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authorMatt A. Tobin <mattatobin@localhost.localdomain>2018-02-02 04:16:08 -0500
committerMatt A. Tobin <mattatobin@localhost.localdomain>2018-02-02 04:16:08 -0500
commit5f8de423f190bbb79a62f804151bc24824fa32d8 (patch)
tree10027f336435511475e392454359edea8e25895d /media/ffvpx/libavcodec/flac.c
parent49ee0794b5d912db1f95dce6eb52d781dc210db5 (diff)
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Add m-esr52 at 52.6.0
Diffstat (limited to 'media/ffvpx/libavcodec/flac.c')
-rw-r--r--media/ffvpx/libavcodec/flac.c237
1 files changed, 237 insertions, 0 deletions
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 */
+}