[SCM] libav/experimental: Calculate an exact frame size before writing. Now the buffer size requirements can be known exactly, so larger frame sizes can be safely encoded without buffer overwrite.

[siretart at users.alioth.debian.org]

The following commit has been merged in the experimental branch:
commit 7c29a5de2553a7229cce33faa3b5646f1764c22e
Author: Justin Ruggles <justin.ruggles at gmail.com>
Date:   Sat Jul 31 20:32:12 2010 +0000

    Calculate an exact frame size before writing.  Now the buffer size requirements
    can be known exactly, so larger frame sizes can be safely encoded without buffer
    overwrite.
    
    Originally committed as revision 24630 to svn://svn.ffmpeg.org/ffmpeg/trunk

diff --git a/libavcodec/flacenc.c b/libavcodec/flacenc.c
index ea3dbe8..dc620b1 100644
--- a/libavcodec/flacenc.c
+++ b/libavcodec/flacenc.c
@@ -493,6 +493,67 @@ static void copy_samples(FlacEncodeContext *s, const int16_t *samples)
 }
 
 
+static int rice_count_exact(int32_t *res, int n, int k)
+{
+    int i;
+    int count = 0;
+
+    for (i = 0; i < n; i++) {
+        int32_t v = -2 * res[i] - 1;
+        v ^= v >> 31;
+        count += (v >> k) + 1 + k;
+    }
+    return count;
+}
+
+
+static int subframe_count_exact(FlacEncodeContext *s, FlacSubframe *sub,
+                                int pred_order)
+{
+    int p, porder, psize;
+    int i, part_end;
+    int count = 0;
+
+    /* subframe header */
+    count += 8;
+
+    /* subframe */
+    if (sub->type == FLAC_SUBFRAME_CONSTANT) {
+        count += sub->obits;
+    } else if (sub->type == FLAC_SUBFRAME_VERBATIM) {
+        count += s->frame.blocksize * sub->obits;
+    } else {
+        /* warm-up samples */
+        count += pred_order * sub->obits;
+
+        /* LPC coefficients */
+        if (sub->type == FLAC_SUBFRAME_LPC)
+            count += 4 + 5 + pred_order * s->options.lpc_coeff_precision;
+
+        /* rice-encoded block */
+        count += 2;
+
+        /* partition order */
+        porder = sub->rc.porder;
+        psize  = s->frame.blocksize >> porder;
+        count += 4;
+
+        /* residual */
+        i        = pred_order;
+        part_end = psize;
+        for (p = 0; p < 1 << porder; p++) {
+            int k = sub->rc.params[p];
+            count += 4;
+            count += rice_count_exact(&sub->residual[i], part_end - i, k);
+            i = part_end;
+            part_end = FFMIN(s->frame.blocksize, part_end + psize);
+        }
+    }
+
+    return count;
+}
+
+
 #define rice_encode_count(sum, n, k) (((n)*((k)+1))+((sum-(n>>1))>>(k)))
 
 /**
@@ -801,14 +862,14 @@ static int encode_residual_ch(FlacEncodeContext *s, int ch)
     if (i == n) {
         sub->type = sub->type_code = FLAC_SUBFRAME_CONSTANT;
         res[0] = smp[0];
-        return sub->obits;
+        return subframe_count_exact(s, sub, 0);
     }
 
     /* VERBATIM */
     if (frame->verbatim_only || n < 5) {
         sub->type = sub->type_code = FLAC_SUBFRAME_VERBATIM;
         memcpy(res, smp, n * sizeof(int32_t));
-        return sub->obits * n;
+        return subframe_count_exact(s, sub, 0);
     }
 
     min_order  = s->options.min_prediction_order;
@@ -834,9 +895,9 @@ static int encode_residual_ch(FlacEncodeContext *s, int ch)
         sub->type_code = sub->type | sub->order;
         if (sub->order != max_order) {
             encode_residual_fixed(res, smp, n, sub->order);
-            return find_subframe_rice_params(s, sub, sub->order);
+            find_subframe_rice_params(s, sub, sub->order);
         }
-        return bits[sub->order];
+        return subframe_count_exact(s, sub, sub->order);
     }
 
     /* LPC */
@@ -908,7 +969,9 @@ static int encode_residual_ch(FlacEncodeContext *s, int ch)
 
     encode_residual_lpc(res, smp, n, sub->order, sub->coefs, sub->shift);
 
-    return find_subframe_rice_params(s, sub, sub->order);
+    find_subframe_rice_params(s, sub, sub->order);
+
+    return subframe_count_exact(s, sub, sub->order);
 }
 
 
@@ -1197,15 +1260,10 @@ static int flac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
 {
     FlacEncodeContext *s;
     const int16_t *samples = data;
-    int out_bytes;
+    int frame_bytes, out_bytes;
 
     s = avctx->priv_data;
 
-    if (buf_size < s->max_framesize * 2) {
-        av_log(avctx, AV_LOG_ERROR, "output buffer too small\n");
-        return 0;
-    }
-
     /* when the last block is reached, update the header in extradata */
     if (!data) {
         s->max_framesize = s->max_encoded_framesize;
@@ -1220,15 +1278,22 @@ static int flac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
 
     channel_decorrelation(s);
 
-    encode_frame(s);
-
+    frame_bytes = encode_frame(s);
+    if (buf_size < frame_bytes) {
+        av_log(avctx, AV_LOG_ERROR, "output buffer too small\n");
+        return 0;
+    }
     out_bytes = write_frame(s, frame, buf_size);
 
     /* fallback to verbatim mode if the compressed frame is larger than it
        would be if encoded uncompressed. */
     if (out_bytes > s->max_framesize) {
         s->frame.verbatim_only = 1;
-        encode_frame(s);
+        frame_bytes = encode_frame(s);
+        if (buf_size < frame_bytes) {
+            av_log(avctx, AV_LOG_ERROR, "output buffer too small\n");
+            return 0;
+        }
         out_bytes = write_frame(s, frame, buf_size);
     }
 

-- 
Libav/FFmpeg packaging