static int64_t get_utf8(GetBitContext *gb)
{
int64_t val;
GET_UTF8(val, get_bits(gb, 8), return -1;)
return val;
}
int ff_flac_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb,
FLACFrameInfo *fi, int log_level_offset)
{
int bs_code, sr_code, bps_code;
/* frame sync code */
if ((get_bits(gb, 15) & 0x7FFF) != 0x7FFC) {
av_log(avctx, AV_LOG_ERROR + log_level_offset, "invalid sync code\n");
return -1;
}
/* variable block size stream code */
fi->is_var_size = get_bits1(gb);
/* block size and sample rate codes */
bs_code = get_bits(gb, 4);
sr_code = get_bits(gb, 4);
/* channels and decorrelation */
fi->ch_mode = get_bits(gb, 4);
if (fi->ch_mode < FLAC_MAX_CHANNELS) {
fi->channels = fi->ch_mode + 1;
fi->ch_mode = FLAC_CHMODE_INDEPENDENT;
} else if (fi->ch_mode < FLAC_MAX_CHANNELS + FLAC_CHMODE_MID_SIDE) {
fi->channels = 2;
fi->ch_mode -= FLAC_MAX_CHANNELS - 1;
} else {
av_log(avctx, AV_LOG_ERROR + log_level_offset,
"invalid channel mode: %d\n", fi->ch_mode);
return -1;
}
/* bits per sample */
bps_code = get_bits(gb, 3);
if (bps_code == 3 || bps_code == 7) {
av_log(avctx, AV_LOG_ERROR + log_level_offset,
"invalid sample size code (%d)\n",
bps_code);
return -1;
}
fi->bps = sample_size_table[bps_code];
/* reserved bit */
if (get_bits1(gb)) {
av_log(avctx, AV_LOG_ERROR + log_level_offset,
"broken stream, invalid padding\n");
return -1;
}
/* sample or frame count */
fi->frame_or_sample_num = get_utf8(gb);
if (fi->frame_or_sample_num < 0) {
av_log(avctx, AV_LOG_ERROR + log_level_offset,
"sample/frame number invalid; utf8 fscked\n");
return -1;
}
/* blocksize */
if (bs_code == 0) {
av_log(avctx, AV_LOG_ERROR + log_level_offset,
"reserved blocksize code: 0\n");
return -1;
} else if (bs_code == 6) {
fi->blocksize = get_bits(gb, 8) + 1;
} else if (bs_code == 7) {
fi->blocksize = get_bits(gb, 16) + 1;
} else {
fi->blocksize = ff_flac_blocksize_table[bs_code];
}
/* sample rate */
if (sr_code < 12) {
fi->samplerate = ff_flac_sample_rate_table[sr_code];
} else if (sr_code == 12) {
fi->samplerate = get_bits(gb, 8) * 1000;
} else if (sr_code == 13) {
fi->samplerate = get_bits(gb, 16);
} else if (sr_code == 14) {
fi->samplerate = get_bits(gb, 16) * 10;
} else {
av_log(avctx, AV_LOG_ERROR + log_level_offset,
"illegal sample rate code %d\n",
sr_code);
return -1;
}
/* header CRC-8 check */
skip_bits(gb, 8);
if (av_crc(av_crc_get_table(AV_CRC_8_ATM), 0, gb->buffer,
get_bits_count(gb)/8)) {
av_log(avctx, AV_LOG_ERROR + log_level_offset,
"header crc mismatch\n");
return -1;
}
return 0;
}
int ff_flac_get_max_frame_size(int blocksize, int ch, int bps)
{
/* Technically, there is no limit to FLAC frame size, but an encoder
should not write a frame that is larger than if verbatim encoding mode
were to be used. */
int count;
count = 16; /* frame header */
count += ch * ((7+bps+7)/8); /* subframe headers */
if (ch == 2) {
/* for stereo, need to account for using decorrelation */
count += (( 2*bps+1) * blocksize + 7) / 8;
} else {
count += ( ch*bps * blocksize + 7) / 8;
}
count += 2; /* frame footer */
return count;
}