/* * Asterisk -- An open source telephony toolkit. * * Copyright (C) 2007, Digium, Inc. * * Joshua Colp * * See http://www.asterisk.org for more information about * the Asterisk project. Please do not directly contact * any of the maintainers of this project for assistance; * the project provides a web site, mailing lists and IRC * channels for your use. * * This program is free software, distributed under the terms of * the GNU General Public License Version 2. See the LICENSE file * at the top of the source tree. */ /*! \file * * \brief Multi-party software based channel mixing * * \author Joshua Colp * * \ingroup bridges * * \todo This bridge operates in 8 kHz mode unless a define is uncommented. * This needs to be improved so the bridge moves between the dominant codec as needed depending * on channels present in the bridge and transcoding capabilities. */ #include "asterisk.h" ASTERISK_FILE_VERSION(__FILE__, "$Revision$") #include #include #include #include #include #include #include #include "asterisk/module.h" #include "asterisk/channel.h" #include "asterisk/bridging.h" #include "asterisk/bridging_technology.h" #include "asterisk/frame.h" #include "asterisk/options.h" #include "asterisk/logger.h" #include "asterisk/slinfactory.h" #include "asterisk/astobj2.h" #include "asterisk/timing.h" /*! \brief Interval at which mixing will take place. Valid options are 10, 20, and 40. */ #define SOFTMIX_INTERVAL 20 /*! \brief Size of the buffer used for sample manipulation */ #define SOFTMIX_DATALEN (160 * (SOFTMIX_INTERVAL / 10)) /*! \brief Number of samples we are dealing with */ #define SOFTMIX_SAMPLES (SOFTMIX_DATALEN / 2) /*! \brief Define used to turn on 16 kHz audio support */ /* #define SOFTMIX_16_SUPPORT */ /*! \brief Structure which contains per-channel mixing information */ struct softmix_channel { /*! Lock to protect this structure */ ast_mutex_t lock; /*! Factory which contains audio read in from the channel */ struct ast_slinfactory factory; /*! Frame that contains mixed audio to be written out to the channel */ struct ast_frame frame; /*! Bit used to indicate that the channel provided audio for this mixing interval */ int have_audio:1; /*! Bit used to indicate that a frame is available to be written out to the channel */ int have_frame:1; /*! Buffer containing final mixed audio from all sources */ short final_buf[SOFTMIX_DATALEN]; /*! Buffer containing only the audio from the channel */ short our_buf[SOFTMIX_DATALEN]; }; /*! \brief Function called when a bridge is created */ static int softmix_bridge_create(struct ast_bridge *bridge) { struct ast_timer *timer; if (!(timer = ast_timer_open())) { return -1; } bridge->bridge_pvt = timer; return 0; } /*! \brief Function called when a bridge is destroyed */ static int softmix_bridge_destroy(struct ast_bridge *bridge) { if (!bridge->bridge_pvt) { return -1; } ast_timer_close((struct ast_timer *) bridge->bridge_pvt); return 0; } /*! \brief Function called when a channel is joined into the bridge */ static int softmix_bridge_join(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel) { struct softmix_channel *sc = NULL; /* Create a new softmix_channel structure and allocate various things on it */ if (!(sc = ast_calloc(1, sizeof(*sc)))) { return -1; } /* Can't forget the lock */ ast_mutex_init(&sc->lock); /* Setup smoother */ ast_slinfactory_init(&sc->factory); /* Setup frame parameters */ sc->frame.frametype = AST_FRAME_VOICE; #ifdef SOFTMIX_16_SUPPORT sc->frame.subclass.codec = AST_FORMAT_SLINEAR16; #else sc->frame.subclass.codec = AST_FORMAT_SLINEAR; #endif sc->frame.data.ptr = sc->final_buf; sc->frame.datalen = SOFTMIX_DATALEN; sc->frame.samples = SOFTMIX_SAMPLES; /* Can't forget to record our pvt structure within the bridged channel structure */ bridge_channel->bridge_pvt = sc; return 0; } /*! \brief Function called when a channel leaves the bridge */ static int softmix_bridge_leave(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel) { struct softmix_channel *sc = bridge_channel->bridge_pvt; /* Drop mutex lock */ ast_mutex_destroy(&sc->lock); /* Drop the factory */ ast_slinfactory_destroy(&sc->factory); /* Eep! drop ourselves */ ast_free(sc); return 0; } /*! \brief Function called when a channel writes a frame into the bridge */ static enum ast_bridge_write_result softmix_bridge_write(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel, struct ast_frame *frame) { struct softmix_channel *sc = bridge_channel->bridge_pvt; /* Only accept audio frames, all others are unsupported */ if (frame->frametype != AST_FRAME_VOICE) { return AST_BRIDGE_WRITE_UNSUPPORTED; } ast_mutex_lock(&sc->lock); /* If a frame was provided add it to the smoother */ #ifdef SOFTMIX_16_SUPPORT if (frame->frametype == AST_FRAME_VOICE && frame->subclass.codec == AST_FORMAT_SLINEAR16) { #else if (frame->frametype == AST_FRAME_VOICE && frame->subclass.codec == AST_FORMAT_SLINEAR) { #endif ast_slinfactory_feed(&sc->factory, frame); } /* If a frame is ready to be written out, do so */ if (sc->have_frame) { ast_write(bridge_channel->chan, &sc->frame); sc->have_frame = 0; } /* Alllll done */ ast_mutex_unlock(&sc->lock); return AST_BRIDGE_WRITE_SUCCESS; } /*! \brief Function called when the channel's thread is poked */ static int softmix_bridge_poke(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel) { struct softmix_channel *sc = bridge_channel->bridge_pvt; ast_mutex_lock(&sc->lock); if (sc->have_frame) { ast_write(bridge_channel->chan, &sc->frame); sc->have_frame = 0; } ast_mutex_unlock(&sc->lock); return 0; } /*! \brief Function which acts as the mixing thread */ static int softmix_bridge_thread(struct ast_bridge *bridge) { struct ast_timer *timer = (struct ast_timer *) bridge->bridge_pvt; int timingfd = ast_timer_fd(timer); ast_timer_set_rate(timer, (1000 / SOFTMIX_INTERVAL)); while (!bridge->stop && !bridge->refresh && bridge->array_num) { struct ast_bridge_channel *bridge_channel = NULL; short buf[SOFTMIX_DATALEN] = {0, }; int timeout = -1; /* Go through pulling audio from each factory that has it available */ AST_LIST_TRAVERSE(&bridge->channels, bridge_channel, entry) { struct softmix_channel *sc = bridge_channel->bridge_pvt; ast_mutex_lock(&sc->lock); /* Try to get audio from the factory if available */ if (ast_slinfactory_available(&sc->factory) >= SOFTMIX_SAMPLES && ast_slinfactory_read(&sc->factory, sc->our_buf, SOFTMIX_SAMPLES)) { short *data1, *data2; int i; /* Put into the local final buffer */ for (i = 0, data1 = buf, data2 = sc->our_buf; i < SOFTMIX_DATALEN; i++, data1++, data2++) ast_slinear_saturated_add(data1, data2); /* Yay we have our own audio */ sc->have_audio = 1; } else { /* Awww we don't have audio ;( */ sc->have_audio = 0; } ast_mutex_unlock(&sc->lock); } /* Next step go through removing the channel's own audio and creating a good frame... */ AST_LIST_TRAVERSE(&bridge->channels, bridge_channel, entry) { struct softmix_channel *sc = bridge_channel->bridge_pvt; int i = 0; /* Copy from local final buffer to our final buffer */ memcpy(sc->final_buf, buf, sizeof(sc->final_buf)); /* If we provided audio then take it out */ if (sc->have_audio) { for (i = 0; i < SOFTMIX_DATALEN; i++) { ast_slinear_saturated_subtract(&sc->final_buf[i], &sc->our_buf[i]); } } /* The frame is now ready for use... */ sc->have_frame = 1; /* Poke bridged channel thread just in case */ pthread_kill(bridge_channel->thread, SIGURG); } ao2_unlock(bridge); /* Wait for the timing source to tell us to wake up and get things done */ ast_waitfor_n_fd(&timingfd, 1, &timeout, NULL); ast_timer_ack(timer, 1); ao2_lock(bridge); } return 0; } static struct ast_bridge_technology softmix_bridge = { .name = "softmix", .capabilities = AST_BRIDGE_CAPABILITY_MULTIMIX | AST_BRIDGE_CAPABILITY_THREAD | AST_BRIDGE_CAPABILITY_MULTITHREADED, .preference = AST_BRIDGE_PREFERENCE_LOW, #ifdef SOFTMIX_16_SUPPORT .formats = AST_FORMAT_SLINEAR16, #else .formats = AST_FORMAT_SLINEAR, #endif .create = softmix_bridge_create, .destroy = softmix_bridge_destroy, .join = softmix_bridge_join, .leave = softmix_bridge_leave, .write = softmix_bridge_write, .thread = softmix_bridge_thread, .poke = softmix_bridge_poke, }; static int unload_module(void) { return ast_bridge_technology_unregister(&softmix_bridge); } static int load_module(void) { return ast_bridge_technology_register(&softmix_bridge); } AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Multi-party software based channel mixing");