1 files changed, 47 insertions, 12 deletions

diff --git a/libavcodec/flacdec.c b/libavcodec/flacdec.c
index 0305d50..85f5202 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 +33,7 @@
 
 #include <limits.h>
 
+#include "libavutil/avassert.h"
 #include "libavutil/channel_layout.h"
 #include "libavutil/crc.h"
 #include "avcodec.h"
@@ -43,6 +44,7 @@
 #include "flac.h"
 #include "flacdata.h"
 #include "flacdsp.h"
+#include "thread.h"
 
 typedef struct FLACContext {
     FLACSTREAMINFO
@@ -127,6 +129,8 @@ static int allocate_buffers(FLACContext *s)
 {
     int buf_size;
 
+    av_assert0(s->max_blocksize);
+
     buf_size = av_samples_get_buffer_size(NULL, s->channels, s->max_blocksize,
                                           AV_SAMPLE_FMT_S32P, 0);
     if (buf_size < 0)
@@ -247,7 +251,8 @@ 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++) {
@@ -485,6 +490,7 @@ 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;
@@ -499,6 +505,16 @@ 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, "skiping 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, "skiping 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 */
@@ -515,19 +531,26 @@ 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_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);
@@ -546,6 +569,17 @@ 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;
@@ -564,10 +598,11 @@ 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 },
 };