Review: https://reviewboard.asterisk.org/r/1770/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356573 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(ast_channel_rawreadformat(bridge_channel->chan));
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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) {
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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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/*! \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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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(ast_channel_rawwriteformat(bridge_channel->chan)),
|
|
ast_format_rate(ast_channel_rawreadformat(bridge_channel->chan)));
|
|
|
|
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, ast_channel_rawwriteformat(bridge_channel->chan), 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,
|
|
.preference = AST_BRIDGE_PREFERENCE_LOW,
|
|
.create = softmix_bridge_create,
|
|
.destroy = softmix_bridge_destroy,
|
|
.join = softmix_bridge_join,
|
|
.leave = softmix_bridge_leave,
|
|
.write = softmix_bridge_write,
|
|
.thread = softmix_bridge_thread,
|
|
.poke = softmix_bridge_poke,
|
|
};
|
|
|
|
static int unload_module(void)
|
|
{
|
|
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");
|