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Code Issues Releases Activity

Make sample_fmts and channel_layouts compound literals const to reduce size of

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.data section.

Originally committed as revision 19787 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Reimar Döffinger 2009-09-06 09:15:07 +00:00
parent 88e70e1b0a
commit b5f09d31c2
18 changed files with 22 additions and 22 deletions
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2
libavcodec/aac.c
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@ -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
},
};

2
libavcodec/aacenc.c
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@ -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"),
};

4
libavcodec/ac3enc.c
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@ -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,

2
libavcodec/adpcm.c
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@ -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

2
libavcodec/adxenc.c
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@ -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"),
};

2
libavcodec/flacenc.c
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@ -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)"),
};

2
libavcodec/g726.c
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@ -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

2
libavcodec/libfaac.c
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@ -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)"),
};

4
libavcodec/libgsm.c
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@ -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"),
};

2
libavcodec/libmp3lame.c
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@ -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)"),
};

2
libavcodec/libopencore-amr.c
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@ -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"),
};

2
libavcodec/libvorbis.c
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@ -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"),
} ;

2
libavcodec/mpegaudioenc.c
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@ -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)"),
};

2
libavcodec/pcm-mpeg.c
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@ -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"),
};

4
libavcodec/pcm.c
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@ -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

2
libavcodec/roqaudioenc.c
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@ -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"),
};

2
libavcodec/vorbis_enc.c
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@ -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"),
};

4
libavcodec/wmaenc.c
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@ -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"),
};