Mercurial > libavcodec.hg
changeset 10145:7955db355703 libavcodec
Make sample_fmts and channel_layouts compound literals const to reduce size of
.data section.
author | reimar |
---|---|
date | Sun, 06 Sep 2009 09:15:07 +0000 |
parents | 50ec8930f99e |
children | 38cfe222e1a4 |
files | aac.c aacenc.c ac3enc.c adpcm.c adxenc.c flacenc.c g726.c libfaac.c libgsm.c libmp3lame.c libopencore-amr.c libvorbis.c mpegaudioenc.c pcm-mpeg.c pcm.c roqaudioenc.c vorbis_enc.c wmaenc.c |
diffstat | 18 files changed, 22 insertions(+), 22 deletions(-) [+] |
line wrap: on
line diff
--- a/aac.c Sun Sep 06 08:56:10 2009 +0000 +++ b/aac.c Sun Sep 06 09:15:07 2009 +0000 @@ -1804,7 +1804,7 @@ 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 }, };
--- a/aacenc.c Sun Sep 06 08:56:10 2009 +0000 +++ b/aacenc.c Sun Sep 06 09:15:07 2009 +0000 @@ -636,6 +636,6 @@ 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"), };
--- a/ac3enc.c Sun Sep 06 08:56:10 2009 +0000 +++ b/ac3enc.c Sun Sep 06 09:15:07 2009 +0000 @@ -1400,9 +1400,9 @@ 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,
--- a/adpcm.c Sun Sep 06 08:56:10 2009 +0000 +++ b/adpcm.c Sun Sep 06 09:15:07 2009 +0000 @@ -1631,7 +1631,7 @@ 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
--- a/adxenc.c Sun Sep 06 08:56:10 2009 +0000 +++ b/adxenc.c Sun Sep 06 09:15:07 2009 +0000 @@ -192,6 +192,6 @@ 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"), };
--- a/flacenc.c Sun Sep 06 08:56:10 2009 +0000 +++ b/flacenc.c Sun Sep 06 09:15:07 2009 +0000 @@ -1324,6 +1324,6 @@ 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)"), };
--- a/g726.c Sun Sep 06 08:56:10 2009 +0000 +++ b/g726.c Sun Sep 06 09:15:07 2009 +0000 @@ -394,7 +394,7 @@ 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
--- a/libfaac.c Sun Sep 06 08:56:10 2009 +0000 +++ b/libfaac.c Sun Sep 06 09:15:07 2009 +0000 @@ -153,6 +153,6 @@ 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)"), };
--- a/libgsm.c Sun Sep 06 08:56:10 2009 +0000 +++ b/libgsm.c Sun Sep 06 09:15:07 2009 +0000 @@ -120,7 +120,7 @@ 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 @@ 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"), };
--- a/libmp3lame.c Sun Sep 06 08:56:10 2009 +0000 +++ b/libmp3lame.c Sun Sep 06 09:15:07 2009 +0000 @@ -223,6 +223,6 @@ 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)"), };
--- a/libopencore-amr.c Sun Sep 06 08:56:10 2009 +0000 +++ b/libopencore-amr.c Sun Sep 06 09:15:07 2009 +0000 @@ -222,7 +222,7 @@ 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"), };
--- a/libvorbis.c Sun Sep 06 08:56:10 2009 +0000 +++ b/libvorbis.c Sun Sep 06 09:15:07 2009 +0000 @@ -224,6 +224,6 @@ 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"), } ;
--- a/mpegaudioenc.c Sun Sep 06 08:56:10 2009 +0000 +++ b/mpegaudioenc.c Sun Sep 06 09:15:07 2009 +0000 @@ -797,7 +797,7 @@ 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)"), };
--- a/pcm-mpeg.c Sun Sep 06 08:56:10 2009 +0000 +++ b/pcm-mpeg.c Sun Sep 06 09:15:07 2009 +0000 @@ -310,7 +310,7 @@ 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"), };
--- a/pcm.c Sun Sep 06 08:56:10 2009 +0000 +++ b/pcm.c Sun Sep 06 09:15:07 2009 +0000 @@ -523,7 +523,7 @@ 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 @@ 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
--- a/roqaudioenc.c Sun Sep 06 08:56:10 2009 +0000 +++ b/roqaudioenc.c Sun Sep 06 09:15:07 2009 +0000 @@ -174,6 +174,6 @@ 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"), };
--- a/vorbis_enc.c Sun Sep 06 08:56:10 2009 +0000 +++ b/vorbis_enc.c Sun Sep 06 09:15:07 2009 +0000 @@ -1045,6 +1045,6 @@ 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"), };
--- a/wmaenc.c Sun Sep 06 08:56:10 2009 +0000 +++ b/wmaenc.c Sun Sep 06 09:15:07 2009 +0000 @@ -392,7 +392,7 @@ 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 @@ 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"), };