https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines Merged revisions 328209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines Introduce <support_level> tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
937 lines
32 KiB
C
937 lines
32 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2011, Digium, Inc.
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*
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* Joshua Colp <jcolp@digium.com>
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* David Vossel <dvossel@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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* \author David Vossel <dvossel@digium.com>
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*
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* \ingroup bridges
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*/
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/*** MODULEINFO
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<support_level>core</support_level>
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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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#include "asterisk/translate.h"
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#define MAX_DATALEN 8096
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/*! \brief Interval at which mixing will take place. Valid options are 10, 20, and 40. */
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#define DEFAULT_SOFTMIX_INTERVAL 20
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/*! \brief Size of the buffer used for sample manipulation */
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#define SOFTMIX_DATALEN(rate, interval) ((rate/50) * (interval / 10))
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/*! \brief Number of samples we are dealing with */
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#define SOFTMIX_SAMPLES(rate, interval) (SOFTMIX_DATALEN(rate, interval) / 2)
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/*! \brief Number of mixing iterations to perform between gathering statistics. */
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#define SOFTMIX_STAT_INTERVAL 100
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/* This is the threshold in ms at which a channel's own audio will stop getting
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* mixed out its own write audio stream because it is not talking. */
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#define DEFAULT_SOFTMIX_SILENCE_THRESHOLD 2500
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#define DEFAULT_SOFTMIX_TALKING_THRESHOLD 160
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#define DEFAULT_ENERGY_HISTORY_LEN 150
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struct video_follow_talker_data {
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/*! audio energy history */
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int energy_history[DEFAULT_ENERGY_HISTORY_LEN];
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/*! The current slot being used in the history buffer, this
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* increments and wraps around */
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int energy_history_cur_slot;
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/*! The current energy sum used for averages. */
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int energy_accum;
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/*! The current energy average */
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int energy_average;
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};
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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 write_frame;
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/*! Frame that contains mixed audio read from the channel */
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struct ast_frame read_frame;
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/*! DSP for detecting silence */
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struct ast_dsp *dsp;
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/*! Bit used to indicate if a channel is talking or not. This affects how
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* the channel's audio is mixed back to it. */
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int talking:1;
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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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/*! Data pertaining to talker mode for video conferencing */
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struct video_follow_talker_data video_talker;
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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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unsigned int internal_mixing_interval;
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};
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struct softmix_stats {
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/*! Each index represents a sample rate used above the internal rate. */
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unsigned int sample_rates[16];
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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[16];
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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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/*! Is the sample rate locked by the bridge, if so what is that rate.*/
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unsigned int locked_rate;
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};
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struct softmix_mixing_array {
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int max_num_entries;
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int used_entries;
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int16_t **buffers;
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};
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struct softmix_translate_helper_entry {
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int num_times_requested; /*!< Once this entry is no longer requested, free the trans_pvt
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and re-init if it was usable. */
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struct ast_format dst_format; /*!< The destination format for this helper */
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struct ast_trans_pvt *trans_pvt; /*!< the translator for this slot. */
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struct ast_frame *out_frame; /*!< The output frame from the last translation */
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AST_LIST_ENTRY(softmix_translate_helper_entry) entry;
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};
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struct softmix_translate_helper {
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struct ast_format slin_src; /*!< the source format expected for all the translators */
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AST_LIST_HEAD_NOLOCK(, softmix_translate_helper_entry) entries;
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};
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static struct softmix_translate_helper_entry *softmix_translate_helper_entry_alloc(struct ast_format *dst)
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{
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struct softmix_translate_helper_entry *entry;
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if (!(entry = ast_calloc(1, sizeof(*entry)))) {
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return NULL;
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}
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ast_format_copy(&entry->dst_format, dst);
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return entry;
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}
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static void *softmix_translate_helper_free_entry(struct softmix_translate_helper_entry *entry)
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{
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if (entry->trans_pvt) {
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ast_translator_free_path(entry->trans_pvt);
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}
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if (entry->out_frame) {
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ast_frfree(entry->out_frame);
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}
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ast_free(entry);
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return NULL;
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}
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static void softmix_translate_helper_init(struct softmix_translate_helper *trans_helper, unsigned int sample_rate)
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{
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memset(trans_helper, 0, sizeof(*trans_helper));
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ast_format_set(&trans_helper->slin_src, ast_format_slin_by_rate(sample_rate), 0);
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}
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static void softmix_translate_helper_destroy(struct softmix_translate_helper *trans_helper)
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{
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struct softmix_translate_helper_entry *entry;
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while ((entry = AST_LIST_REMOVE_HEAD(&trans_helper->entries, entry))) {
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softmix_translate_helper_free_entry(entry);
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}
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}
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static void softmix_translate_helper_change_rate(struct softmix_translate_helper *trans_helper, unsigned int sample_rate)
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{
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struct softmix_translate_helper_entry *entry;
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ast_format_set(&trans_helper->slin_src, ast_format_slin_by_rate(sample_rate), 0);
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AST_LIST_TRAVERSE_SAFE_BEGIN(&trans_helper->entries, entry, entry) {
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if (entry->trans_pvt) {
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ast_translator_free_path(entry->trans_pvt);
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if (!(entry->trans_pvt = ast_translator_build_path(&entry->dst_format, &trans_helper->slin_src))) {
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AST_LIST_REMOVE_CURRENT(entry);
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entry = softmix_translate_helper_free_entry(entry);
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}
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}
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}
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AST_LIST_TRAVERSE_SAFE_END;
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}
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/*!
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* \internal
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* \brief Get the next available audio on the softmix channel's read stream
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* and determine if it should be mixed out or not on the write stream.
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*
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* \retval pointer to buffer containing the exact number of samples requested on success.
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* \retval NULL if no samples are present
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*/
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static int16_t *softmix_process_read_audio(struct softmix_channel *sc, unsigned int num_samples)
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{
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if ((ast_slinfactory_available(&sc->factory) >= num_samples) &&
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ast_slinfactory_read(&sc->factory, sc->our_buf, num_samples)) {
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sc->have_audio = 1;
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return sc->our_buf;
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}
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sc->have_audio = 0;
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return NULL;
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}
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/*!
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* \internal
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* \brief Process a softmix channel's write audio
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*
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* \details This function will remove the channel's talking from its own audio if present and
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* possibly even do the channel's write translation for it depending on how many other
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* channels use the same write format.
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*/
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static void softmix_process_write_audio(struct softmix_translate_helper *trans_helper,
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struct ast_format *raw_write_fmt,
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struct softmix_channel *sc)
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{
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struct softmix_translate_helper_entry *entry = NULL;
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int i;
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/* If we provided audio that was not determined to be silence,
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* then take it out while in slinear format. */
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if (sc->have_audio && sc->talking) {
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for (i = 0; i < sc->write_frame.samples; 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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/* do not do any special write translate optimization if we had to make
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* a special mix for them to remove their own audio. */
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return;
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}
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AST_LIST_TRAVERSE(&trans_helper->entries, entry, entry) {
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if (ast_format_cmp(&entry->dst_format, raw_write_fmt) == AST_FORMAT_CMP_EQUAL) {
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entry->num_times_requested++;
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} else {
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continue;
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}
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if (!entry->trans_pvt && (entry->num_times_requested > 1)) {
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entry->trans_pvt = ast_translator_build_path(&entry->dst_format, &trans_helper->slin_src);
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}
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if (entry->trans_pvt && !entry->out_frame) {
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entry->out_frame = ast_translate(entry->trans_pvt, &sc->write_frame, 0);
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}
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if (entry->out_frame && (entry->out_frame->datalen < MAX_DATALEN)) {
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ast_format_copy(&sc->write_frame.subclass.format, &entry->out_frame->subclass.format);
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memcpy(sc->final_buf, entry->out_frame->data.ptr, entry->out_frame->datalen);
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sc->write_frame.datalen = entry->out_frame->datalen;
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sc->write_frame.samples = entry->out_frame->samples;
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}
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break;
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}
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/* add new entry into list if this format destination was not matched. */
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if (!entry && (entry = softmix_translate_helper_entry_alloc(raw_write_fmt))) {
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AST_LIST_INSERT_HEAD(&trans_helper->entries, entry, entry);
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}
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}
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static void softmix_translate_helper_cleanup(struct softmix_translate_helper *trans_helper)
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{
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struct softmix_translate_helper_entry *entry = NULL;
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AST_LIST_TRAVERSE(&trans_helper->entries, entry, entry) {
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if (entry->out_frame) {
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ast_frfree(entry->out_frame);
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entry->out_frame = NULL;
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}
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entry->num_times_requested = 0;
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}
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}
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static void softmix_bridge_data_destroy(void *obj)
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{
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struct softmix_bridge_data *softmix_data = obj;
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ast_timer_close(softmix_data->timer);
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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 *softmix_data;
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if (!(softmix_data = ao2_alloc(sizeof(*softmix_data), softmix_bridge_data_destroy))) {
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return -1;
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}
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if (!(softmix_data->timer = ast_timer_open())) {
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ao2_ref(softmix_data, -1);
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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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softmix_data->internal_rate = 8000;
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softmix_data->internal_mixing_interval = DEFAULT_SOFTMIX_INTERVAL;
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bridge->bridge_pvt = softmix_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 *softmix_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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ao2_ref(softmix_data, -1);
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bridge->bridge_pvt = NULL;
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return 0;
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}
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static void set_softmix_bridge_data(int rate, int interval, 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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unsigned int channel_read_rate = ast_format_rate(&bridge_channel->chan->rawreadformat);
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ast_mutex_lock(&sc->lock);
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if (reset) {
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ast_slinfactory_destroy(&sc->factory);
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ast_dsp_free(sc->dsp);
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}
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/* Setup read/write frame parameters */
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sc->write_frame.frametype = AST_FRAME_VOICE;
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ast_format_set(&sc->write_frame.subclass.format, ast_format_slin_by_rate(rate), 0);
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sc->write_frame.data.ptr = sc->final_buf;
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sc->write_frame.datalen = SOFTMIX_DATALEN(rate, interval);
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sc->write_frame.samples = SOFTMIX_SAMPLES(rate, interval);
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sc->read_frame.frametype = AST_FRAME_VOICE;
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ast_format_set(&sc->read_frame.subclass.format, ast_format_slin_by_rate(channel_read_rate), 0);
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sc->read_frame.data.ptr = sc->our_buf;
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sc->read_frame.datalen = SOFTMIX_DATALEN(channel_read_rate, interval);
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sc->read_frame.samples = SOFTMIX_SAMPLES(channel_read_rate, interval);
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/* Setup smoother */
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ast_slinfactory_init_with_format(&sc->factory, &sc->write_frame.subclass.format);
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/* set new read and write formats on channel. */
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ast_set_read_format(bridge_channel->chan, &sc->read_frame.subclass.format);
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ast_set_write_format(bridge_channel->chan, &sc->write_frame.subclass.format);
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/* set up new DSP. This is on the read side only right before the read frame enters the smoother. */
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sc->dsp = ast_dsp_new_with_rate(channel_read_rate);
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/* we want to aggressively detect silence to avoid feedback */
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if (bridge_channel->tech_args.talking_threshold) {
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ast_dsp_set_threshold(sc->dsp, bridge_channel->tech_args.talking_threshold);
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} else {
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ast_dsp_set_threshold(sc->dsp, DEFAULT_SOFTMIX_TALKING_THRESHOLD);
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}
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ast_mutex_unlock(&sc->lock);
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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 *softmix_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(softmix_data->internal_rate,
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softmix_data->internal_mixing_interval ? softmix_data->internal_mixing_interval : DEFAULT_SOFTMIX_INTERVAL,
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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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if (!(bridge_channel->bridge_pvt)) {
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return 0;
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}
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bridge_channel->bridge_pvt = NULL;
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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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/* Drop the DSP */
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ast_dsp_free(sc->dsp);
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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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/*!
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* \internal
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* \brief If the bridging core passes DTMF to us, then they want it to be distributed out to all memebers. Do that here.
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*/
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static void softmix_pass_dtmf(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel, struct ast_frame *frame)
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{
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struct ast_bridge_channel *tmp;
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AST_LIST_TRAVERSE(&bridge->channels, tmp, entry) {
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if (tmp == bridge_channel) {
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continue;
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}
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ast_write(tmp->chan, frame);
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}
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}
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static void softmix_pass_video_top_priority(struct ast_bridge *bridge, struct ast_frame *frame)
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{
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struct ast_bridge_channel *tmp;
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AST_LIST_TRAVERSE(&bridge->channels, tmp, entry) {
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if (tmp->suspended) {
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continue;
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}
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if (ast_bridge_is_video_src(bridge, tmp->chan) == 1) {
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ast_write(tmp->chan, frame);
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break;
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}
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}
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}
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static void softmix_pass_video_all(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel, struct ast_frame *frame, int echo)
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{
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struct ast_bridge_channel *tmp;
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AST_LIST_TRAVERSE(&bridge->channels, tmp, entry) {
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if (tmp->suspended) {
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continue;
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}
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|
if ((tmp->chan == bridge_channel->chan) && !echo) {
|
|
continue;
|
|
}
|
|
ast_write(tmp->chan, frame);
|
|
}
|
|
}
|
|
|
|
/*! \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;
|
|
struct softmix_bridge_data *softmix_data = bridge->bridge_pvt;
|
|
int totalsilence = 0;
|
|
int cur_energy = 0;
|
|
int silence_threshold = bridge_channel->tech_args.silence_threshold ?
|
|
bridge_channel->tech_args.silence_threshold :
|
|
DEFAULT_SOFTMIX_SILENCE_THRESHOLD;
|
|
char update_talking = -1; /* if this is set to 0 or 1, tell the bridge that the channel has started or stopped talking. */
|
|
int res = AST_BRIDGE_WRITE_SUCCESS;
|
|
|
|
/* Only accept audio frames, all others are unsupported */
|
|
if (frame->frametype == AST_FRAME_DTMF_END || frame->frametype == AST_FRAME_DTMF_BEGIN) {
|
|
softmix_pass_dtmf(bridge, bridge_channel, frame);
|
|
goto bridge_write_cleanup;
|
|
} else if (frame->frametype != AST_FRAME_VOICE && frame->frametype != AST_FRAME_VIDEO) {
|
|
res = AST_BRIDGE_WRITE_UNSUPPORTED;
|
|
goto bridge_write_cleanup;
|
|
} else if (frame->datalen == 0) {
|
|
goto bridge_write_cleanup;
|
|
}
|
|
|
|
/* Determine if this video frame should be distributed or not */
|
|
if (frame->frametype == AST_FRAME_VIDEO) {
|
|
int num_src = ast_bridge_number_video_src(bridge);
|
|
int video_src_priority = ast_bridge_is_video_src(bridge, bridge_channel->chan);
|
|
|
|
switch (bridge->video_mode.mode) {
|
|
case AST_BRIDGE_VIDEO_MODE_NONE:
|
|
break;
|
|
case AST_BRIDGE_VIDEO_MODE_SINGLE_SRC:
|
|
if (video_src_priority == 1) {
|
|
softmix_pass_video_all(bridge, bridge_channel, frame, 1);
|
|
}
|
|
break;
|
|
case AST_BRIDGE_VIDEO_MODE_TALKER_SRC:
|
|
ast_mutex_lock(&sc->lock);
|
|
ast_bridge_update_talker_src_video_mode(bridge, bridge_channel->chan, sc->video_talker.energy_average, ast_format_get_video_mark(&frame->subclass.format));
|
|
ast_mutex_unlock(&sc->lock);
|
|
if (video_src_priority == 1) {
|
|
int echo = num_src > 1 ? 0 : 1;
|
|
softmix_pass_video_all(bridge, bridge_channel, frame, echo);
|
|
} else if (video_src_priority == 2) {
|
|
softmix_pass_video_top_priority(bridge, frame);
|
|
}
|
|
break;
|
|
}
|
|
goto bridge_write_cleanup;
|
|
}
|
|
|
|
/* If we made it here, we are going to write the frame into the conference */
|
|
ast_mutex_lock(&sc->lock);
|
|
ast_dsp_silence_with_energy(sc->dsp, frame, &totalsilence, &cur_energy);
|
|
|
|
if (bridge->video_mode.mode == AST_BRIDGE_VIDEO_MODE_TALKER_SRC) {
|
|
int cur_slot = sc->video_talker.energy_history_cur_slot;
|
|
sc->video_talker.energy_accum -= sc->video_talker.energy_history[cur_slot];
|
|
sc->video_talker.energy_accum += cur_energy;
|
|
sc->video_talker.energy_history[cur_slot] = cur_energy;
|
|
sc->video_talker.energy_average = sc->video_talker.energy_accum / DEFAULT_ENERGY_HISTORY_LEN;
|
|
sc->video_talker.energy_history_cur_slot++;
|
|
if (sc->video_talker.energy_history_cur_slot == DEFAULT_ENERGY_HISTORY_LEN) {
|
|
sc->video_talker.energy_history_cur_slot = 0; /* wrap around */
|
|
}
|
|
}
|
|
|
|
if (totalsilence < silence_threshold) {
|
|
if (!sc->talking) {
|
|
update_talking = 1;
|
|
}
|
|
sc->talking = 1; /* tell the write process we have audio to be mixed out */
|
|
} else {
|
|
if (sc->talking) {
|
|
update_talking = 0;
|
|
}
|
|
sc->talking = 0;
|
|
}
|
|
|
|
/* Before adding audio in, make sure we haven't fallen behind. If audio has fallen
|
|
* behind 4 times the amount of samples mixed on every iteration of the mixer, Re-sync
|
|
* the audio by flushing the buffer before adding new audio in. */
|
|
if (ast_slinfactory_available(&sc->factory) > (4 * SOFTMIX_SAMPLES(softmix_data->internal_rate, softmix_data->internal_mixing_interval))) {
|
|
ast_slinfactory_flush(&sc->factory);
|
|
}
|
|
|
|
/* If a frame was provided add it to the smoother, unless drop silence is enabled and this frame
|
|
* is not determined to be talking. */
|
|
if (!(bridge_channel->tech_args.drop_silence && !sc->talking) &&
|
|
(frame->frametype == AST_FRAME_VOICE && ast_format_is_slinear(&frame->subclass.format))) {
|
|
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->write_frame);
|
|
sc->have_frame = 0;
|
|
}
|
|
|
|
/* Alllll done */
|
|
ast_mutex_unlock(&sc->lock);
|
|
|
|
if (update_talking != -1) {
|
|
ast_bridge_notify_talking(bridge, bridge_channel, update_talking);
|
|
}
|
|
|
|
return res;
|
|
|
|
bridge_write_cleanup:
|
|
/* Even though the frame is not being written into the conference because it is not audio,
|
|
* we should use this opportunity to check to see if a frame is ready to be written out from
|
|
* the conference to the channel. */
|
|
ast_mutex_lock(&sc->lock);
|
|
if (sc->have_frame) {
|
|
ast_write(bridge_channel->chan, &sc->write_frame);
|
|
sc->have_frame = 0;
|
|
}
|
|
ast_mutex_unlock(&sc->lock);
|
|
|
|
return res;
|
|
}
|
|
|
|
/*! \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->write_frame);
|
|
sc->have_frame = 0;
|
|
}
|
|
|
|
ast_mutex_unlock(&sc->lock);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static void gather_softmix_stats(struct softmix_stats *stats,
|
|
const struct softmix_bridge_data *softmix_data,
|
|
struct ast_bridge_channel *bridge_channel)
|
|
{
|
|
int channel_native_rate;
|
|
int i;
|
|
/* Gather stats about channel sample rates. */
|
|
channel_native_rate = MAX(ast_format_rate(&bridge_channel->chan->rawwriteformat),
|
|
ast_format_rate(&bridge_channel->chan->rawreadformat));
|
|
|
|
if (channel_native_rate > stats->highest_supported_rate) {
|
|
stats->highest_supported_rate = channel_native_rate;
|
|
}
|
|
if (channel_native_rate > softmix_data->internal_rate) {
|
|
for (i = 0; i < ARRAY_LEN(stats->sample_rates); i++) {
|
|
if (stats->sample_rates[i] == channel_native_rate) {
|
|
stats->num_channels[i]++;
|
|
break;
|
|
} else if (!stats->sample_rates[i]) {
|
|
stats->sample_rates[i] = channel_native_rate;
|
|
stats->num_channels[i]++;
|
|
break;
|
|
}
|
|
}
|
|
stats->num_above_internal_rate++;
|
|
} else if (channel_native_rate == softmix_data->internal_rate) {
|
|
stats->num_at_internal_rate++;
|
|
}
|
|
}
|
|
/*!
|
|
* \internal
|
|
* \brief Analyse mixing statistics and change bridges internal rate
|
|
* if necessary.
|
|
*
|
|
* \retval 0, no changes to internal rate
|
|
* \ratval 1, internal rate was changed, update all the channels on the next mixing iteration.
|
|
*/
|
|
static unsigned int analyse_softmix_stats(struct softmix_stats *stats, struct softmix_bridge_data *softmix_data)
|
|
{
|
|
int i;
|
|
/* Re-adjust the internal bridge sample rate if
|
|
* 1. The bridge's internal sample rate is locked in at a sample
|
|
* rate other than the current sample rate being used.
|
|
* 2. two or more channels support a higher sample rate
|
|
* 3. no channels support the current sample rate or a higher rate
|
|
*/
|
|
if (stats->locked_rate) {
|
|
/* if the rate is locked by the bridge, only update it if it differs
|
|
* from the current rate we are using. */
|
|
if (softmix_data->internal_rate != stats->locked_rate) {
|
|
softmix_data->internal_rate = stats->locked_rate;
|
|
ast_debug(1, " Bridge is locked in at sample rate %d\n", softmix_data->internal_rate);
|
|
return 1;
|
|
}
|
|
} else if (stats->num_above_internal_rate >= 2) {
|
|
/* the highest rate is just used as a starting point */
|
|
unsigned int best_rate = stats->highest_supported_rate;
|
|
int best_index = -1;
|
|
|
|
for (i = 0; i < ARRAY_LEN(stats->num_channels); i++) {
|
|
if (stats->num_channels[i]) {
|
|
break;
|
|
}
|
|
/* best_rate starts out being the first sample rate
|
|
* greater than the internal sample rate that 2 or
|
|
* more channels support. */
|
|
if (stats->num_channels[i] >= 2 && (best_index == -1)) {
|
|
best_rate = stats->sample_rates[i];
|
|
best_index = i;
|
|
/* If it has been detected that multiple rates above
|
|
* the internal rate are present, compare those rates
|
|
* to each other and pick the highest one two or more
|
|
* channels support. */
|
|
} else if (((best_index != -1) &&
|
|
(stats->num_channels[i] >= 2) &&
|
|
(stats->sample_rates[best_index] < stats->sample_rates[i]))) {
|
|
best_rate = stats->sample_rates[i];
|
|
best_index = i;
|
|
/* It is possible that multiple channels exist with native sample
|
|
* rates above the internal sample rate, but none of those channels
|
|
* have the same rate in common. In this case, the lowest sample
|
|
* rate among those channels is picked. Over time as additional
|
|
* statistic runs are made the internal sample rate number will
|
|
* adjust to the most optimal sample rate, but it may take multiple
|
|
* iterations. */
|
|
} else if (best_index == -1) {
|
|
best_rate = MIN(best_rate, stats->sample_rates[i]);
|
|
}
|
|
}
|
|
|
|
ast_debug(1, " Bridge changed from %d To %d\n", softmix_data->internal_rate, best_rate);
|
|
softmix_data->internal_rate = best_rate;
|
|
return 1;
|
|
} else if (!stats->num_at_internal_rate && !stats->num_above_internal_rate) {
|
|
/* In this case, the highest supported rate is actually lower than the internal rate */
|
|
softmix_data->internal_rate = stats->highest_supported_rate;
|
|
ast_debug(1, " Bridge changed from %d to %d\n", softmix_data->internal_rate, stats->highest_supported_rate);
|
|
return 1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static int softmix_mixing_array_init(struct softmix_mixing_array *mixing_array, unsigned int starting_num_entries)
|
|
{
|
|
memset(mixing_array, 0, sizeof(*mixing_array));
|
|
mixing_array->max_num_entries = starting_num_entries;
|
|
if (!(mixing_array->buffers = ast_calloc(mixing_array->max_num_entries, sizeof(int16_t *)))) {
|
|
ast_log(LOG_NOTICE, "Failed to allocate softmix mixing structure. \n");
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static void softmix_mixing_array_destroy(struct softmix_mixing_array *mixing_array)
|
|
{
|
|
ast_free(mixing_array->buffers);
|
|
}
|
|
|
|
static int softmix_mixing_array_grow(struct softmix_mixing_array *mixing_array, unsigned int num_entries)
|
|
{
|
|
int16_t **tmp;
|
|
/* give it some room to grow since memory is cheap but allocations can be expensive */
|
|
mixing_array->max_num_entries = num_entries;
|
|
if (!(tmp = ast_realloc(mixing_array->buffers, (mixing_array->max_num_entries * sizeof(int16_t *))))) {
|
|
ast_log(LOG_NOTICE, "Failed to re-allocate softmix mixing structure. \n");
|
|
return -1;
|
|
}
|
|
mixing_array->buffers = tmp;
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Function which acts as the mixing thread */
|
|
static int softmix_bridge_thread(struct ast_bridge *bridge)
|
|
{
|
|
struct softmix_stats stats = { { 0 }, };
|
|
struct softmix_mixing_array mixing_array;
|
|
struct softmix_bridge_data *softmix_data = bridge->bridge_pvt;
|
|
struct ast_timer *timer;
|
|
struct softmix_translate_helper trans_helper;
|
|
int16_t buf[MAX_DATALEN] = { 0, };
|
|
unsigned int stat_iteration_counter = 0; /* counts down, gather stats at zero and reset. */
|
|
int timingfd;
|
|
int update_all_rates = 0; /* set this when the internal sample rate has changed */
|
|
int i, x;
|
|
int res = -1;
|
|
|
|
if (!(softmix_data = bridge->bridge_pvt)) {
|
|
goto softmix_cleanup;
|
|
}
|
|
|
|
ao2_ref(softmix_data, 1);
|
|
timer = softmix_data->timer;
|
|
timingfd = ast_timer_fd(timer);
|
|
softmix_translate_helper_init(&trans_helper, softmix_data->internal_rate);
|
|
ast_timer_set_rate(timer, (1000 / softmix_data->internal_mixing_interval));
|
|
|
|
/* Give the mixing array room to grow, memory is cheap but allocations are expensive. */
|
|
if (softmix_mixing_array_init(&mixing_array, bridge->num + 10)) {
|
|
ast_log(LOG_NOTICE, "Failed to allocate softmix mixing structure. \n");
|
|
goto softmix_cleanup;
|
|
}
|
|
|
|
while (!bridge->stop && !bridge->refresh && bridge->array_num) {
|
|
struct ast_bridge_channel *bridge_channel = NULL;
|
|
int timeout = -1;
|
|
enum ast_format_id cur_slin_id = ast_format_slin_by_rate(softmix_data->internal_rate);
|
|
unsigned int softmix_samples = SOFTMIX_SAMPLES(softmix_data->internal_rate, softmix_data->internal_mixing_interval);
|
|
unsigned int softmix_datalen = SOFTMIX_DATALEN(softmix_data->internal_rate, softmix_data->internal_mixing_interval);
|
|
|
|
if (softmix_datalen > MAX_DATALEN) {
|
|
/* This should NEVER happen, but if it does we need to know about it. Almost
|
|
* all the memcpys used during this process depend on this assumption. Rather
|
|
* than checking this over and over again through out the code, this single
|
|
* verification is done on each iteration. */
|
|
ast_log(LOG_WARNING, "Conference mixing error, requested mixing length greater than mixing buffer.\n");
|
|
goto softmix_cleanup;
|
|
}
|
|
|
|
/* Grow the mixing array buffer as participants are added. */
|
|
if (mixing_array.max_num_entries < bridge->num && softmix_mixing_array_grow(&mixing_array, bridge->num + 5)) {
|
|
goto softmix_cleanup;
|
|
}
|
|
|
|
/* init the number of buffers stored in the mixing array to 0.
|
|
* As buffers are added for mixing, this number is incremented. */
|
|
mixing_array.used_entries = 0;
|
|
|
|
/* These variables help determine if a rate change is required */
|
|
if (!stat_iteration_counter) {
|
|
memset(&stats, 0, sizeof(stats));
|
|
stats.locked_rate = bridge->internal_sample_rate;
|
|
}
|
|
|
|
/* If the sample rate has changed, update the translator helper */
|
|
if (update_all_rates) {
|
|
softmix_translate_helper_change_rate(&trans_helper, softmix_data->internal_rate);
|
|
}
|
|
|
|
/* 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;
|
|
|
|
/* Update the sample rate to match the bridge's native sample rate if necessary. */
|
|
if (update_all_rates) {
|
|
set_softmix_bridge_data(softmix_data->internal_rate, softmix_data->internal_mixing_interval, bridge_channel, 1);
|
|
}
|
|
|
|
/* If stat_iteration_counter is 0, then collect statistics during this mixing interation */
|
|
if (!stat_iteration_counter) {
|
|
gather_softmix_stats(&stats, softmix_data, bridge_channel);
|
|
}
|
|
|
|
/* if the channel is suspended, don't check for audio, but still gather stats */
|
|
if (bridge_channel->suspended) {
|
|
continue;
|
|
}
|
|
|
|
/* Try to get audio from the factory if available */
|
|
ast_mutex_lock(&sc->lock);
|
|
if ((mixing_array.buffers[mixing_array.used_entries] = softmix_process_read_audio(sc, softmix_samples))) {
|
|
mixing_array.used_entries++;
|
|
}
|
|
ast_mutex_unlock(&sc->lock);
|
|
}
|
|
|
|
/* mix it like crazy */
|
|
memset(buf, 0, softmix_datalen);
|
|
for (i = 0; i < mixing_array.used_entries; i++) {
|
|
for (x = 0; x < softmix_samples; x++) {
|
|
ast_slinear_saturated_add(buf + x, mixing_array.buffers[i] + x);
|
|
}
|
|
}
|
|
|
|
/* 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;
|
|
|
|
if (bridge_channel->suspended) {
|
|
continue;
|
|
}
|
|
|
|
ast_mutex_lock(&sc->lock);
|
|
|
|
/* Make SLINEAR write frame from local buffer */
|
|
if (sc->write_frame.subclass.format.id != cur_slin_id) {
|
|
ast_format_set(&sc->write_frame.subclass.format, cur_slin_id, 0);
|
|
}
|
|
sc->write_frame.datalen = softmix_datalen;
|
|
sc->write_frame.samples = softmix_samples;
|
|
memcpy(sc->final_buf, buf, softmix_datalen);
|
|
|
|
/* process the softmix channel's new write audio */
|
|
softmix_process_write_audio(&trans_helper, &bridge_channel->chan->rawwriteformat, sc);
|
|
|
|
/* The frame is now ready for use... */
|
|
sc->have_frame = 1;
|
|
|
|
ast_mutex_unlock(&sc->lock);
|
|
|
|
/* Poke bridged channel thread just in case */
|
|
pthread_kill(bridge_channel->thread, SIGURG);
|
|
}
|
|
|
|
update_all_rates = 0;
|
|
if (!stat_iteration_counter) {
|
|
update_all_rates = analyse_softmix_stats(&stats, softmix_data);
|
|
stat_iteration_counter = SOFTMIX_STAT_INTERVAL;
|
|
}
|
|
stat_iteration_counter--;
|
|
|
|
ao2_unlock(bridge);
|
|
/* cleanup any translation frame data from the previous mixing iteration. */
|
|
softmix_translate_helper_cleanup(&trans_helper);
|
|
/* 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);
|
|
|
|
/* make sure to detect mixing interval changes if they occur. */
|
|
if (bridge->internal_mixing_interval && (bridge->internal_mixing_interval != softmix_data->internal_mixing_interval)) {
|
|
softmix_data->internal_mixing_interval = bridge->internal_mixing_interval;
|
|
ast_timer_set_rate(timer, (1000 / softmix_data->internal_mixing_interval));
|
|
update_all_rates = 1; /* if the interval changes, the rates must be adjusted as well just to be notified new interval.*/
|
|
}
|
|
}
|
|
|
|
res = 0;
|
|
|
|
softmix_cleanup:
|
|
softmix_translate_helper_destroy(&trans_helper);
|
|
softmix_mixing_array_destroy(&mixing_array);
|
|
if (softmix_data) {
|
|
ao2_ref(softmix_data, -1);
|
|
}
|
|
return res;
|
|
}
|
|
|
|
static struct ast_bridge_technology softmix_bridge = {
|
|
.name = "softmix",
|
|
.capabilities = AST_BRIDGE_CAPABILITY_MULTIMIX | AST_BRIDGE_CAPABILITY_THREAD | AST_BRIDGE_CAPABILITY_MULTITHREADED | AST_BRIDGE_CAPABILITY_OPTIMIZE | AST_BRIDGE_CAPABILITY_VIDEO,
|
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.preference = AST_BRIDGE_PREFERENCE_LOW,
|
|
.create = softmix_bridge_create,
|
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.destroy = softmix_bridge_destroy,
|
|
.join = softmix_bridge_join,
|
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.leave = softmix_bridge_leave,
|
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.write = softmix_bridge_write,
|
|
.thread = softmix_bridge_thread,
|
|
.poke = softmix_bridge_poke,
|
|
};
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|
|
|
static int unload_module(void)
|
|
{
|
|
ast_format_cap_destroy(softmix_bridge.format_capabilities);
|
|
return ast_bridge_technology_unregister(&softmix_bridge);
|
|
}
|
|
|
|
static int load_module(void)
|
|
{
|
|
struct ast_format tmp;
|
|
if (!(softmix_bridge.format_capabilities = ast_format_cap_alloc())) {
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
ast_format_cap_add(softmix_bridge.format_capabilities, ast_format_set(&tmp, AST_FORMAT_SLINEAR, 0));
|
|
return ast_bridge_technology_register(&softmix_bridge);
|
|
}
|
|
|
|
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Multi-party software based channel mixing");
|