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-rw-r--r--libavcodec/flacdec.c76
1 files changed, 60 insertions, 16 deletions
diff --git a/libavcodec/flacdec.c b/libavcodec/flacdec.c
index 6cce6923f3..f1316658e7 100644
--- a/libavcodec/flacdec.c
+++ b/libavcodec/flacdec.c
@@ -2,20 +2,20 @@
* FLAC (Free Lossless Audio Codec) decoder
* Copyright (c) 2003 Alex Beregszaszi
*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * 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.
*
- * Libav is distributed in the hope that it will be useful,
+ * 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 Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
@@ -33,6 +33,8 @@
#include <limits.h>
+#include "libavutil/avassert.h"
+#include "libavutil/crc.h"
#include "avcodec.h"
#include "internal.h"
#include "get_bits.h"
@@ -41,6 +43,9 @@
#include "flac.h"
#include "flacdata.h"
#include "flacdsp.h"
+#include "thread.h"
+#include "unary.h"
+
typedef struct FLACContext {
FLACSTREAMINFO
@@ -124,6 +129,9 @@ static void dump_headers(AVCodecContext *avctx, FLACStreaminfo *s)
static int allocate_buffers(FLACContext *s)
{
int buf_size;
+ int ret;
+
+ av_assert0(s->max_blocksize);
buf_size = av_samples_get_buffer_size(NULL, s->channels, s->max_blocksize,
AV_SAMPLE_FMT_S32P, 0);
@@ -134,9 +142,10 @@ static int allocate_buffers(FLACContext *s)
if (!s->decoded_buffer)
return AVERROR(ENOMEM);
- return av_samples_fill_arrays((uint8_t **)s->decoded, NULL,
+ ret = av_samples_fill_arrays((uint8_t **)s->decoded, NULL,
s->decoded_buffer, s->channels,
s->max_blocksize, AV_SAMPLE_FMT_S32P, 0);
+ return ret < 0 ? ret : 0;
}
/**
@@ -213,6 +222,12 @@ static int decode_residuals(FLACContext *s, int32_t *decoded, int pred_order)
rice_order = get_bits(&s->gb, 4);
samples= s->blocksize >> rice_order;
+ if (samples << rice_order != s->blocksize) {
+ av_log(s->avctx, AV_LOG_ERROR, "invalid rice order: %i blocksize %i\n",
+ rice_order, s->blocksize);
+ return AVERROR_INVALIDDATA;
+ }
+
if (pred_order > samples) {
av_log(s->avctx, AV_LOG_ERROR, "invalid predictor order: %i > %i\n",
pred_order, samples);
@@ -245,7 +260,8 @@ static int decode_subframe_fixed(FLACContext *s, int32_t *decoded,
int pred_order, int bps)
{
const int blocksize = s->blocksize;
- int a, b, c, d, i, ret;
+ int av_uninit(a), av_uninit(b), av_uninit(c), av_uninit(d), i;
+ int ret;
/* warm up samples */
for (i = 0; i < pred_order; i++) {
@@ -350,7 +366,6 @@ static inline int decode_subframe(FLACContext *s, int channel)
if (get_bits1(&s->gb)) {
int left = get_bits_left(&s->gb);
- wasted = 1;
if ( left < 0 ||
(left < bps && !show_bits_long(&s->gb, left)) ||
!show_bits_long(&s->gb, bps)) {
@@ -359,8 +374,7 @@ static inline int decode_subframe(FLACContext *s, int channel)
bps, left);
return AVERROR_INVALIDDATA;
}
- while (!get_bits1(&s->gb))
- wasted++;
+ wasted = 1 + get_unary(&s->gb, 1, get_bits_left(&s->gb));
bps -= wasted;
}
if (bps > 32) {
@@ -483,6 +497,7 @@ static int flac_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
AVFrame *frame = data;
+ ThreadFrame tframe = { .f = data };
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
FLACContext *s = avctx->priv_data;
@@ -497,6 +512,16 @@ static int flac_decode_frame(AVCodecContext *avctx, void *data,
FLAC_MAX_CHANNELS, 32);
}
+ if (buf_size > 5 && !memcmp(buf, "\177FLAC", 5)) {
+ av_log(s->avctx, AV_LOG_DEBUG, "skipping flac header packet 1\n");
+ return buf_size;
+ }
+
+ if (buf_size > 0 && (*buf & 0x7F) == FLAC_METADATA_TYPE_VORBIS_COMMENT) {
+ av_log(s->avctx, AV_LOG_DEBUG, "skipping vorbis comment\n");
+ return buf_size;
+ }
+
/* check that there is at least the smallest decodable amount of data.
this amount corresponds to the smallest valid FLAC frame possible.
FF F8 69 02 00 00 9A 00 00 34 46 */
@@ -513,19 +538,26 @@ static int flac_decode_frame(AVCodecContext *avctx, void *data,
}
/* decode frame */
- init_get_bits(&s->gb, buf, buf_size*8);
+ if ((ret = init_get_bits8(&s->gb, buf, buf_size)) < 0)
+ return ret;
if ((ret = decode_frame(s)) < 0) {
av_log(s->avctx, AV_LOG_ERROR, "decode_frame() failed\n");
return ret;
}
- bytes_read = (get_bits_count(&s->gb)+7)/8;
+ bytes_read = get_bits_count(&s->gb)/8;
+
+ if ((s->avctx->err_recognition & (AV_EF_CRCCHECK|AV_EF_COMPLIANT)) &&
+ av_crc(av_crc_get_table(AV_CRC_16_ANSI),
+ 0, buf, bytes_read)) {
+ av_log(s->avctx, AV_LOG_ERROR, "CRC error at PTS %"PRId64"\n", avpkt->pts);
+ if (s->avctx->err_recognition & AV_EF_EXPLODE)
+ return AVERROR_INVALIDDATA;
+ }
/* get output buffer */
frame->nb_samples = s->blocksize;
- if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
- av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ if ((ret = ff_thread_get_buffer(avctx, &tframe, 0)) < 0)
return ret;
- }
s->dsp.decorrelate[s->ch_mode](frame->data, s->decoded, s->channels,
s->blocksize, s->sample_shift);
@@ -544,6 +576,17 @@ static int flac_decode_frame(AVCodecContext *avctx, void *data,
return bytes_read;
}
+static int init_thread_copy(AVCodecContext *avctx)
+{
+ FLACContext *s = avctx->priv_data;
+ s->decoded_buffer = NULL;
+ s->decoded_buffer_size = 0;
+ s->avctx = avctx;
+ if (s->max_blocksize)
+ return allocate_buffers(s);
+ return 0;
+}
+
static av_cold int flac_decode_close(AVCodecContext *avctx)
{
FLACContext *s = avctx->priv_data;
@@ -562,10 +605,11 @@ AVCodec ff_flac_decoder = {
.init = flac_decode_init,
.close = flac_decode_close,
.decode = flac_decode_frame,
- .capabilities = CODEC_CAP_DR1,
+ .init_thread_copy = ONLY_IF_THREADS_ENABLED(init_thread_copy),
+ .capabilities = CODEC_CAP_DR1 | CODEC_CAP_FRAME_THREADS,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_S32,
AV_SAMPLE_FMT_S32P,
- -1 },
+ AV_SAMPLE_FMT_NONE },
};
generated by cgit v1.2.3 (git 2.39.1) at 2024-06-29 18:25:40 +0000