diff --git a/apps/app_mixmonitor.c b/apps/app_mixmonitor.c
index bb4b736b54d4ae10c3533730e756a526a1c6901c..ca3a23991b8ad05b63f4946bef73e0946ea084cf 100644
--- a/apps/app_mixmonitor.c
+++ b/apps/app_mixmonitor.c
@@ -46,6 +46,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/app.h"
#include "asterisk/channel.h"
#include "asterisk/autochan.h"
+#include "asterisk/manager.h"
/*** DOCUMENTATION
<application name="MixMonitor" language="en_US">
@@ -124,7 +125,27 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
<ref type="application">MixMonitor</ref>
</see-also>
</application>
-
+ <manager name="MixMonitorMute" language="en_US">
+ <synopsis>
+ Mute / unMute a Mixmonitor recording.
+ </synopsis>
+ <syntax>
+ <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
+ <parameter name="Channel" required="true">
+ <para>Used to specify the channel to mute.</para>
+ </parameter>
+ <parameter name="Direction">
+ <para>Which part of the recording to mute: read, write or both (from channel, to channel or both channels).</para>
+ </parameter>
+ <parameter name="State">
+ <para>Turn mute on or off : 1 to turn on, 0 to turn off.</para>
+ </parameter>
+ </syntax>
+ <description>
+ <para>This action may be used to mute a MixMonitor recording.</para>
+ </description>
+ </manager>
+
***/
#define get_volfactor(x) x ? ((x > 0) ? (1 << x) : ((1 << abs(x)) * -1)) : 0
@@ -606,6 +627,73 @@ static char *handle_cli_mixmonitor(struct ast_cli_entry *e, int cmd, struct ast_
return CLI_SUCCESS;
}
+/*! \brief Mute / unmute a MixMonitor channel */
+static int manager_mute_mixmonitor(struct mansession *s, const struct message *m)
+{
+ struct ast_channel *c = NULL;
+
+ const char *name = astman_get_header(m, "Channel");
+ const char *id = astman_get_header(m, "ActionID");
+ const char *state = astman_get_header(m, "State");
+ const char *direction = astman_get_header(m,"Direction");
+
+ int clearmute = 1;
+
+ enum ast_audiohook_flags flag;
+
+ if (ast_strlen_zero(direction)) {
+ astman_send_error(s, m, "No direction specified. Must be read, write or both");
+ return AMI_SUCCESS;
+ }
+
+ if (!strcasecmp(direction, "read")) {
+ flag = AST_AUDIOHOOK_MUTE_READ;
+ } else if (!strcasecmp(direction, "write")) {
+ flag = AST_AUDIOHOOK_MUTE_WRITE;
+ } else if (!strcasecmp(direction, "both")) {
+ flag = AST_AUDIOHOOK_MUTE_READ | AST_AUDIOHOOK_MUTE_WRITE;
+ } else {
+ astman_send_error(s, m, "Invalid direction specified. Must be read, write or both");
+ return AMI_SUCCESS;
+ }
+
+ if (ast_strlen_zero(name)) {
+ astman_send_error(s, m, "No channel specified");
+ return AMI_SUCCESS;
+ }
+
+ if (ast_strlen_zero(state)) {
+ astman_send_error(s, m, "No state specified");
+ return AMI_SUCCESS;
+ }
+
+ clearmute = ast_false(state);
+ c = ast_channel_get_by_name(name);
+
+ if (!c) {
+ astman_send_error(s, m, "No such channel");
+ return AMI_SUCCESS;
+ }
+
+ if (ast_audiohook_set_mute(c, mixmonitor_spy_type, flag, clearmute)) {
+ c = ast_channel_unref(c);
+ astman_send_error(s, m, "Cannot set mute flag");
+ return AMI_SUCCESS;
+ }
+
+ astman_append(s, "Response: Success\r\n");
+
+ if (!ast_strlen_zero(id)) {
+ astman_append(s, "ActionID: %s\r\n", id);
+ }
+
+ astman_append(s, "\r\n");
+
+ c = ast_channel_unref(c);
+
+ return AMI_SUCCESS;
+}
+
static struct ast_cli_entry cli_mixmonitor[] = {
AST_CLI_DEFINE(handle_cli_mixmonitor, "Execute a MixMonitor command")
};
@@ -617,6 +705,7 @@ static int unload_module(void)
ast_cli_unregister_multiple(cli_mixmonitor, ARRAY_LEN(cli_mixmonitor));
res = ast_unregister_application(stop_app);
res |= ast_unregister_application(app);
+ res |= ast_manager_unregister("MixMonitorMute");
return res;
}
@@ -628,6 +717,7 @@ static int load_module(void)
ast_cli_register_multiple(cli_mixmonitor, ARRAY_LEN(cli_mixmonitor));
res = ast_register_application_xml(app, mixmonitor_exec);
res |= ast_register_application_xml(stop_app, stop_mixmonitor_exec);
+ res |= ast_manager_register_xml("MixMonitorMute", 0, manager_mute_mixmonitor);
return res;
}
diff --git a/include/asterisk/audiohook.h b/include/asterisk/audiohook.h
index 91081ce08fa1d08ec55cb32cb78f1d099cbb10a5..b1e77f8a84fcf65451efed9679065be67adab606 100644
--- a/include/asterisk/audiohook.h
+++ b/include/asterisk/audiohook.h
@@ -62,6 +62,8 @@ enum ast_audiohook_flags {
* slinfactories. We will flush the factories if they contain too many samples.
*/
AST_AUDIOHOOK_SMALL_QUEUE = (1 << 3),
+ AST_AUDIOHOOK_MUTE_READ = (1 << 4), /*!< audiohook should be mute frames read */
+ AST_AUDIOHOOK_MUTE_WRITE = (1 << 5), /*!< audiohook should be mute frames written */
};
#define AST_AUDIOHOOK_SYNC_TOLERANCE 100 /*< Tolerance in milliseconds for audiohooks synchronization */
@@ -277,6 +279,16 @@ int ast_audiohook_volume_get(struct ast_channel *chan, enum ast_audiohook_direct
*/
int ast_audiohook_volume_adjust(struct ast_channel *chan, enum ast_audiohook_direction direction, int volume);
+/*! \brief Mute frames read from or written to a channel
+ * \param chan Channel to muck with
+ * \param source Type of audiohook
+ * \param flag which direction to set / clear
+ * \param clear set or clear muted frames on direction based on flag parameter
+ * \retval 0 success
+ * \retval -1 failure
+ */
+int ast_audiohook_set_mute(struct ast_channel *chan, const char *source, enum ast_audiohook_flags flag, int clear);
+
#if defined(__cplusplus) || defined(c_plusplus)
}
#endif
diff --git a/include/asterisk/frame.h b/include/asterisk/frame.h
index 68a0c7eb6d5e20d41e9d4fce8eeef01ca684693a..0d8d557d2ebce69ea35c3a6f5e74b38040afc420 100644
--- a/include/asterisk/frame.h
+++ b/include/asterisk/frame.h
@@ -728,6 +728,11 @@ static force_inline int ast_format_rate(format_t format)
}
}
+/*!
+ * \brief Clear all audio samples from an ast_frame. The frame must be AST_FRAME_VOICE and AST_FORMAT_SLINEAR
+ */
+int ast_frame_clear(struct ast_frame *frame);
+
#if defined(__cplusplus) || defined(c_plusplus)
}
#endif
diff --git a/main/audiohook.c b/main/audiohook.c
index 4a24a2fce554ad1a2b55d064e0bdb4831333f225..f586b09114d0eb36cdf83bcd13f378f291f3f394 100644
--- a/main/audiohook.c
+++ b/main/audiohook.c
@@ -127,6 +127,7 @@ int ast_audiohook_write_frame(struct ast_audiohook *audiohook, enum ast_audiohoo
int our_factory_ms;
int other_factory_samples;
int other_factory_ms;
+ int muteme = 0;
/* Update last feeding time to be current */
*rwtime = ast_tvnow();
@@ -151,6 +152,17 @@ int ast_audiohook_write_frame(struct ast_audiohook *audiohook, enum ast_audiohoo
ast_slinfactory_flush(other_factory);
}
+ /* swap frame data for zeros if mute is required */
+ if ((ast_test_flag(audiohook, AST_AUDIOHOOK_MUTE_READ) && (direction == AST_AUDIOHOOK_DIRECTION_READ)) ||
+ (ast_test_flag(audiohook, AST_AUDIOHOOK_MUTE_WRITE) && (direction == AST_AUDIOHOOK_DIRECTION_WRITE)) ||
+ (ast_test_flag(audiohook, AST_AUDIOHOOK_MUTE_READ | AST_AUDIOHOOK_MUTE_WRITE) == (AST_AUDIOHOOK_MUTE_READ | AST_AUDIOHOOK_MUTE_WRITE))) {
+ muteme = 1;
+ }
+
+ if (muteme && frame->datalen > 0) {
+ ast_frame_clear(frame);
+ }
+
/* Write frame out to respective factory */
ast_slinfactory_feed(factory, frame);
@@ -1001,3 +1013,37 @@ int ast_audiohook_volume_adjust(struct ast_channel *chan, enum ast_audiohook_dir
return 0;
}
+
+/*! \brief Mute frames read from or written to a channel
+ * \param chan Channel to muck with
+ * \param source Type of audiohook
+ * \param flag which flag to set / clear
+ * \param clear set or clear
+ * \return Returns 0 on success, -1 on failure
+ */
+int ast_audiohook_set_mute(struct ast_channel *chan, const char *source, enum ast_audiohook_flags flag, int clear)
+{
+ struct ast_audiohook *audiohook = NULL;
+
+ ast_channel_lock(chan);
+
+ /* Ensure the channel has audiohooks on it */
+ if (!chan->audiohooks) {
+ ast_channel_unlock(chan);
+ return -1;
+ }
+
+ audiohook = find_audiohook_by_source(chan->audiohooks, source);
+
+ if (audiohook) {
+ if (clear) {
+ ast_clear_flag(audiohook, flag);
+ } else {
+ ast_set_flag(audiohook, flag);
+ }
+ }
+
+ ast_channel_unlock(chan);
+
+ return (audiohook ? 0 : -1);
+}
diff --git a/main/frame.c b/main/frame.c
index c90469bba1574ef06b2d0dffee0b6e57108ba971..af782edc2e851c7aace0ab78509ca97b82d0191a 100644
--- a/main/frame.c
+++ b/main/frame.c
@@ -1587,3 +1587,15 @@ int ast_frame_slinear_sum(struct ast_frame *f1, struct ast_frame *f2)
return 0;
}
+
+int ast_frame_clear(struct ast_frame *frame)
+{
+ struct ast_frame *next;
+
+ for (next = AST_LIST_NEXT(frame, frame_list);
+ frame;
+ frame = next, next = frame ? AST_LIST_NEXT(frame, frame_list) : NULL) {
+ memset(frame->data.ptr, 0, frame->datalen);
+ }
+ return 0;
+}
diff --git a/res/res_mutestream.c b/res/res_mutestream.c
index e4fd82cee274a06fa291c77078fc0a826977c525..4faf464783e16872cce78ae785c95447d88664a6 100644
--- a/res/res_mutestream.c
+++ b/res/res_mutestream.c
@@ -116,19 +116,6 @@ static const struct ast_datastore_info mute_datastore = {
.destroy = destroy_callback
};
-/*! \brief Wipe out all audio samples from an ast_frame. Clean it. */
-static void ast_frame_clear(struct ast_frame *frame)
-{
- struct ast_frame *next;
-
- for (next = AST_LIST_NEXT(frame, frame_list);
- frame;
- frame = next, next = frame ? AST_LIST_NEXT(frame, frame_list) : NULL) {
- memset(frame->data.ptr, 0, frame->datalen);
- }
-}
-
-
/*! \brief The callback from the audiohook subsystem. We basically get a frame to have fun with */
static int mute_callback(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
{