[Ffmpeg-cvslog] r7518 - in trunk/libavcodec: avcodec.h flac.c pcm.c utils.c

[michael]

Author: michael
Date: Mon Jan 15 00:50:06 2007
New Revision: 7518

Modified:
   trunk/libavcodec/avcodec.h
   trunk/libavcodec/flac.c
   trunk/libavcodec/pcm.c
   trunk/libavcodec/utils.c

Log:
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


Modified: trunk/libavcodec/avcodec.h
==============================================================================
--- trunk/libavcodec/avcodec.h	(original)
+++ trunk/libavcodec/avcodec.h	Mon Jan 15 00:50:06 2007
@@ -2511,18 +2511,21 @@
  */
 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,

Modified: trunk/libavcodec/flac.c
==============================================================================
--- trunk/libavcodec/flac.c	(original)
+++ trunk/libavcodec/flac.c	Mon Jan 15 00:50:06 2007
@@ -454,7 +454,7 @@
     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 @@
         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 @@
     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 @@
             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;

Modified: trunk/libavcodec/pcm.c
==============================================================================
--- trunk/libavcodec/pcm.c	(original)
+++ trunk/libavcodec/pcm.c	Mon Jan 15 00:50:06 2007
@@ -410,8 +410,8 @@
     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:

Modified: trunk/libavcodec/utils.c
==============================================================================
--- trunk/libavcodec/utils.c	(original)
+++ trunk/libavcodec/utils.c	Mon Jan 15 00:50:06 2007
@@ -918,22 +918,44 @@
    *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. */