From 5f8de423f190bbb79a62f804151bc24824fa32d8 Mon Sep 17 00:00:00 2001 From: "Matt A. Tobin" Date: Fri, 2 Feb 2018 04:16:08 -0500 Subject: Add m-esr52 at 52.6.0 --- media/ffvpx/libavcodec/flac.c | 237 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 237 insertions(+) create mode 100644 media/ffvpx/libavcodec/flac.c (limited to 'media/ffvpx/libavcodec/flac.c') diff --git a/media/ffvpx/libavcodec/flac.c b/media/ffvpx/libavcodec/flac.c new file mode 100644 index 000000000..f5154b914 --- /dev/null +++ b/media/ffvpx/libavcodec/flac.c @@ -0,0 +1,237 @@ +/* + * 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/channel_layout.h" +#include "libavutil/crc.h" +#include "libavutil/log.h" +#include "bytestream.h" +#include "get_bits.h" +#include "flac.h" +#include "flacdata.h" + +static const int8_t sample_size_table[] = { 0, 8, 12, 0, 16, 20, 24, 0 }; + +static const uint64_t flac_channel_layouts[8] = { + AV_CH_LAYOUT_MONO, + AV_CH_LAYOUT_STEREO, + AV_CH_LAYOUT_SURROUND, + AV_CH_LAYOUT_QUAD, + AV_CH_LAYOUT_5POINT0, + AV_CH_LAYOUT_5POINT1, + AV_CH_LAYOUT_6POINT1, + AV_CH_LAYOUT_7POINT1 +}; + +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 AVERROR_INVALIDDATA; + } + + /* 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 AVERROR_INVALIDDATA; + } + + /* 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 AVERROR_INVALIDDATA; + } + 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 AVERROR_INVALIDDATA; + } + + /* 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 AVERROR_INVALIDDATA; + } + + /* blocksize */ + if (bs_code == 0) { + av_log(avctx, AV_LOG_ERROR + log_level_offset, + "reserved blocksize code: 0\n"); + return AVERROR_INVALIDDATA; + } 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 AVERROR_INVALIDDATA; + } + + /* 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 AVERROR_INVALIDDATA; + } + + 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; +} + +int ff_flac_is_extradata_valid(AVCodecContext *avctx, + enum FLACExtradataFormat *format, + uint8_t **streaminfo_start) +{ + if (!avctx->extradata || avctx->extradata_size < FLAC_STREAMINFO_SIZE) { + av_log(avctx, AV_LOG_ERROR, "extradata NULL or too small.\n"); + return 0; + } + if (AV_RL32(avctx->extradata) != MKTAG('f','L','a','C')) { + /* extradata contains STREAMINFO only */ + if (avctx->extradata_size != FLAC_STREAMINFO_SIZE) { + av_log(avctx, AV_LOG_WARNING, "extradata contains %d bytes too many.\n", + FLAC_STREAMINFO_SIZE-avctx->extradata_size); + } + *format = FLAC_EXTRADATA_FORMAT_STREAMINFO; + *streaminfo_start = avctx->extradata; + } else { + if (avctx->extradata_size < 8+FLAC_STREAMINFO_SIZE) { + av_log(avctx, AV_LOG_ERROR, "extradata too small.\n"); + return 0; + } + *format = FLAC_EXTRADATA_FORMAT_FULL_HEADER; + *streaminfo_start = &avctx->extradata[8]; + } + return 1; +} + +void ff_flac_set_channel_layout(AVCodecContext *avctx) +{ + if (avctx->channels <= FF_ARRAY_ELEMS(flac_channel_layouts)) + avctx->channel_layout = flac_channel_layouts[avctx->channels - 1]; + else + avctx->channel_layout = 0; +} + +void ff_flac_parse_streaminfo(AVCodecContext *avctx, struct FLACStreaminfo *s, + const uint8_t *buffer) +{ + GetBitContext gb; + init_get_bits(&gb, buffer, FLAC_STREAMINFO_SIZE*8); + + skip_bits(&gb, 16); /* skip min blocksize */ + s->max_blocksize = get_bits(&gb, 16); + if (s->max_blocksize < FLAC_MIN_BLOCKSIZE) { + av_log(avctx, AV_LOG_WARNING, "invalid max blocksize: %d\n", + s->max_blocksize); + s->max_blocksize = 16; + } + + skip_bits(&gb, 24); /* skip min frame size */ + s->max_framesize = get_bits_long(&gb, 24); + + s->samplerate = get_bits_long(&gb, 20); + s->channels = get_bits(&gb, 3) + 1; + s->bps = get_bits(&gb, 5) + 1; + + avctx->channels = s->channels; + avctx->sample_rate = s->samplerate; + avctx->bits_per_raw_sample = s->bps; + + if (!avctx->channel_layout || + av_get_channel_layout_nb_channels(avctx->channel_layout) != avctx->channels) + ff_flac_set_channel_layout(avctx); + + s->samples = get_bits64(&gb, 36); + + skip_bits_long(&gb, 64); /* md5 sum */ + skip_bits_long(&gb, 64); /* md5 sum */ +} -- cgit v1.2.3