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

4 files changed, 41 insertions, 10 deletions

diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index 8c908105cc..ed92f1a980 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -2511,18 +2511,21 @@ int avcodec_default_execute(AVCodecContext *c, int (*func)(AVCodecContext *c2, v
  */
 int avcodec_open(AVCodecContext *avctx, AVCodec *codec);
 
+
+attribute_deprecated int avcodec_decode_audio(AVCodecContext *avctx, int16_t *samples,
+                         int *frame_size_ptr,
+                         uint8_t *buf, int buf_size);
 /**
  * Decode an audio frame.
  *
  * @param avctx the codec context.
  * @param samples output buffer, 16 byte aligned
- * @param frame_size_ptr the output buffer size in bytes, zero if no frame could be compressed
+ * @param frame_size_ptr the output buffer size in bytes (you MUST set this to the allocated size before calling avcodec_decode_audio2()), zero if no frame could be compressed
  * @param buf input buffer, 16 byte aligned
  * @param buf_size the input buffer size
  * @return 0 if successful, -1 if not.
  */
-
-int avcodec_decode_audio(AVCodecContext *avctx, int16_t *samples,
+int avcodec_decode_audio2(AVCodecContext *avctx, int16_t *samples,
                          int *frame_size_ptr,
                          uint8_t *buf, int buf_size);
 int avcodec_decode_video(AVCodecContext *avctx, AVFrame *picture,
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;
diff --git a/libavcodec/pcm.c b/libavcodec/pcm.c
index 26c38b3298..4011ed3b57 100644
--- a/libavcodec/pcm.c
+++ b/libavcodec/pcm.c
@@ -410,8 +410,8 @@ static int pcm_decode_frame(AVCodecContext *avctx,
     samples = data;
     src = buf;
 
-    if(buf_size > AVCODEC_MAX_AUDIO_FRAME_SIZE/2)
-        buf_size = AVCODEC_MAX_AUDIO_FRAME_SIZE/2;
+    buf_size= FFMIN(buf_size, *data_size/2);
+    *data_size=0;
 
     switch(avctx->codec->id) {
     case CODEC_ID_PCM_S32LE:
diff --git a/libavcodec/utils.c b/libavcodec/utils.c
index 899061ec27..f6f0613603 100644
--- a/libavcodec/utils.c
+++ b/libavcodec/utils.c
@@ -918,22 +918,44 @@ int avcodec_decode_video(AVCodecContext *avctx, AVFrame *picture,
    *number of bytes used. If no frame could be decompressed,
    *frame_size_ptr is zero. Otherwise, it is the decompressed frame
    *size in BYTES. */
-int avcodec_decode_audio(AVCodecContext *avctx, int16_t *samples,
+int avcodec_decode_audio2(AVCodecContext *avctx, int16_t *samples,
                          int *frame_size_ptr,
                          uint8_t *buf, int buf_size)
 {
     int ret;
 
-    *frame_size_ptr= 0;
+    //FIXME remove the check below _after_ ensuring that all audio check that the available space is enough
+    if(*frame_size_ptr < AVCODEC_MAX_AUDIO_FRAME_SIZE){
+        av_log(avctx, AV_LOG_ERROR, "buffer smaller then AVCODEC_MAX_AUDIO_FRAME_SIZE\n");
+        return -1;
+    }
+    if(*frame_size_ptr < FF_MIN_BUFFER_SIZE ||
+       *frame_size_ptr < avctx->channels * avctx->frame_size * sizeof(int16_t) ||
+       *frame_size_ptr < buf_size){
+        av_log(avctx, AV_LOG_ERROR, "buffer too small\n");
+        return -1;
+    }
     if((avctx->codec->capabilities & CODEC_CAP_DELAY) || buf_size){
         ret = avctx->codec->decode(avctx, samples, frame_size_ptr,
                                 buf, buf_size);
         avctx->frame_number++;
-    }else
+    }else{
         ret= 0;
+        *frame_size_ptr=0;
+    }
     return ret;
 }
 
+#if LIBAVCODEC_VERSION_INT < ((52<<16)+(0<<8)+0)
+int avcodec_decode_audio(AVCodecContext *avctx, int16_t *samples,
+                         int *frame_size_ptr,
+                         uint8_t *buf, int buf_size){
+    *frame_size_ptr= AVCODEC_MAX_AUDIO_FRAME_SIZE;
+    return avcodec_decode_audio2(avctx, samples, frame_size_ptr, buf, buf_size);
+}
+#endif
+
+
 /* decode a subtitle message. return -1 if error, otherwise return the
    *number of bytes used. If no subtitle could be decompressed,
    *got_sub_ptr is zero. Otherwise, the subtitle is stored in *sub. */