Make sample_fmts and channel_layouts compound literals const to reduce size of

.data section.

Originally committed as revision 19787 to svn://svn.ffmpeg.org/ffmpeg/trunk

diff --git a/libavcodec/aac.c b/libavcodec/aac.c
index 8f9249d..fb3c212 100644 (file)
--- a/libavcodec/aac.c
+++ b/libavcodec/aac.c
@@ -1804,7 +1804,7 @@ AVCodec aac_decoder = {
     aac_decode_close,
     aac_decode_frame,
     .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
-    .sample_fmts = (enum SampleFormat[]) {
+    .sample_fmts = (const enum SampleFormat[]) {
         SAMPLE_FMT_S16,SAMPLE_FMT_NONE
     },
 };

diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c
index dfa5190..ce102b6 100644 (file)
--- a/libavcodec/aacenc.c
+++ b/libavcodec/aacenc.c
@@ -636,6 +636,6 @@ AVCodec aac_encoder = {
     aac_encode_frame,
     aac_encode_end,
     .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
-    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
+    .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
     .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
 };

diff --git a/libavcodec/ac3enc.c b/libavcodec/ac3enc.c
index 5384fdc..9b3e00e 100644 (file)
--- a/libavcodec/ac3enc.c
+++ b/libavcodec/ac3enc.c
@@ -1400,9 +1400,9 @@ AVCodec ac3_encoder = {
     AC3_encode_frame,
     AC3_encode_close,
     NULL,
-    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
+    .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
     .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
-    .channel_layouts = (int64_t[]){
+    .channel_layouts = (const int64_t[]){
         CH_LAYOUT_MONO,
         CH_LAYOUT_STEREO,
         CH_LAYOUT_2_1,

diff --git a/libavcodec/adpcm.c b/libavcodec/adpcm.c
index d8908d3..13f20b4 100644 (file)
--- a/libavcodec/adpcm.c
+++ b/libavcodec/adpcm.c
@@ -1631,7 +1631,7 @@ AVCodec name ## _encoder = {                    \
     adpcm_encode_frame,                         \
     adpcm_encode_close,                         \
     NULL,                                       \
-    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, \
+    .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, \
     .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
 };
 #else

diff --git a/libavcodec/adxenc.c b/libavcodec/adxenc.c
index 7c5a95d..04e9b78 100644 (file)
--- a/libavcodec/adxenc.c
+++ b/libavcodec/adxenc.c
@@ -192,6 +192,6 @@ AVCodec adpcm_adx_encoder = {
     adx_encode_frame,
     adx_encode_close,
     NULL,
-    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
+    .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
     .long_name = NULL_IF_CONFIG_SMALL("SEGA CRI ADX ADPCM"),
 };

diff --git a/libavcodec/flacenc.c b/libavcodec/flacenc.c
index 164e0c0..81a35a0 100644 (file)
--- a/libavcodec/flacenc.c
+++ b/libavcodec/flacenc.c
@@ -1324,6 +1324,6 @@ AVCodec flac_encoder = {
     flac_encode_close,
     NULL,
     .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
-    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
+    .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
     .long_name = NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"),
 };

diff --git a/libavcodec/g726.c b/libavcodec/g726.c
index 3004041..a4b2099 100644 (file)
--- a/libavcodec/g726.c
+++ b/libavcodec/g726.c
@@ -394,7 +394,7 @@ AVCodec adpcm_g726_encoder = {
     g726_encode_frame,
     g726_close,
     NULL,
-    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
+    .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
     .long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
 };
 #endif

diff --git a/libavcodec/libfaac.c b/libavcodec/libfaac.c
index cd4d38f..18b056b 100644 (file)
--- a/libavcodec/libfaac.c
+++ b/libavcodec/libfaac.c
@@ -153,6 +153,6 @@ AVCodec libfaac_encoder = {
     Faac_encode_init,
     Faac_encode_frame,
     Faac_encode_close,
-    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
+    .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
     .long_name = NULL_IF_CONFIG_SMALL("libfaac AAC (Advanced Audio Codec)"),
 };

diff --git a/libavcodec/libgsm.c b/libavcodec/libgsm.c
index 7dc2af8..a9ee7d0 100644 (file)
--- a/libavcodec/libgsm.c
+++ b/libavcodec/libgsm.c
@@ -120,7 +120,7 @@ AVCodec libgsm_encoder = {
     libgsm_init,
     libgsm_encode_frame,
     libgsm_close,
-    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
+    .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
     .long_name = NULL_IF_CONFIG_SMALL("libgsm GSM"),
 };
 
@@ -132,7 +132,7 @@ AVCodec libgsm_ms_encoder = {
     libgsm_init,
     libgsm_encode_frame,
     libgsm_close,
-    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
+    .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
     .long_name = NULL_IF_CONFIG_SMALL("libgsm GSM Microsoft variant"),
 };
 

diff --git a/libavcodec/libmp3lame.c b/libavcodec/libmp3lame.c
index c7d5291..62f6c02 100644 (file)
--- a/libavcodec/libmp3lame.c
+++ b/libavcodec/libmp3lame.c
@@ -223,6 +223,6 @@ AVCodec libmp3lame_encoder = {
     MP3lame_encode_frame,
     MP3lame_encode_close,
     .capabilities= CODEC_CAP_DELAY,
-    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
+    .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
     .long_name= NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
 };

diff --git a/libavcodec/libopencore-amr.c b/libavcodec/libopencore-amr.c
index 1544db7..8aee7c5 100644 (file)
--- a/libavcodec/libopencore-amr.c
+++ b/libavcodec/libopencore-amr.c
@@ -222,7 +222,7 @@ AVCodec libopencore_amrnb_encoder = {
     amr_nb_encode_frame,
     amr_nb_encode_close,
     NULL,
-    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
+    .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
     .long_name = NULL_IF_CONFIG_SMALL("OpenCORE Adaptive Multi-Rate (AMR) Narrow-Band"),
 };
 

diff --git a/libavcodec/libvorbis.c b/libavcodec/libvorbis.c
index 11d9a28..530db5c 100644 (file)
--- a/libavcodec/libvorbis.c
+++ b/libavcodec/libvorbis.c
@@ -224,6 +224,6 @@ AVCodec libvorbis_encoder = {
     oggvorbis_encode_frame,
     oggvorbis_encode_close,
     .capabilities= CODEC_CAP_DELAY,
-    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
+    .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
     .long_name= NULL_IF_CONFIG_SMALL("libvorbis Vorbis"),
 } ;

diff --git a/libavcodec/mpegaudioenc.c b/libavcodec/mpegaudioenc.c
index f48d4c8..aefdbc6 100644 (file)
--- a/libavcodec/mpegaudioenc.c
+++ b/libavcodec/mpegaudioenc.c
@@ -797,7 +797,7 @@ AVCodec mp2_encoder = {
     MPA_encode_frame,
     MPA_encode_close,
     NULL,
-    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
+    .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
     .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
 };
 

diff --git a/libavcodec/pcm-mpeg.c b/libavcodec/pcm-mpeg.c
index 4612ac2..eed36cc 100644 (file)
--- a/libavcodec/pcm-mpeg.c
+++ b/libavcodec/pcm-mpeg.c
@@ -310,7 +310,7 @@ AVCodec pcm_bluray_decoder = {
     NULL,
     NULL,
     pcm_bluray_decode_frame,
-    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16, SAMPLE_FMT_S32,
+    .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16, SAMPLE_FMT_S32,
                                          SAMPLE_FMT_NONE},
     .long_name = NULL_IF_CONFIG_SMALL("PCM signed 16|20|24-bit big-endian for Blu-ray media"),
 };

diff --git a/libavcodec/pcm.c b/libavcodec/pcm.c
index 8530cd6..a0ce9f5 100644 (file)
--- a/libavcodec/pcm.c
+++ b/libavcodec/pcm.c
@@ -523,7 +523,7 @@ AVCodec name ## _encoder = {                    \
     pcm_encode_frame,                           \
     pcm_encode_close,                           \
     NULL,                                       \
-    .sample_fmts = (enum SampleFormat[]){sample_fmt_,SAMPLE_FMT_NONE}, \
+    .sample_fmts = (const enum SampleFormat[]){sample_fmt_,SAMPLE_FMT_NONE}, \
     .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
 };
 #else
@@ -541,7 +541,7 @@ AVCodec name ## _decoder = {                    \
     NULL,                                       \
     NULL,                                       \
     pcm_decode_frame,                           \
-    .sample_fmts = (enum SampleFormat[]){sample_fmt_,SAMPLE_FMT_NONE}, \
+    .sample_fmts = (const enum SampleFormat[]){sample_fmt_,SAMPLE_FMT_NONE}, \
     .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
 };
 #else

diff --git a/libavcodec/roqaudioenc.c b/libavcodec/roqaudioenc.c
index 5e5dde7..9cff80c 100644 (file)
--- a/libavcodec/roqaudioenc.c
+++ b/libavcodec/roqaudioenc.c
@@ -174,6 +174,6 @@ AVCodec roq_dpcm_encoder = {
     roq_dpcm_encode_frame,
     roq_dpcm_encode_close,
     NULL,
-    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
+    .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
     .long_name = NULL_IF_CONFIG_SMALL("id RoQ DPCM"),
 };

diff --git a/libavcodec/vorbis_enc.c b/libavcodec/vorbis_enc.c
index b6533ac..78302aa 100644 (file)
--- a/libavcodec/vorbis_enc.c
+++ b/libavcodec/vorbis_enc.c
@@ -1045,6 +1045,6 @@ AVCodec vorbis_encoder = {
     vorbis_encode_frame,
     vorbis_encode_close,
     .capabilities= CODEC_CAP_DELAY,
-    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
+    .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
     .long_name = NULL_IF_CONFIG_SMALL("Vorbis"),
 };

diff --git a/libavcodec/wmaenc.c b/libavcodec/wmaenc.c
index 8386983..a1da283 100644 (file)
--- a/libavcodec/wmaenc.c
+++ b/libavcodec/wmaenc.c
@@ -392,7 +392,7 @@ AVCodec wmav1_encoder =
     encode_init,
     encode_superframe,
     ff_wma_end,
-    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
+    .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
     .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 1"),
 };
 
@@ -405,6 +405,6 @@ AVCodec wmav2_encoder =
     encode_init,
     encode_superframe,
     ff_wma_end,
-    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
+    .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
     .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 2"),
 };