avcodec_decode_audio2()

difference to avcodec_decode_audio() is that the user can pass the allocated size of the output buffer to the decoder and the decoder can check if theres enough space

Originally committed as revision 7518 to svn://svn.ffmpeg.org/ffmpeg/trunk

1 files changed, 8 insertions, 2 deletions

diff --git a/libavcodec/flac.c b/libavcodec/flac.c
index 6c64ad0a1b..e704d990e3 100644
--- a/libavcodec/flac.c
+++ b/libavcodec/flac.c
@@ -454,7 +454,7 @@ static inline int decode_subframe(FLACContext *s, int channel)
     return 0;
 }
 
-static int decode_frame(FLACContext *s)
+static int decode_frame(FLACContext *s, int alloc_data_size)
 {
     int blocksize_code, sample_rate_code, sample_size_code, assignment, i, crc8;
     int decorrelation, bps, blocksize, samplerate;
@@ -516,6 +516,9 @@ static int decode_frame(FLACContext *s)
         return -1;
     }
 
+    if(blocksize * s->channels * sizeof(int16_t) > alloc_data_size)
+        return -1;
+
     if (sample_rate_code == 0){
         samplerate= s->samplerate;
     }else if ((sample_rate_code > 3) && (sample_rate_code < 12))
@@ -579,6 +582,9 @@ static int flac_decode_frame(AVCodecContext *avctx,
     FLACContext *s = avctx->priv_data;
     int tmp = 0, i, j = 0, input_buf_size = 0;
     int16_t *samples = data;
+    int alloc_data_size= *data_size;
+
+    *data_size=0;
 
     if(s->max_framesize == 0){
         s->max_framesize= 65536; // should hopefully be enough for the first header
@@ -617,7 +623,7 @@ static int flac_decode_frame(AVCodecContext *avctx,
             goto end; // we may not have enough bits left to decode a frame, so try next time
         }
         skip_bits(&s->gb, 16);
-        if (decode_frame(s) < 0){
+        if (decode_frame(s, alloc_data_size) < 0){
             av_log(s->avctx, AV_LOG_ERROR, "decode_frame() failed\n");
             s->bitstream_size=0;
             s->bitstream_index=0;