Merge "contrib/sip_to_pjsip: handle setvar in conversion"
[asterisk/asterisk.git] / funcs / func_speex.c
index 78ac4ba..1a88fae 100644 (file)
@@ -4,7 +4,7 @@
  * Copyright (C) 2008, Digium, Inc.
  *
  * Brian Degenhardt <bmd@digium.com>
- * Brett Bryant <bbryant@digium.com> 
+ * Brett Bryant <bbryant@digium.com>
  *
  * See http://www.asterisk.org for more information about
  * the Asterisk project. Please do not directly contact
@@ -21,8 +21,8 @@
  *
  * \brief Noise reduction and automatic gain control (AGC)
  *
- * \author Brian Degenhardt <bmd@digium.com> 
- * \author Brett Bryant <bbryant@digium.com> 
+ * \author Brian Degenhardt <bmd@digium.com>
+ * \author Brett Bryant <bbryant@digium.com>
  *
  * \ingroup functions
  *
@@ -63,7 +63,7 @@
                        channel that it is executed on. Using <literal>rx</literal> for audio received
                        and <literal>tx</literal> for audio transmitted to the channel. When using this
                        function you set a target audio level. It is primarily intended for use with
-                       analog lines, but could be useful for other channels as well. The target volume 
+                       analog lines, but could be useful for other channels as well. The target volume
                        is set with a number between <literal>1-32768</literal>. The larger the number
                        the louder (more gain) the channel will receive.</para>
                        <para>Examples:</para>
@@ -77,7 +77,7 @@
                </synopsis>
                <syntax>
                        <parameter name="channeldirection" required="true">
-                               <para>This can be either <literal>rx</literal> or <literal>tx</literal> 
+                               <para>This can be either <literal>rx</literal> or <literal>tx</literal>
                                the values that can be set to this are either <literal>on</literal> and
                                <literal>off</literal></para>
                        </parameter>
@@ -108,7 +108,7 @@ struct speex_info {
        struct speex_direction_info *tx, *rx;
 };
 
-static void destroy_callback(void *data) 
+static void destroy_callback(void *data)
 {
        struct speex_info *si = data;
 
@@ -251,13 +251,13 @@ static int speex_write(struct ast_channel *chan, const char *cmd, char *data, co
        if (!strcasecmp(cmd, "agc")) {
                if (!sscanf(value, "%30f", &(*sdi)->agclevel))
                        (*sdi)->agclevel = ast_true(value) ? DEFAULT_AGC_LEVEL : 0.0;
-       
+
                if ((*sdi)->agclevel > 32768.0) {
-                       ast_log(LOG_WARNING, "AGC(%s)=%.01f is greater than 32768... setting to 32768 instead\n", 
+                       ast_log(LOG_WARNING, "AGC(%s)=%.01f is greater than 32768... setting to 32768 instead\n",
                                        ((*sdi == si->rx) ? "rx" : "tx"), (*sdi)->agclevel);
                        (*sdi)->agclevel = 32768.0;
                }
-       
+
                (*sdi)->agc = !!((*sdi)->agclevel);
 
                if ((*sdi)->state) {
@@ -292,11 +292,11 @@ static int speex_write(struct ast_channel *chan, const char *cmd, char *data, co
                        ast_audiohook_remove(chan, &si->audiohook);
                        ast_audiohook_detach(&si->audiohook);
                }
-               
+
                ast_datastore_free(datastore);
        }
 
-       if (is_new) { 
+       if (is_new) {
                datastore->data = si;
                ast_channel_lock(chan);
                ast_channel_datastore_add(chan, datastore);