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: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 14 insertions, 14 deletions

diff --git a/formats/format_wav.c b/formats/format_wav.c
index 91a309ef4..74c14f21c 100644
--- a/formats/format_wav.c
+++ b/formats/format_wav.c
@@ -319,7 +319,7 @@ static int wav_open(struct ast_filestream *s)
 	   if we did, it would go here.  We also might want to check
 	   and be sure it's a valid file.  */
 	struct wav_desc *tmp = (struct wav_desc *)s->_private;
-	if ((tmp->maxlen = check_header(s->f, (s->fmt->format == AST_FORMAT_SLINEAR16 ? 16000 : 8000))) < 0)
+	if ((tmp->maxlen = check_header(s->f, (s->fmt->format.id == AST_FORMAT_SLINEAR16 ? 16000 : 8000))) < 0)
 		return -1;
 	return 0;
 }
@@ -331,7 +331,7 @@ static int wav_rewrite(struct ast_filestream *s, const char *comment)
 	   and be sure it's a valid file.  */
 
 	struct wav_desc *tmp = (struct wav_desc *)s->_private;
-	tmp->hz = (s->fmt->format == AST_FORMAT_SLINEAR16 ? 16000 : 8000);
+	tmp->hz = (s->fmt->format.id == AST_FORMAT_SLINEAR16 ? 16000 : 8000);
 	if (write_header(s->f,tmp->hz))
 		return -1;
 	return 0;
@@ -376,7 +376,7 @@ static struct ast_frame *wav_read(struct ast_filestream *s, int *whennext)
 		bytes = 0;
 /* 	ast_debug(1, "here: %d, maxlen: %d, bytes: %d\n", here, s->maxlen, bytes); */
 	s->fr.frametype = AST_FRAME_VOICE;
-	s->fr.subclass.codec = (fs->hz == 16000 ? AST_FORMAT_SLINEAR16 : AST_FORMAT_SLINEAR);
+	ast_format_set(&s->fr.subclass.format, (fs->hz == 16000 ? AST_FORMAT_SLINEAR16 : AST_FORMAT_SLINEAR), 0);
 	s->fr.mallocd = 0;
 	AST_FRAME_SET_BUFFER(&s->fr, s->buf, AST_FRIENDLY_OFFSET, bytes);
 	
@@ -412,11 +412,11 @@ static int wav_write(struct ast_filestream *fs, struct ast_frame *f)
 		ast_log(LOG_WARNING, "Asked to write non-voice frame!\n");
 		return -1;
 	}
-	if ((f->subclass.codec != AST_FORMAT_SLINEAR) && (f->subclass.codec != AST_FORMAT_SLINEAR16)) {
-		ast_log(LOG_WARNING, "Asked to write non-SLINEAR%s frame (%s)!\n", s->hz == 16000 ? "16" : "", ast_getformatname(f->subclass.codec));
+	if ((f->subclass.format.id != AST_FORMAT_SLINEAR) && (f->subclass.format.id != AST_FORMAT_SLINEAR16)) {
+		ast_log(LOG_WARNING, "Asked to write non-SLINEAR%s frame (%s)!\n", s->hz == 16000 ? "16" : "", ast_getformatname(&f->subclass.format));
 		return -1;
 	}
-	if (f->subclass.codec != fs->fmt->format) {
+	if (ast_format_cmp(&f->subclass.format, &fs->fmt->format) == AST_FORMAT_CMP_NOT_EQUAL) {
 		ast_log(LOG_WARNING, "Can't change SLINEAR frequency during write\n");
 		return -1;
 	}
@@ -486,10 +486,9 @@ static off_t wav_tell(struct ast_filestream *fs)
 	return (offset - 44)/2;
 }
 
-static const struct ast_format wav16_f = {
+static struct ast_format_def wav16_f = {
 	.name = "wav16",
 	.exts = "wav16",
-	.format = AST_FORMAT_SLINEAR16,
 	.open =	wav_open,
 	.rewrite = wav_rewrite,
 	.write = wav_write,
@@ -502,10 +501,9 @@ static const struct ast_format wav16_f = {
 	.desc_size = sizeof(struct wav_desc),
 };
 
-static const struct ast_format wav_f = {
+static struct ast_format_def wav_f = {
 	.name = "wav",
 	.exts = "wav",
-	.format = AST_FORMAT_SLINEAR,
 	.open =	wav_open,
 	.rewrite = wav_rewrite,
 	.write = wav_write,
@@ -520,16 +518,18 @@ static const struct ast_format wav_f = {
 
 static int load_module(void)
 {
-	if (ast_format_register(&wav_f)
-		|| ast_format_register(&wav16_f))
+	ast_format_set(&wav_f.format, AST_FORMAT_SLINEAR, 0);
+	ast_format_set(&wav16_f.format, AST_FORMAT_SLINEAR16, 0);
+	if (ast_format_def_register(&wav_f)
+		|| ast_format_def_register(&wav16_f))
 		return AST_MODULE_LOAD_FAILURE;
 	return AST_MODULE_LOAD_SUCCESS;
 }
 
 static int unload_module(void)
 {
-	return ast_format_unregister(wav_f.name)
-		|| ast_format_unregister(wav16_f.name);
+	return ast_format_def_unregister(wav_f.name)
+		|| ast_format_def_unregister(wav16_f.name);
 }
 
 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Microsoft WAV/WAV16 format (8kHz/16kHz Signed Linear)",