1 files changed, 303 insertions, 0 deletions
diff --git a/bridges/bridge_softmix.c b/bridges/bridge_softmix.c
new file mode 100644
index 000000000..4f1e4d76f
--- /dev/null
+++ b/bridges/bridge_softmix.c
@@ -0,0 +1,303 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2007, Digium, Inc.
+ *
+ * Joshua Colp <jcolp@digium.com>
+ *
+ * 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 <jcolp@digium.com>
+ *
+ * \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 <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <sys/time.h>
+#include <signal.h>
+#include <errno.h>
+#include <unistd.h>
+
+#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)
+{
+	int timingfd;
+
+	if ((timingfd = ast_timer_open()) < 0) {
+		return -1;
+	}
+
+	ast_timer_close(timingfd);
+
+	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 = AST_FORMAT_SLINEAR16;
+#else
+	sc->frame.subclass = 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 == AST_FORMAT_SLINEAR16) {
+#else
+	if (frame->frametype == AST_FRAME_VOICE && frame->subclass == 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)
+{
+	int timingfd;
+
+	if ((timingfd = ast_timer_open()) < 0) {
+		return -1;
+	}
+
+	ast_timer_set_rate(timingfd, (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(timingfd, 1);
+
+		ao2_lock(bridge);
+	}
+
+	ast_timer_set_rate(timingfd, 0);
+	ast_timer_close(timingfd);
+
+	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,
+	.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");