[asterisk-commits] rmudgett: trunk r247914 - in /trunk: ./ channels/chan_misdn.c

Fri Feb 19 11:33:37 CST 2010

Author: rmudgett
Date: Fri Feb 19 11:33:33 2010
New Revision: 247914

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=247914
Log:
Merged revisions 247910 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

................
  r247910 | rmudgett | 2010-02-19 11:18:49 -0600 (Fri, 19 Feb 2010) | 55 lines
  
  Merged revision 247904 from
  https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
  
  ..........
  r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri, 19 Feb 2010) | 49 lines
  
  Make chan_misdn DTMF processing consistent with other channel technologies.
  
  The processing of DTMF tones on the receiving side of an ISDN channel is
  inconsistent with the way it is handled in other channels, especially
  DAHDI analog.  This causes DTMF tones sent from an ISDN phone to be
  doubled at the connected party.
  
  We are using the following 2 options of misdn.conf
  1) astdtmf=yes
  2) senddtmf=yes
  
  Option one is necessary because the asterisk DSP DTMF detection is better
  than mISDN's internal DSP.  Not as many false positives.
  
  Option two is necessary to transmit DTMF tones end to end when mISDN
  channels are connected to SIP channels with out of band DTMF for example.
  
  The symptom is that DTMF tones sent by an ISDN phone are doubled on the
  way through asterisk when two mISDN channels are connected with a Local
  channel in between or if it is bridged to an analog channel.
  
  The doubling of DTMF tones is because DTMF is passed inband to asterisk by
  the mISDN channel and passed out of band once again after the release of
  the DTMF tone.  Passing it inband is wrong.  Neither an analog channel nor
  SIP channel passes DTMF inband if configured to inband DTMF.  Analog and
  SIP channels filter out the DTMF tones because they use the voice frames
  returned by ast_dsp_process.  But chan_misdn passes the unfiltered input
  voice frames instead.
  
  To overcome one aspect of the problem, the doubling of DTMF tones when two
  mISDN channels are directly bridged, someone made an 'optimization', where
  in that case the DTMF tone passed out-of-band to the peer channel is not
  translated to an inband tone at the transmit side.  This optimization is
  bad because it does not work in general.  For example, analog channels or
  mISDN channels when bridged through an intermediary local channel will
  generate DTMF tones from out-of-band information.  Also, of course, it
  must not be done when there is no inband DTMF available.
  
  This patch fixes the issue.  Now chan_misdn will filter the received
  inband DTMF signal the same as other channel types.
  
  Another change included: No need to build an extra translation path
  because ast_process_dsp does it if required.
  
  Patches:
  	misdn-dtmf.patch
  
  JIRA ABE-2080
................

Modified:
    trunk/   (props changed)
    trunk/channels/chan_misdn.c

Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.

Modified: trunk/channels/chan_misdn.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_misdn.c?view=diff&rev=247914&r1=247913&r2=247914
==============================================================================

--- trunk/channels/chan_misdn.c (original)
+++ trunk/channels/chan_misdn.c Fri Feb 19 11:33:33 2010
@@ -475,13 +475,6 @@
 	 * \note Must use ast_dsp_free() to clean up.
 	 */
 	struct ast_dsp *dsp;
-
-	/*!
-	 * \brief Allocated audio frame sample translator
-	 * \note ast_translator_build_path() creates the translator path.
-	 * \note Must use ast_translator_free_path() to clean up.
-	 */
-	struct ast_trans_pvt *trans;
 
 	/*!
 	 * \brief Associated Asterisk channel structure.
@@ -5940,9 +5933,6 @@
 		if (ch->dsp) {
 			ast_dsp_set_features(ch->dsp, DSP_FEATURE_DIGIT_DETECT | (ch->faxdetect ? DSP_FEATURE_FAX_DETECT : 0));
 		}
-		if (!ch->trans) {
-			ch->trans = ast_translator_build_path(AST_FORMAT_SLINEAR, AST_FORMAT_ALAW);
-		}
 	}
 
 	/* AOCD initialization */
@@ -6615,12 +6605,6 @@
 		misdn_lib_send_event(bc, EVENT_INFORMATION);
 		break;
 	default:
-		/* Do not send Digits in CONNECTED State, when
-		 * the other side is also mISDN. */
-		if (p->other_ch) {
-			return 0;
-		}
-
 		if (bc->send_dtmf) {
 			send_digit_to_chan(p, digit);
 		}
@@ -6960,21 +6944,17 @@
 
 static struct ast_frame *process_ast_dsp(struct chan_list *tmp, struct ast_frame *frame)
 {
-	struct ast_frame *f,*f2;
-
- 	if (tmp->trans) {
- 		f2 = ast_translate(tmp->trans, frame, 0);
- 		f = ast_dsp_process(tmp->ast, tmp->dsp, f2);
+	struct ast_frame *f;
+
+ 	if (tmp->dsp) {
+ 		f = ast_dsp_process(tmp->ast, tmp->dsp, frame);
  	} else {
-		chan_misdn_log(0, tmp->bc->port, "No T-Path found\n");
+		chan_misdn_log(0, tmp->bc->port, "No DSP-Path found\n");
 		return NULL;
 	}
 
 	if (!f || (f->frametype != AST_FRAME_DTMF)) {
-		if (f) {
-			ast_frfree(f);
-		}
-		return frame;
+		return f;
 	}
 
 	ast_debug(1, "Detected inband DTMF digit: %c\n", f->subclass.integer);
@@ -7971,9 +7951,6 @@
 	if (chan->dsp) {
 		ast_dsp_free(chan->dsp);
 	}
-	if (chan->trans) {
-		ast_translator_free_path(chan->trans);
-	}
 
 	ast_mutex_lock(&cl_te_lock);
 	if (!*list) {
@@ -11915,9 +11892,6 @@
 		if (ch->dsp) {
 			ast_dsp_set_features(ch->dsp, DSP_FEATURE_DIGIT_DETECT | DSP_FEATURE_FAX_DETECT);
 		}
-		if (!ch->trans) {
-			ch->trans = ast_translator_build_path(AST_FORMAT_SLINEAR, AST_FORMAT_ALAW);
-		}
 	}
 
 	if (ch->ast_dsp) {


