diff options
| -rw-r--r-- | apps/app_mixmonitor.c | 92 | ||||
| -rw-r--r-- | include/asterisk/audiohook.h | 12 | ||||
| -rw-r--r-- | include/asterisk/frame.h | 5 | ||||
| -rw-r--r-- | main/audiohook.c | 46 | ||||
| -rw-r--r-- | main/frame.c | 12 | ||||
| -rw-r--r-- | res/res_mutestream.c | 13 |
6 files changed, 166 insertions, 14 deletions
diff --git a/apps/app_mixmonitor.c b/apps/app_mixmonitor.c index bb4b736b5..ca3a23991 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 91081ce08..b1e77f8a8 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 68a0c7eb6..0d8d557d2 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 4a24a2fce..f586b0911 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 c90469bba..af782edc2 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 e4fd82cee..4faf46478 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) { |