[svn-commits] jrose: trunk r310588 - in /trunk: ./ funcs/func_volume.c

[SVN commits to the Digium repositories]

Author: jrose
Date: Mon Mar 14 10:40:43 2011
New Revision: 310588

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=310588
Log:
Merged revisions 310587 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r310587 | jrose | 2011-03-14 10:27:57 -0500 (Mon, 14 Mar 2011) | 15 lines
  
  Merged revisions 310585 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r310585 | jrose | 2011-03-14 08:56:22 -0500 (Mon, 14 Mar 2011) | 8 lines
    
    Adds 'p' as an option to func_volume.  When it is on, the old behavior with DTMF controlling volume adjustment will be enforced.
    When it is off, DTMF will not be processed by the function.
    
    Programmed by Jonathan Rose
    Reviewed by David Vossel, Leif Madsen, and Russell Bryant
    
    http://reviewboard.digium.internal/r/93/
  ........
................

Modified:
    trunk/   (props changed)
    trunk/funcs/func_volume.c

Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.8-merged' - no diff available.

Modified: trunk/funcs/func_volume.c
URL: http://svnview.digium.com/svn/asterisk/trunk/funcs/func_volume.c?view=diff&rev=310588&r1=310587&r2=310588
==============================================================================
--- trunk/funcs/func_volume.c (original)
+++ trunk/funcs/func_volume.c Mon Mar 14 10:40:43 2011
@@ -1,7 +1,7 @@
 /*
  * Asterisk -- An open source telephony toolkit.
  *
- * Copyright (C) 2007, Digium, Inc.
+ * Copyright (C) 2011, Digium, Inc.
  *
  * Joshua Colp <jcolp at digium.com> 
  *
@@ -35,6 +35,7 @@
 #include "asterisk/pbx.h"
 #include "asterisk/utils.h"
 #include "asterisk/audiohook.h"
+#include "asterisk/app.h"
 
 /*** DOCUMENTATION
 	<function name="VOLUME" language="en_US">
@@ -45,6 +46,13 @@
 			<parameter name="direction" required="true">
 				<para>Must be <literal>TX</literal> or <literal>RX</literal>.</para>
 			</parameter>
+			<parameter name="options">
+				<optionlist>
+					<option name="p">
+						<para>Enable DTMF volume control</para>
+					</option>
+				</optionlist>
+			</parameter>
 		</syntax>
 		<description>
 			<para>The VOLUME function can be used to increase or decrease the <literal>tx</literal> or
@@ -52,6 +60,8 @@
 			<para>For example:</para>
 			<para>Set(VOLUME(TX)=3)</para>
 			<para>Set(VOLUME(RX)=2)</para>
+			<para>Set(VOLUME(TX,p)=3)</para>
+			<para>Set(VOLUME(RX,p)=3></para>
 		</description>
 	</function>
  ***/
@@ -60,7 +70,16 @@
 	struct ast_audiohook audiohook;
 	int tx_gain;
 	int rx_gain;
-};
+	unsigned int flags;
+};
+
+enum volume_flags {
+	VOLUMEFLAG_CHANGE = (1 << 1),
+};
+
+AST_APP_OPTIONS(volume_opts, {
+	AST_APP_OPTION('p', VOLUMEFLAG_CHANGE),
+});
 
 static void destroy_callback(void *data)
 {
@@ -96,18 +115,23 @@
 	vi = datastore->data;
 
 	/* If this is DTMF then allow them to increase/decrease the gains */
-	if (frame->frametype == AST_FRAME_DTMF) {
-		/* Only use DTMF coming from the source... not going to it */
-		if (direction != AST_AUDIOHOOK_DIRECTION_READ)
-			return 0;
-		if (frame->subclass.integer == '*') {
-			vi->tx_gain += 1;
-			vi->rx_gain += 1;
-		} else if (frame->subclass.integer == '#') {
-			vi->tx_gain -= 1;
-			vi->rx_gain -= 1;
+	if (ast_test_flag(vi, VOLUMEFLAG_CHANGE)) {
+		if (frame->frametype == AST_FRAME_DTMF) {
+			/* Only use DTMF coming from the source... not going to it */
+			if (direction != AST_AUDIOHOOK_DIRECTION_READ)
+				return 0; 
+			if (frame->subclass.integer == '*') {
+				vi->tx_gain += 1;
+				vi->rx_gain += 1;
+			} else if (frame->subclass.integer == '#') {
+				vi->tx_gain -= 1;
+				vi->rx_gain -= 1;
+			}
 		}
-	} else if (frame->frametype == AST_FRAME_VOICE) {
+	}
+
+	
+	if (frame->frametype == AST_FRAME_VOICE) {
 		/* Based on direction of frame grab the gain, and confirm it is applicable */
 		if (!(gain = (direction == AST_AUDIOHOOK_DIRECTION_READ) ? &vi->rx_gain : &vi->tx_gain) || !*gain)
 			return 0;
@@ -124,7 +148,18 @@
 	struct volume_information *vi = NULL;
 	int is_new = 0;
 
+	/* Separate options from argument */
+	
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(direction);
+		AST_APP_ARG(options);
+	);
+	
+	AST_STANDARD_APP_ARGS(args, data);
+
+	ast_channel_lock(chan);
 	if (!(datastore = ast_channel_datastore_find(chan, &volume_datastore, NULL))) {
+		ast_channel_unlock(chan);
 		/* Allocate a new datastore to hold the reference to this volume and audiohook information */
 		if (!(datastore = ast_datastore_alloc(&volume_datastore, NULL)))
 			return 0;
@@ -137,19 +172,40 @@
 		ast_set_flag(&vi->audiohook, AST_AUDIOHOOK_WANTS_DTMF);
 		is_new = 1;
 	} else {
+		ast_channel_unlock(chan);
 		vi = datastore->data;
 	}
 
 	/* Adjust gain on volume information structure */
-	if (!strcasecmp(data, "tx"))
-		vi->tx_gain = atoi(value);
-	else if (!strcasecmp(data, "rx"))
+	if (ast_strlen_zero(args.direction)) {
+		ast_log(LOG_ERROR, "Direction must be specified for VOLUME function\n");
+		return -1;
+	}
+
+	if (!strcasecmp(args.direction, "tx")) { 
+		vi->tx_gain = atoi(value); 
+	} else if (!strcasecmp(args.direction, "rx")) {
 		vi->rx_gain = atoi(value);
+	} else {
+		ast_log(LOG_ERROR, "Direction must be either RX or TX\n");
+	}
 
 	if (is_new) {
 		datastore->data = vi;
+		ast_channel_lock(chan);
 		ast_channel_datastore_add(chan, datastore);
+		ast_channel_unlock(chan);
 		ast_audiohook_attach(chan, &vi->audiohook);
+	}
+
+	/* Add Option data to struct */
+	
+	if (!ast_strlen_zero(args.options)) {
+		struct ast_flags flags = { 0 };
+		ast_app_parse_options(volume_opts, &flags, &data, args.options);
+		vi->flags = flags.flags;
+	} else { 
+		vi->flags = 0; 
 	}
 
 	return 0;