422 lines
13 KiB
C
422 lines
13 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2007, Digium, Inc.
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*
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* Joshua Colp <jcolp@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*! \file
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*
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* \brief Multi-party software based channel mixing
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*
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* \author Joshua Colp <jcolp@digium.com>
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*
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* \ingroup bridges
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*
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* \todo This bridge operates in 8 kHz mode unless a define is uncommented.
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* This needs to be improved so the bridge moves between the dominant codec as needed depending
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* on channels present in the bridge and transcoding capabilities.
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*/
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <sys/time.h>
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#include <signal.h>
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#include <errno.h>
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#include <unistd.h>
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#include "asterisk/module.h"
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#include "asterisk/channel.h"
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#include "asterisk/bridging.h"
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#include "asterisk/bridging_technology.h"
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#include "asterisk/frame.h"
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#include "asterisk/options.h"
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#include "asterisk/logger.h"
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#include "asterisk/slinfactory.h"
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#include "asterisk/astobj2.h"
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#include "asterisk/timing.h"
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#define MAX_DATALEN 3840
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/*! \brief Interval at which mixing will take place. Valid options are 10, 20, and 40. */
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#define SOFTMIX_INTERVAL 20
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/*! \brief Size of the buffer used for sample manipulation */
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#define SOFTMIX_DATALEN(rate) ((rate/50) * (SOFTMIX_INTERVAL / 10))
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/*! \brief Number of samples we are dealing with */
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#define SOFTMIX_SAMPLES(rate) (SOFTMIX_DATALEN(rate) / 2)
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/*! \brief Define used to turn on 16 kHz audio support */
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/* #define SOFTMIX_16_SUPPORT */
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/*! \brief Structure which contains per-channel mixing information */
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struct softmix_channel {
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/*! Lock to protect this structure */
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ast_mutex_t lock;
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/*! Factory which contains audio read in from the channel */
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struct ast_slinfactory factory;
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/*! Frame that contains mixed audio to be written out to the channel */
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struct ast_frame frame;
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/*! Bit used to indicate that the channel provided audio for this mixing interval */
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int have_audio:1;
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/*! Bit used to indicate that a frame is available to be written out to the channel */
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int have_frame:1;
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/*! Buffer containing final mixed audio from all sources */
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short final_buf[MAX_DATALEN];
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/*! Buffer containing only the audio from the channel */
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short our_buf[MAX_DATALEN];
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};
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struct softmix_bridge_data {
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struct ast_timer *timer;
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unsigned int internal_rate;
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};
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/*! \brief Function called when a bridge is created */
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static int softmix_bridge_create(struct ast_bridge *bridge)
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{
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struct softmix_bridge_data *bridge_data;
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if (!(bridge_data = ast_calloc(1, sizeof(*bridge_data)))) {
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return -1;
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}
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if (!(bridge_data->timer = ast_timer_open())) {
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ast_free(bridge_data);
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return -1;
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}
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/* start at 8khz, let it grow from there */
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bridge_data->internal_rate = 8000;
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bridge->bridge_pvt = bridge_data;
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return 0;
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}
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/*! \brief Function called when a bridge is destroyed */
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static int softmix_bridge_destroy(struct ast_bridge *bridge)
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{
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struct softmix_bridge_data *bridge_data = bridge->bridge_pvt;
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if (!bridge->bridge_pvt) {
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return -1;
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}
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ast_timer_close(bridge_data->timer);
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ast_free(bridge_data);
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return 0;
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}
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static void set_softmix_bridge_data(int rate, struct ast_bridge_channel *bridge_channel, int reset)
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{
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struct softmix_channel *sc = bridge_channel->bridge_pvt;
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if (reset) {
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ast_slinfactory_destroy(&sc->factory);
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}
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/* Setup frame parameters */
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sc->frame.frametype = AST_FRAME_VOICE;
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ast_format_set(&sc->frame.subclass.format, ast_format_slin_by_rate(rate), 0);
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sc->frame.data.ptr = sc->final_buf;
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sc->frame.datalen = SOFTMIX_DATALEN(rate);
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sc->frame.samples = SOFTMIX_SAMPLES(rate);
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/* Setup smoother */
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ast_slinfactory_init_with_format(&sc->factory, &sc->frame.subclass.format);
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ast_set_read_format(bridge_channel->chan, &sc->frame.subclass.format);
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ast_set_write_format(bridge_channel->chan, &sc->frame.subclass.format);
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}
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/*! \brief Function called when a channel is joined into the bridge */
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static int softmix_bridge_join(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel)
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{
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struct softmix_channel *sc = NULL;
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struct softmix_bridge_data *bridge_data = bridge->bridge_pvt;
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/* Create a new softmix_channel structure and allocate various things on it */
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if (!(sc = ast_calloc(1, sizeof(*sc)))) {
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return -1;
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}
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/* Can't forget the lock */
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ast_mutex_init(&sc->lock);
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/* Can't forget to record our pvt structure within the bridged channel structure */
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bridge_channel->bridge_pvt = sc;
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set_softmix_bridge_data(bridge_data->internal_rate, bridge_channel, 0);
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return 0;
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}
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/*! \brief Function called when a channel leaves the bridge */
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static int softmix_bridge_leave(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel)
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{
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struct softmix_channel *sc = bridge_channel->bridge_pvt;
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/* Drop mutex lock */
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ast_mutex_destroy(&sc->lock);
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/* Drop the factory */
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ast_slinfactory_destroy(&sc->factory);
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/* Eep! drop ourselves */
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ast_free(sc);
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return 0;
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}
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/*! \brief Function called when a channel writes a frame into the bridge */
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static enum ast_bridge_write_result softmix_bridge_write(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel, struct ast_frame *frame)
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{
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struct softmix_channel *sc = bridge_channel->bridge_pvt;
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/* Only accept audio frames, all others are unsupported */
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if (frame->frametype != AST_FRAME_VOICE) {
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return AST_BRIDGE_WRITE_UNSUPPORTED;
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}
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ast_mutex_lock(&sc->lock);
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/* If a frame was provided add it to the smoother */
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if (frame->frametype == AST_FRAME_VOICE && ast_format_is_slinear(&frame->subclass.format)) {
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ast_slinfactory_feed(&sc->factory, frame);
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}
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/* If a frame is ready to be written out, do so */
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if (sc->have_frame) {
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ast_write(bridge_channel->chan, &sc->frame);
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sc->have_frame = 0;
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}
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/* Alllll done */
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ast_mutex_unlock(&sc->lock);
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return AST_BRIDGE_WRITE_SUCCESS;
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}
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/*! \brief Function called when the channel's thread is poked */
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static int softmix_bridge_poke(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel)
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{
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struct softmix_channel *sc = bridge_channel->bridge_pvt;
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ast_mutex_lock(&sc->lock);
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if (sc->have_frame) {
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ast_write(bridge_channel->chan, &sc->frame);
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sc->have_frame = 0;
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}
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ast_mutex_unlock(&sc->lock);
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return 0;
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}
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/*! \brief Function which acts as the mixing thread */
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static int softmix_bridge_thread(struct ast_bridge *bridge)
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{
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struct {
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/*! Each index represents a sample rate used above the internal rate. */
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unsigned int sample_rates[8];
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/*! Each index represents the number of channels using the same index in the sample_rates array. */
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unsigned int num_channels[8];
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/*! the number of channels above the internal sample rate */
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unsigned int num_above_internal_rate;
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/*! the number of channels at the internal sample rate */
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unsigned int num_at_internal_rate;
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/*! the absolute highest sample rate supported by any channel in the bridge */
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unsigned int highest_supported_rate;
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} stats;
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struct softmix_bridge_data *bridge_data = bridge->bridge_pvt;
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struct ast_timer *timer = bridge_data->timer;
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int timingfd = ast_timer_fd(timer);
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int update_all_rates = 0; /* set this when the internal sample rate has changed */
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int i;
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ast_timer_set_rate(timer, (1000 / SOFTMIX_INTERVAL));
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while (!bridge->stop && !bridge->refresh && bridge->array_num) {
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struct ast_bridge_channel *bridge_channel = NULL;
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short buf[MAX_DATALEN] = {0, };
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int timeout = -1;
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/* these variables help determine if a rate change is required */
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memset(&stats, 0, sizeof(stats));
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stats.highest_supported_rate = 8000;
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/* Go through pulling audio from each factory that has it available */
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AST_LIST_TRAVERSE(&bridge->channels, bridge_channel, entry) {
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struct softmix_channel *sc = bridge_channel->bridge_pvt;
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int channel_native_rate;
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ast_mutex_lock(&sc->lock);
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if (update_all_rates) {
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set_softmix_bridge_data(bridge_data->internal_rate, bridge_channel, 1);
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}
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/* Try to get audio from the factory if available */
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if (ast_slinfactory_available(&sc->factory) >= SOFTMIX_SAMPLES(bridge_data->internal_rate) &&
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ast_slinfactory_read(&sc->factory, sc->our_buf, SOFTMIX_SAMPLES(bridge_data->internal_rate))) {
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short *data1, *data2;
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int i;
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/* Put into the local final buffer */
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for (i = 0, data1 = buf, data2 = sc->our_buf; i < SOFTMIX_DATALEN(bridge_data->internal_rate); i++, data1++, data2++)
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ast_slinear_saturated_add(data1, data2);
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/* Yay we have our own audio */
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sc->have_audio = 1;
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} else {
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/* Awww we don't have audio ;( */
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sc->have_audio = 0;
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}
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/* Gather stats about channel sample rates. */
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channel_native_rate = MAX(ast_format_rate(&bridge_channel->chan->rawwriteformat),
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ast_format_rate(&bridge_channel->chan->rawreadformat));
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if (channel_native_rate > stats.highest_supported_rate) {
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stats.highest_supported_rate = channel_native_rate;
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}
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if (channel_native_rate > bridge_data->internal_rate) {
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for (i = 0; i < ARRAY_LEN(stats.sample_rates); i++) {
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if (stats.sample_rates[i] == channel_native_rate) {
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stats.num_channels[i]++;
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break;
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} else if (!stats.sample_rates[i]) {
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stats.sample_rates[i] = channel_native_rate;
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stats.num_channels[i]++;
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break;
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}
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}
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stats.num_above_internal_rate++;
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} else if (channel_native_rate == bridge_data->internal_rate) {
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stats.num_at_internal_rate++;
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}
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ast_mutex_unlock(&sc->lock);
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}
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/* Next step go through removing the channel's own audio and creating a good frame... */
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AST_LIST_TRAVERSE(&bridge->channels, bridge_channel, entry) {
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struct softmix_channel *sc = bridge_channel->bridge_pvt;
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int i = 0;
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/* Copy from local final buffer to our final buffer */
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memcpy(sc->final_buf, buf, sizeof(sc->final_buf));
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/* If we provided audio then take it out */
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if (sc->have_audio) {
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for (i = 0; i < SOFTMIX_DATALEN(bridge_data->internal_rate); i++) {
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ast_slinear_saturated_subtract(&sc->final_buf[i], &sc->our_buf[i]);
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}
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}
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/* The frame is now ready for use... */
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sc->have_frame = 1;
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/* Poke bridged channel thread just in case */
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pthread_kill(bridge_channel->thread, SIGURG);
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}
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/* Re-adjust the internal bridge sample rate if
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* 1. two or more channels support a higher sample rate
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* 2. no channels support the current sample rate or a higher rate
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*/
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if (stats.num_above_internal_rate >= 2) {
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/* the highest rate is just used as a starting point */
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unsigned int best_rate = stats.highest_supported_rate;
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int best_index = -1;
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/* 1. pick the best sample rate two or more channels support
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* 2. if two or more channels do not support the same rate, pick the
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* lowest sample rate that is still above the internal rate. */
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for (i = 0; ((i < ARRAY_LEN(stats.num_channels)) && stats.num_channels[i]); i++) {
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if ((stats.num_channels[i] >= 2 && (best_index == -1)) ||
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((best_index != -1) &&
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(stats.num_channels[i] >= 2) &&
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(stats.sample_rates[best_index] < stats.sample_rates[i]))) {
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best_rate = stats.sample_rates[i];
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best_index = i;
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} else if (best_index == -1) {
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best_rate = MIN(best_rate, stats.sample_rates[i]);
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}
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}
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ast_debug(1, " Bridge changed from %d To %d\n", bridge_data->internal_rate, best_rate);
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bridge_data->internal_rate = best_rate;
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update_all_rates = 1;
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} else if (!stats.num_at_internal_rate && !stats.num_above_internal_rate) {
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update_all_rates = 1;
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/* in this case, the highest supported rate is actually lower than the internal rate */
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bridge_data->internal_rate = stats.highest_supported_rate;
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ast_debug(1, " Bridge changed from %d to %d\n", bridge_data->internal_rate, stats.highest_supported_rate);
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update_all_rates = 1;
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} else {
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update_all_rates = 0;
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}
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ao2_unlock(bridge);
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/* Wait for the timing source to tell us to wake up and get things done */
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ast_waitfor_n_fd(&timingfd, 1, &timeout, NULL);
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ast_timer_ack(timer, 1);
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ao2_lock(bridge);
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}
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return 0;
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}
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static struct ast_bridge_technology softmix_bridge = {
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.name = "softmix",
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.capabilities = AST_BRIDGE_CAPABILITY_MULTIMIX | AST_BRIDGE_CAPABILITY_THREAD | AST_BRIDGE_CAPABILITY_MULTITHREADED,
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.preference = AST_BRIDGE_PREFERENCE_LOW,
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.create = softmix_bridge_create,
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.destroy = softmix_bridge_destroy,
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.join = softmix_bridge_join,
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.leave = softmix_bridge_leave,
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.write = softmix_bridge_write,
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.thread = softmix_bridge_thread,
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.poke = softmix_bridge_poke,
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};
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static int unload_module(void)
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{
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ast_format_cap_destroy(softmix_bridge.format_capabilities);
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return ast_bridge_technology_unregister(&softmix_bridge);
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}
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static int load_module(void)
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{
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struct ast_format tmp;
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if (!(softmix_bridge.format_capabilities = ast_format_cap_alloc())) {
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return AST_MODULE_LOAD_DECLINE;
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}
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#ifdef SOFTMIX_16_SUPPORT
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ast_format_cap_add(softmix_bridge.format_capabilities, ast_format_set(&tmp, AST_FORMAT_SLINEAR16, 0));
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#else
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ast_format_cap_add(softmix_bridge.format_capabilities, ast_format_set(&tmp, AST_FORMAT_SLINEAR, 0));
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#endif
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return ast_bridge_technology_register(&softmix_bridge);
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}
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AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Multi-party software based channel mixing");
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