diff options
| author | dvossel <dvossel@f38db490-d61c-443f-a65b-d21fe96a405b> | 2011-02-03 16:22:10 +0000 |
|---|---|---|
| committer | dvossel <dvossel@f38db490-d61c-443f-a65b-d21fe96a405b> | 2011-02-03 16:22:10 +0000 |
| commit | 4aca3187a3db25ff4d2208f116f618b363dec7d5 (patch) | |
| tree | 00da0caa5a07b7b25729f089dbcafb08129fa9be /formats/format_wav_gsm.c | |
| parent | 8170aae0a0882a93ca1ef80736cb95c2d6126865 (diff) | |
Asterisk media architecture conversion - no more format bitfields
This patch is the foundation of an entire new way of looking at media in Asterisk.
The code present in this patch is everything required to complete phase1 of my
Media Architecture proposal. For more information about this project visit the link below.
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal
The primary function of this patch is to convert all the usages of format
bitfields in Asterisk to use the new format and format_cap APIs. Functionally
no change in behavior should be present in this patch. Thanks to twilson
and russell for all the time they spent reviewing these changes.
Review: https://reviewboard.asterisk.org/r/1083/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306010 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'formats/format_wav_gsm.c')
| -rw-r--r-- | formats/format_wav_gsm.c | 14 |
1 files changed, 7 insertions, 7 deletions
diff --git a/formats/format_wav_gsm.c b/formats/format_wav_gsm.c index f864f2814..ae1f11ed8 100644 --- a/formats/format_wav_gsm.c +++ b/formats/format_wav_gsm.c @@ -395,7 +395,7 @@ static struct ast_frame *wav_read(struct ast_filestream *s, int *whennext) struct wavg_desc *fs = (struct wavg_desc *)s->_private; s->fr.frametype = AST_FRAME_VOICE; - s->fr.subclass.codec = AST_FORMAT_GSM; + ast_format_set(&s->fr.subclass.format, AST_FORMAT_GSM, 0); s->fr.offset = AST_FRIENDLY_OFFSET; s->fr.samples = GSM_SAMPLES; s->fr.mallocd = 0; @@ -432,8 +432,8 @@ static int wav_write(struct ast_filestream *s, struct ast_frame *f) ast_log(LOG_WARNING, "Asked to write non-voice frame!\n"); return -1; } - if (f->subclass.codec != AST_FORMAT_GSM) { - ast_log(LOG_WARNING, "Asked to write non-GSM frame (%s)!\n", ast_getformatname(f->subclass.codec)); + if (f->subclass.format.id != AST_FORMAT_GSM) { + ast_log(LOG_WARNING, "Asked to write non-GSM frame (%s)!\n", ast_getformatname(&f->subclass.format)); return -1; } /* XXX this might fail... if the input is a multiple of MSGSM_FRAME_SIZE @@ -521,10 +521,9 @@ static off_t wav_tell(struct ast_filestream *fs) return (offset - MSGSM_DATA_OFFSET)/MSGSM_FRAME_SIZE*MSGSM_SAMPLES; } -static const struct ast_format wav49_f = { +static struct ast_format_def wav49_f = { .name = "wav49", .exts = "WAV|wav49", - .format = AST_FORMAT_GSM, .open = wav_open, .rewrite = wav_rewrite, .write = wav_write, @@ -538,14 +537,15 @@ static const struct ast_format wav49_f = { static int load_module(void) { - if (ast_format_register(&wav49_f)) + ast_format_set(&wav49_f.format, AST_FORMAT_GSM, 0); + if (ast_format_def_register(&wav49_f)) return AST_MODULE_LOAD_FAILURE; return AST_MODULE_LOAD_SUCCESS; } static int unload_module(void) { - return ast_format_unregister(wav49_f.name); + return ast_format_def_unregister(wav49_f.name); } AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Microsoft WAV format (Proprietary GSM)", |