From b5f09d31c21d1e89f7c4693f28046d3b26209da3 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Reimar=20D=C3=B6ffinger?= <Reimar.Doeffinger@gmx.de>
Date: Sun, 6 Sep 2009 09:15:07 +0000
Subject: [PATCH] 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
---
libavcodec/aac.c | 2 +-
libavcodec/aacenc.c | 2 +-
libavcodec/ac3enc.c | 4 ++--
libavcodec/adpcm.c | 2 +-
libavcodec/adxenc.c | 2 +-
libavcodec/flacenc.c | 2 +-
libavcodec/g726.c | 2 +-
libavcodec/libfaac.c | 2 +-
libavcodec/libgsm.c | 4 ++--
libavcodec/libmp3lame.c | 2 +-
libavcodec/libopencore-amr.c | 2 +-
libavcodec/libvorbis.c | 2 +-
libavcodec/mpegaudioenc.c | 2 +-
libavcodec/pcm-mpeg.c | 2 +-
libavcodec/pcm.c | 4 ++--
libavcodec/roqaudioenc.c | 2 +-
libavcodec/vorbis_enc.c | 2 +-
libavcodec/wmaenc.c | 4 ++--
18 files changed, 22 insertions(+), 22 deletions(-)
diff --git a/libavcodec/aac.c b/libavcodec/aac.c
index 8f9249d7f33..fb3c212b706 100644
--- 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 dfa519059c9..ce102b6e3b1 100644
--- 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 5384fdc4c96..9b3e00e6e42 100644
--- 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 d8908d34d26..13f20b42938 100644
--- 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 7c5a95d564b..04e9b789b54 100644
--- 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 164e0c03240..81a35a0d544 100644
--- 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 3004041b08b..a4b20998329 100644
--- 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 cd4d38f5ebb..18b056b04f5 100644
--- 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 7dc2af84059..a9ee7d040da 100644
--- 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 c7d529163d0..62f6c02179f 100644
--- 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 1544db7bed1..8aee7c58170 100644
--- 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 11d9a28636f..530db5c14ca 100644
--- 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 f48d4c8ba66..aefdbc6ba1d 100644
--- 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 4612ac2371f..eed36cc03f6 100644
--- 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 8530cd689a2..a0ce9f5783e 100644
--- 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 5e5dde73fe1..9cff80c0aa0 100644
--- 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 b6533aca534..78302aaeab8 100644
--- 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 83869830146..a1da2830e56 100644
--- 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"),
};
--
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