[SCM] libav/experimental: Avoid allocating MPADecodeContext on stack.

[siretart at users.alioth.debian.org]

The following commit has been merged in the experimental branch:
commit cf92cec7d8899ef887869628c66da30737ee54af
Author: Andreas Öman <andreas at lonelycoder.com>
Date:   Fri Jan 23 12:09:32 2009 +0000

    Avoid allocating MPADecodeContext on stack.
    
    Instead move relevant fields into MPADecodeHeader and use it
    where appropriate.
    
    Originally committed as revision 16728 to svn://svn.ffmpeg.org/ffmpeg/trunk

diff --git a/libavcodec/mpegaudio.h b/libavcodec/mpegaudio.h
index 96bea31..f32e6ab 100644
--- a/libavcodec/mpegaudio.h
+++ b/libavcodec/mpegaudio.h
@@ -90,23 +90,30 @@ typedef int32_t MPA_INT;
 
 struct GranuleDef;
 
+#define MPA_DECODE_HEADER \
+    int frame_size; \
+    int error_protection; \
+    int layer; \
+    int sample_rate; \
+    int sample_rate_index; /* between 0 and 8 */ \
+    int bit_rate; \
+    int nb_channels; \
+    int mode; \
+    int mode_ext; \
+    int lsf;
+
+typedef struct MPADecodeHeader {
+  MPA_DECODE_HEADER
+} MPADecodeHeader;
+
 typedef struct MPADecodeContext {
+    MPA_DECODE_HEADER
     DECLARE_ALIGNED_8(uint8_t, last_buf[2*BACKSTEP_SIZE + EXTRABYTES]);
     int last_buf_size;
-    int frame_size;
     /* next header (used in free format parsing) */
     uint32_t free_format_next_header;
-    int error_protection;
-    int layer;
-    int sample_rate;
-    int sample_rate_index; /* between 0 and 8 */
-    int bit_rate;
     GetBitContext gb;
     GetBitContext in_gb;
-    int nb_channels;
-    int mode;
-    int mode_ext;
-    int lsf;
     DECLARE_ALIGNED_16(MPA_INT, synth_buf[MPA_MAX_CHANNELS][512 * 2]);
     int synth_buf_offset[MPA_MAX_CHANNELS];
     DECLARE_ALIGNED_16(int32_t, sb_samples[MPA_MAX_CHANNELS][36][SBLIMIT]);
diff --git a/libavcodec/mpegaudio_parser.c b/libavcodec/mpegaudio_parser.c
index b06b15a..646da8b 100644
--- a/libavcodec/mpegaudio_parser.c
+++ b/libavcodec/mpegaudio_parser.c
@@ -46,8 +46,7 @@ typedef struct MpegAudioParseContext {
    header, otherwise the coded frame size in bytes */
 int ff_mpa_decode_header(AVCodecContext *avctx, uint32_t head, int *sample_rate, int *channels, int *frame_size, int *bit_rate)
 {
-    MPADecodeContext s1, *s = &s1;
-    s1.avctx = avctx;
+    MPADecodeHeader s1, *s = &s1;
 
     if (ff_mpa_check_header(head) != 0)
         return -1;
@@ -145,7 +144,7 @@ static int mpegaudio_parse(AVCodecParserContext *s1,
 
 #if 0
                     /* free format: prepare to compute frame size */
-                    if (ff_mpegaudio_decode_header(s, header) == 1) {
+                    if (ff_mpegaudio_decode_header((MPADecodeHeader *)s, header) == 1) {
                         s->frame_size = -1;
                     }
 #endif
@@ -200,7 +199,7 @@ static int mpegaudio_parse(AVCodecParserContext *s1,
                             s->free_format_frame_size -= padding;
                         dprintf(avctx, "free frame size=%d padding=%d\n",
                                 s->free_format_frame_size, padding);
-                        ff_mpegaudio_decode_header(s, header1);
+                        ff_mpegaudio_decode_header((MPADecodeHeader *)s, header1);
                         goto next_data;
                     }
                     p++;
diff --git a/libavcodec/mpegaudiodec.c b/libavcodec/mpegaudiodec.c
index fab487e..080da05 100644
--- a/libavcodec/mpegaudiodec.c
+++ b/libavcodec/mpegaudiodec.c
@@ -2277,7 +2277,7 @@ retry:
         goto retry;
     }
 
-    if (ff_mpegaudio_decode_header(s, header) == 1) {
+    if (ff_mpegaudio_decode_header((MPADecodeHeader *)s, header) == 1) {
         /* free format: prepare to compute frame size */
         s->frame_size = -1;
         return -1;
@@ -2342,7 +2342,7 @@ static int decode_frame_adu(AVCodecContext * avctx,
         return buf_size;
     }
 
-    ff_mpegaudio_decode_header(s, header);
+    ff_mpegaudio_decode_header((MPADecodeHeader *)s, header);
     /* update codec info */
     avctx->sample_rate = s->sample_rate;
     avctx->channels = s->nb_channels;
@@ -2491,7 +2491,7 @@ static int decode_frame_mp3on4(AVCodecContext * avctx,
         if (ff_mpa_check_header(header) < 0) // Bad header, discard block
             break;
 
-        ff_mpegaudio_decode_header(m, header);
+        ff_mpegaudio_decode_header((MPADecodeHeader *)m, header);
         out_size += mp_decode_frame(m, outptr, buf, fsize);
         buf += fsize;
         len -= fsize;
diff --git a/libavcodec/mpegaudiodecheader.c b/libavcodec/mpegaudiodecheader.c
index 2e5168e..807d624 100644
--- a/libavcodec/mpegaudiodecheader.c
+++ b/libavcodec/mpegaudiodecheader.c
@@ -30,7 +30,7 @@
 #include "mpegaudiodata.h"
 
 
-int ff_mpegaudio_decode_header(MPADecodeContext *s, uint32_t header)
+int ff_mpegaudio_decode_header(MPADecodeHeader *s, uint32_t header)
 {
     int sample_rate, frame_size, mpeg25, padding;
     int sample_rate_index, bitrate_index;
diff --git a/libavcodec/mpegaudiodecheader.h b/libavcodec/mpegaudiodecheader.h
index 0975781..b7775f6 100644
--- a/libavcodec/mpegaudiodecheader.h
+++ b/libavcodec/mpegaudiodecheader.h
@@ -34,6 +34,6 @@
 /* header decoding. MUST check the header before because no
    consistency check is done there. Return 1 if free format found and
    that the frame size must be computed externally */
-int ff_mpegaudio_decode_header(MPADecodeContext *s, uint32_t header);
+int ff_mpegaudio_decode_header(MPADecodeHeader *s, uint32_t header);
 
 #endif /* AVCODEC_MPEGAUDIODECHEADER_H */
diff --git a/libavformat/mp3.c b/libavformat/mp3.c
index eca9ad4..8602206 100644
--- a/libavformat/mp3.c
+++ b/libavformat/mp3.c
@@ -395,7 +395,7 @@ static int mp3_parse_vbr_tags(AVFormatContext *s, AVStream *st, int64_t base)
     uint32_t v, spf;
     int frames = -1; /* Total number of frames in file */
     const int64_t xing_offtbl[2][2] = {{32, 17}, {17,9}};
-    MPADecodeContext c;
+    MPADecodeHeader c;
     int vbrtag_size = 0;
 
     v = get_be32(s->pb);

-- 
Libav/FFmpeg packaging