[asterisk-commits] file: branch group/pimp_my_sip r388472 - in /team/group/pimp_my_sip: channels...

[SVN commits to the Asterisk project]

Author: file
Date: Fri May 10 17:56:55 2013
New Revision: 388472

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=388472
Log:
Add inband DTMF detection for inbound media.

Review: https://reviewboard.asterisk.org/r/2527/

Modified:
    team/group/pimp_my_sip/channels/chan_gulp.c
    team/group/pimp_my_sip/include/asterisk/res_sip_session.h
    team/group/pimp_my_sip/res/res_sip_sdp_rtp.c
    team/group/pimp_my_sip/res/res_sip_session.c

Modified: team/group/pimp_my_sip/channels/chan_gulp.c
URL: http://svnview.digium.com/svn/asterisk/team/group/pimp_my_sip/channels/chan_gulp.c?view=diff&rev=388472&r1=388471&r2=388472
==============================================================================
--- team/group/pimp_my_sip/channels/chan_gulp.c (original)
+++ team/group/pimp_my_sip/channels/chan_gulp.c Fri May 10 17:56:55 2013
@@ -53,6 +53,7 @@
 #include "asterisk/musiconhold.h"
 #include "asterisk/causes.h"
 #include "asterisk/taskprocessor.h"
+#include "asterisk/dsp.h"
 
 #include "asterisk/res_sip.h"
 #include "asterisk/res_sip_session.h"
@@ -610,6 +611,7 @@
 static struct ast_frame *gulp_read(struct ast_channel *ast)
 {
 	struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
+	struct ast_sip_session *session = pvt->session;
 	struct ast_frame *f;
 	struct ast_sip_session_media *media = NULL;
 	int rtcp = 0;
@@ -636,14 +638,27 @@
 		return &ast_null_frame;
 	}
 
-	f = ast_rtp_instance_read(media->rtp, rtcp);
-
-	if (f && f->frametype == AST_FRAME_VOICE) {
-		if (!(ast_format_cap_iscompatible(ast_channel_nativeformats(ast), &f->subclass.format))) {
-			ast_debug(1, "Oooh, format changed to %s\n", ast_getformatname(&f->subclass.format));
-			ast_format_cap_set(ast_channel_nativeformats(ast), &f->subclass.format);
-			ast_set_read_format(ast, ast_channel_readformat(ast));
-			ast_set_write_format(ast, ast_channel_writeformat(ast));
+	if (!(f = ast_rtp_instance_read(media->rtp, rtcp))) {
+		return f;
+	}
+
+	if (f->frametype != AST_FRAME_VOICE) {
+		return f;
+	}
+
+	if (!(ast_format_cap_iscompatible(ast_channel_nativeformats(ast), &f->subclass.format))) {
+		ast_debug(1, "Oooh, format changed to %s\n", ast_getformatname(&f->subclass.format));
+		ast_format_cap_set(ast_channel_nativeformats(ast), &f->subclass.format);
+		ast_set_read_format(ast, ast_channel_readformat(ast));
+		ast_set_write_format(ast, ast_channel_writeformat(ast));
+	}
+
+	if (session->dsp) {
+		f = ast_dsp_process(ast, session->dsp, f);
+
+		if (f && (f->frametype == AST_FRAME_DTMF)) {
+			ast_debug(3, "* Detected inband DTMF '%c' on '%s'\n", f->subclass.integer,
+				ast_channel_name(ast));
 		}
 	}
 

Modified: team/group/pimp_my_sip/include/asterisk/res_sip_session.h
URL: http://svnview.digium.com/svn/asterisk/team/group/pimp_my_sip/include/asterisk/res_sip_session.h?view=diff&rev=388472&r1=388471&r2=388472
==============================================================================
--- team/group/pimp_my_sip/include/asterisk/res_sip_session.h (original)
+++ team/group/pimp_my_sip/include/asterisk/res_sip_session.h Fri May 10 17:56:55 2013
@@ -43,6 +43,7 @@
 struct pjmedia_sdp_media;
 struct pjmedia_sdp_session;
 struct ast_rtp_instance;
+struct ast_dsp;
 
 struct ast_sip_session_sdp_handler;
 
@@ -107,6 +108,8 @@
 	struct ast_format_cap *req_caps;
 	/* Codecs overriden by dialplan on an outgoing request */
 	struct ast_codec_pref override_prefs;
+	/* Optional DSP, used only for inband DTMF detection if configured */
+	struct ast_dsp *dsp;
 };
 
 typedef int (*ast_sip_session_request_creation_cb)(struct ast_sip_session *session, pjsip_tx_data *tdata);

Modified: team/group/pimp_my_sip/res/res_sip_sdp_rtp.c
URL: http://svnview.digium.com/svn/asterisk/team/group/pimp_my_sip/res/res_sip_sdp_rtp.c?view=diff&rev=388472&r1=388471&r2=388472
==============================================================================
--- team/group/pimp_my_sip/res/res_sip_sdp_rtp.c (original)
+++ team/group/pimp_my_sip/res/res_sip_sdp_rtp.c Fri May 10 17:56:55 2013
@@ -117,6 +117,10 @@
 
 	ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(session_media->rtp),
 					 session_media->rtp, &session->endpoint->prefs);
+
+	if (session->endpoint->dtmf == AST_SIP_DTMF_INBAND) {
+		ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_INBAND);
+	}
 
 	if (!session->endpoint->ice_support && (ice = ast_rtp_instance_get_ice(session_media->rtp))) {
 		ice->stop(session_media->rtp);

Modified: team/group/pimp_my_sip/res/res_sip_session.c
URL: http://svnview.digium.com/svn/asterisk/team/group/pimp_my_sip/res/res_sip_session.c?view=diff&rev=388472&r1=388471&r2=388472
==============================================================================
--- team/group/pimp_my_sip/res/res_sip_session.c (original)
+++ team/group/pimp_my_sip/res/res_sip_session.c Fri May 10 17:56:55 2013
@@ -41,6 +41,7 @@
 #include "asterisk/taskprocessor.h"
 #include "asterisk/causes.h"
 #include "asterisk/sdp_srtp.h"
+#include "asterisk/dsp.h"
 
 #define SDP_HANDLER_BUCKETS 11
 
@@ -928,6 +929,10 @@
 	ast_party_id_free(&session->id);
 	ao2_cleanup(session->endpoint);
 	ast_format_cap_destroy(session->req_caps);
+
+	if (session->dsp) {
+		ast_dsp_free(session->dsp);
+	}
 }
 
 static int add_supplements(struct ast_sip_session *session)
@@ -991,6 +996,14 @@
 	session->endpoint = endpoint;
 	session->inv_session = inv_session;
 	session->req_caps = ast_format_cap_alloc_nolock();
+
+	if (endpoint->dtmf == AST_SIP_DTMF_INBAND) {
+		if (!(session->dsp = ast_dsp_new())) {
+			return NULL;
+		}
+
+		ast_dsp_set_features(session->dsp, DSP_FEATURE_DIGIT_DETECT);
+	}
 
 	if (add_supplements(session)) {
 		return NULL;