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, 13 insertions, 11 deletions

diff --git a/main/app.c b/main/app.c
index 0054c4cde..27858a29b 100644
--- a/main/app.c
+++ b/main/app.c
@@ -445,15 +445,15 @@ struct linear_state {
 	int fd;
 	int autoclose;
 	int allowoverride;
-	int origwfmt;
+	struct ast_format origwfmt;
 };
 
 static void linear_release(struct ast_channel *chan, void *params)
 {
 	struct linear_state *ls = params;
 
-	if (ls->origwfmt && ast_set_write_format(chan, ls->origwfmt)) {
-		ast_log(LOG_WARNING, "Unable to restore channel '%s' to format '%d'\n", chan->name, ls->origwfmt);
+	if (ls->origwfmt.id && ast_set_write_format(chan, &ls->origwfmt)) {
+		ast_log(LOG_WARNING, "Unable to restore channel '%s' to format '%d'\n", chan->name, ls->origwfmt.id);
 	}
 
 	if (ls->autoclose) {
@@ -469,12 +469,13 @@ static int linear_generator(struct ast_channel *chan, void *data, int len, int s
 	struct linear_state *ls = data;
 	struct ast_frame f = {
 		.frametype = AST_FRAME_VOICE,
-		.subclass.codec = AST_FORMAT_SLINEAR,
 		.data.ptr = buf + AST_FRIENDLY_OFFSET / 2,
 		.offset = AST_FRIENDLY_OFFSET,
 	};
 	int res;
 
+	ast_format_set(&f.subclass.format, AST_FORMAT_SLINEAR, 0);
+
 	len = samples * 2;
 	if (len > sizeof(buf) - AST_FRIENDLY_OFFSET) {
 		ast_log(LOG_WARNING, "Can't generate %d bytes of data!\n" , len);
@@ -507,9 +508,9 @@ static void *linear_alloc(struct ast_channel *chan, void *params)
 		ast_clear_flag(chan, AST_FLAG_WRITE_INT);
 	}
 
-	ls->origwfmt = chan->writeformat;
+	ast_format_copy(&ls->origwfmt, &chan->writeformat);
 
-	if (ast_set_write_format(chan, AST_FORMAT_SLINEAR)) {
+	if (ast_set_write_format_by_id(chan, AST_FORMAT_SLINEAR)) {
 		ast_log(LOG_WARNING, "Unable to set '%s' to linear format (write)\n", chan->name);
 		ast_free(ls);
 		ls = params = NULL;
@@ -741,10 +742,11 @@ static int __ast_play_and_record(struct ast_channel *chan, const char *playfile,
 	int totalsilence = 0;
 	int dspsilence = 0;
 	int olddspsilence = 0;
-	int rfmt = 0;
+	struct ast_format rfmt;
 	struct ast_silence_generator *silgen = NULL;
 	char prependfile[80];
 
+	ast_format_clear(&rfmt);
 	if (silencethreshold < 0) {
 		silencethreshold = global_silence_threshold;
 	}
@@ -815,8 +817,8 @@ static int __ast_play_and_record(struct ast_channel *chan, const char *playfile,
 			return -1;
 		}
 		ast_dsp_set_threshold(sildet, silencethreshold);
-		rfmt = chan->readformat;
-		res = ast_set_read_format(chan, AST_FORMAT_SLINEAR);
+		ast_format_copy(&rfmt, &chan->readformat);
+		res = ast_set_read_format_by_id(chan, AST_FORMAT_SLINEAR);
 		if (res < 0) {
 			ast_log(LOG_WARNING, "Unable to set to linear mode, giving up\n");
 			ast_dsp_free(sildet);
@@ -1005,8 +1007,8 @@ static int __ast_play_and_record(struct ast_channel *chan, const char *playfile,
 			ast_filedelete(prependfile, sfmt[x]);
 		}
 	}
-	if (rfmt && ast_set_read_format(chan, rfmt)) {
-		ast_log(LOG_WARNING, "Unable to restore format %s to channel '%s'\n", ast_getformatname(rfmt), chan->name);
+	if (rfmt.id && ast_set_read_format(chan, &rfmt)) {
+		ast_log(LOG_WARNING, "Unable to restore format %s to channel '%s'\n", ast_getformatname(&rfmt), chan->name);
 	}
 	if (outmsg == 2) {
 		ast_stream_and_wait(chan, "auth-thankyou", "");