[asterisk-commits] file: branch group/pimp_my_sip r379702 - /team/group/pimp_my_sip/res/

[SVN commits to the Asterisk project]

Author: file
Date: Mon Jan 21 11:01:00 2013
New Revision: 379702

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=379702
Log:
Don't apply joint format information to a non-existent channel when processing an offer, this will get done elsewhere.

Modified:
    team/group/pimp_my_sip/res/res_sip_sdp_audio.c

Modified: team/group/pimp_my_sip/res/res_sip_sdp_audio.c
URL: http://svnview.digium.com/svn/asterisk/team/group/pimp_my_sip/res/res_sip_sdp_audio.c?view=diff&rev=379702&r1=379701&r2=379702
==============================================================================
--- team/group/pimp_my_sip/res/res_sip_sdp_audio.c (original)
+++ team/group/pimp_my_sip/res/res_sip_sdp_audio.c Mon Jan 21 11:01:00 2013
@@ -92,7 +92,7 @@
 	struct ast_rtp_codecs codecs;
 	const pjmedia_sdp_attr *attr;
 	struct ast_format_cap *cap, *jointcap = NULL, *peercap = NULL;
-	struct ast_format fmt;
+//	struct ast_format fmt;
 
 	/* If the stream has been rejected stop media if active */
 	if (!stream->desc.port) {
@@ -170,15 +170,15 @@
 	ast_rtp_codecs_payloads_copy(&codecs, ast_rtp_instance_get_codecs(session->media.audio), session->media.audio);
 
 	/* Now that we have joint formats for audio remove the existing ones from the channel and add the new ones */
-	ast_format_cap_copy(cap, ast_channel_nativeformats(session->channel));
-	ast_format_cap_remove_bytype(cap, AST_FORMAT_TYPE_AUDIO);
-	ast_format_cap_append(cap, jointcap);
+//	ast_format_cap_copy(cap, ast_channel_nativeformats(session->channel));
+//	ast_format_cap_remove_bytype(cap, AST_FORMAT_TYPE_AUDIO);
+//	ast_format_cap_append(cap, jointcap);
 
 	/* Apply the new formats to the channel, potentially changing read/write formats while doing so */
-	ast_format_cap_copy(ast_channel_nativeformats(session->channel), cap);
-	ast_codec_choose(&session->endpoint->prefs, cap, 1, &fmt);
-	ast_set_read_format(session->channel, &fmt);
-	ast_set_write_format(session->channel, &fmt);
+//	ast_format_cap_copy(ast_channel_nativeformats(session->channel), cap);
+//	ast_codec_choose(&session->endpoint->prefs, cap, 1, &fmt);
+//	ast_set_read_format(session->channel, &fmt);
+//	ast_set_write_format(session->channel, &fmt);
 
 cleanup:
 	ast_format_cap_destroy(peercap);