Avoid allocating MPADecodeContext on stack.

Instead move relevant fields into MPADecodeHeader and use it
where appropriate.

git-svn-id: file:///var/local/repositories/ffmpeg/trunk@16728 9553f0bf-9b14-0410-a0b8-cfaf0461ba5b

diff --git a/libavcodec/mpegaudio.h b/libavcodec/mpegaudio.h
index 96bea310a8163389402b6431efa031848b6547c2..f32e6abf2ee23ec2fddc28f81ae655e9dbd9544e 100644 (file)
--- 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 b06b15a56df91c4d59374121aac13fff330ff4b2..646da8b67d7e804bad25cbf42434560726d654b4 100644 (file)
--- 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 fab487edc7e0ad5680f56813ece0b2a0cff8b074..080da058edf046701348b122194577158c81058e 100644 (file)
--- 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 2e5168e05d4da2283d8fcda8f65769900827febe..807d6244dcc16e8a1bdabed5413f083f303301d0 100644 (file)
--- 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 09757817ffc02e219047dfbaa379886fc905354a..b7775f6001b4437e49da90e88f912fc4fa3db0eb 100644 (file)
--- 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 eca9ad4240040337864309f3561672481ee5ec2d..8602206bd1f928d4ad9d56f06f0c1f589e5cdc80 100644 (file)
--- 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);