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039e9fe01c
* 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>
153 lines
4.6 KiB
C
153 lines
4.6 KiB
C
/*
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* FLAC common code
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* Copyright (c) 2009 Justin Ruggles
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/crc.h"
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#include "flac.h"
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#include "flacdata.h"
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static const int8_t sample_size_table[] = { 0, 8, 12, 0, 16, 20, 24, 0 };
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static int64_t get_utf8(GetBitContext *gb)
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{
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int64_t val;
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GET_UTF8(val, get_bits(gb, 8), return -1;)
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return val;
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}
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int ff_flac_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb,
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FLACFrameInfo *fi, int log_level_offset)
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{
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int bs_code, sr_code, bps_code;
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/* frame sync code */
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if ((get_bits(gb, 15) & 0x7FFF) != 0x7FFC) {
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av_log(avctx, AV_LOG_ERROR + log_level_offset, "invalid sync code\n");
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return -1;
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}
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/* variable block size stream code */
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fi->is_var_size = get_bits1(gb);
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/* block size and sample rate codes */
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bs_code = get_bits(gb, 4);
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sr_code = get_bits(gb, 4);
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/* channels and decorrelation */
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fi->ch_mode = get_bits(gb, 4);
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if (fi->ch_mode < FLAC_MAX_CHANNELS) {
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fi->channels = fi->ch_mode + 1;
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fi->ch_mode = FLAC_CHMODE_INDEPENDENT;
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} else if (fi->ch_mode < FLAC_MAX_CHANNELS + FLAC_CHMODE_MID_SIDE) {
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fi->channels = 2;
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fi->ch_mode -= FLAC_MAX_CHANNELS - 1;
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} else {
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av_log(avctx, AV_LOG_ERROR + log_level_offset,
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"invalid channel mode: %d\n", fi->ch_mode);
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return -1;
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}
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/* bits per sample */
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bps_code = get_bits(gb, 3);
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if (bps_code == 3 || bps_code == 7) {
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av_log(avctx, AV_LOG_ERROR + log_level_offset,
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"invalid sample size code (%d)\n",
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bps_code);
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return -1;
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}
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fi->bps = sample_size_table[bps_code];
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/* reserved bit */
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if (get_bits1(gb)) {
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av_log(avctx, AV_LOG_ERROR + log_level_offset,
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"broken stream, invalid padding\n");
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return -1;
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}
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/* sample or frame count */
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fi->frame_or_sample_num = get_utf8(gb);
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if (fi->frame_or_sample_num < 0) {
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av_log(avctx, AV_LOG_ERROR + log_level_offset,
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"sample/frame number invalid; utf8 fscked\n");
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return -1;
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}
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/* blocksize */
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if (bs_code == 0) {
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av_log(avctx, AV_LOG_ERROR + log_level_offset,
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"reserved blocksize code: 0\n");
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return -1;
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} else if (bs_code == 6) {
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fi->blocksize = get_bits(gb, 8) + 1;
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} else if (bs_code == 7) {
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fi->blocksize = get_bits(gb, 16) + 1;
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} else {
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fi->blocksize = ff_flac_blocksize_table[bs_code];
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}
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/* sample rate */
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if (sr_code < 12) {
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fi->samplerate = ff_flac_sample_rate_table[sr_code];
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} else if (sr_code == 12) {
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fi->samplerate = get_bits(gb, 8) * 1000;
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} else if (sr_code == 13) {
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fi->samplerate = get_bits(gb, 16);
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} else if (sr_code == 14) {
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fi->samplerate = get_bits(gb, 16) * 10;
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} else {
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av_log(avctx, AV_LOG_ERROR + log_level_offset,
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"illegal sample rate code %d\n",
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sr_code);
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return -1;
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}
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/* header CRC-8 check */
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skip_bits(gb, 8);
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if (av_crc(av_crc_get_table(AV_CRC_8_ATM), 0, gb->buffer,
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get_bits_count(gb)/8)) {
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av_log(avctx, AV_LOG_ERROR + log_level_offset,
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"header crc mismatch\n");
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return -1;
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}
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return 0;
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}
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int ff_flac_get_max_frame_size(int blocksize, int ch, int bps)
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{
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/* Technically, there is no limit to FLAC frame size, but an encoder
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should not write a frame that is larger than if verbatim encoding mode
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were to be used. */
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int count;
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count = 16; /* frame header */
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count += ch * ((7+bps+7)/8); /* subframe headers */
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if (ch == 2) {
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/* for stereo, need to account for using decorrelation */
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count += (( 2*bps+1) * blocksize + 7) / 8;
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} else {
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count += ( ch*bps * blocksize + 7) / 8;
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}
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count += 2; /* frame footer */
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return count;
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}
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