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-rw-r--r--bridges/bridge_softmix.c712
1 files changed, 560 insertions, 152 deletions
diff --git a/bridges/bridge_softmix.c b/bridges/bridge_softmix.c
index 1ac2780de..eb476932f 100644
--- a/bridges/bridge_softmix.c
+++ b/bridges/bridge_softmix.c
@@ -1,9 +1,10 @@
/*
* Asterisk -- An open source telephony toolkit.
*
- * Copyright (C) 2007, Digium, Inc.
+ * Copyright (C) 2011, Digium, Inc.
*
* Joshua Colp <jcolp@digium.com>
+ * David Vossel <dvossel@digium.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
@@ -21,12 +22,9 @@
* \brief Multi-party software based channel mixing
*
* \author Joshua Colp <jcolp@digium.com>
+ * \author David Vossel <dvossel@digium.com>
*
* \ingroup bridges
- *
- * \todo This bridge operates in 8 kHz mode unless a define is uncommented.
- * This needs to be improved so the bridge moves between the dominant codec as needed depending
- * on channels present in the bridge and transcoding capabilities.
*/
#include "asterisk.h"
@@ -51,20 +49,26 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/slinfactory.h"
#include "asterisk/astobj2.h"
#include "asterisk/timing.h"
+#include "asterisk/translate.h"
-#define MAX_DATALEN 3840
+#define MAX_DATALEN 8096
/*! \brief Interval at which mixing will take place. Valid options are 10, 20, and 40. */
-#define SOFTMIX_INTERVAL 20
+#define DEFAULT_SOFTMIX_INTERVAL 20
/*! \brief Size of the buffer used for sample manipulation */
-#define SOFTMIX_DATALEN(rate) ((rate/50) * (SOFTMIX_INTERVAL / 10))
+#define SOFTMIX_DATALEN(rate, interval) ((rate/50) * (interval / 10))
/*! \brief Number of samples we are dealing with */
-#define SOFTMIX_SAMPLES(rate) (SOFTMIX_DATALEN(rate) / 2)
+#define SOFTMIX_SAMPLES(rate, interval) (SOFTMIX_DATALEN(rate, interval) / 2)
+
+/*! \brief Number of mixing iterations to perform between gathering statistics. */
+#define SOFTMIX_STAT_INTERVAL 100
-/*! \brief Define used to turn on 16 kHz audio support */
-/* #define SOFTMIX_16_SUPPORT */
+/* This is the threshold in ms at which a channel's own audio will stop getting
+ * mixed out its own write audio stream because it is not talking. */
+#define DEFAULT_SOFTMIX_SILENCE_THRESHOLD 2500
+#define DEFAULT_SOFTMIX_TALKING_THRESHOLD 160
/*! \brief Structure which contains per-channel mixing information */
struct softmix_channel {
@@ -73,7 +77,14 @@ struct softmix_channel {
/*! Factory which contains audio read in from the channel */
struct ast_slinfactory factory;
/*! Frame that contains mixed audio to be written out to the channel */
- struct ast_frame frame;
+ struct ast_frame write_frame;
+ /*! Frame that contains mixed audio read from the channel */
+ struct ast_frame read_frame;
+ /*! DSP for detecting silence */
+ struct ast_dsp *dsp;
+ /*! Bit used to indicate if a channel is talking or not. This affects how
+ * the channel's audio is mixed back to it. */
+ int talking:1;
/*! Bit used to indicate that the channel provided audio for this mixing interval */
int have_audio:1;
/*! Bit used to indicate that a frame is available to be written out to the channel */
@@ -87,66 +98,268 @@ struct softmix_channel {
struct softmix_bridge_data {
struct ast_timer *timer;
unsigned int internal_rate;
+ unsigned int internal_mixing_interval;
+};
+
+struct softmix_stats {
+ /*! Each index represents a sample rate used above the internal rate. */
+ unsigned int sample_rates[16];
+ /*! Each index represents the number of channels using the same index in the sample_rates array. */
+ unsigned int num_channels[16];
+ /*! the number of channels above the internal sample rate */
+ unsigned int num_above_internal_rate;
+ /*! the number of channels at the internal sample rate */
+ unsigned int num_at_internal_rate;
+ /*! the absolute highest sample rate supported by any channel in the bridge */
+ unsigned int highest_supported_rate;
+ /*! Is the sample rate locked by the bridge, if so what is that rate.*/
+ unsigned int locked_rate;
};
+struct softmix_mixing_array {
+ int max_num_entries;
+ int used_entries;
+ int16_t **buffers;
+};
+
+struct softmix_translate_helper_entry {
+ int num_times_requested; /*!< Once this entry is no longer requested, free the trans_pvt
+ and re-init if it was usable. */
+ struct ast_format dst_format; /*!< The destination format for this helper */
+ struct ast_trans_pvt *trans_pvt; /*!< the translator for this slot. */
+ struct ast_frame *out_frame; /*!< The output frame from the last translation */
+ AST_LIST_ENTRY(softmix_translate_helper_entry) entry;
+};
+
+struct softmix_translate_helper {
+ struct ast_format slin_src; /*!< the source format expected for all the translators */
+ AST_LIST_HEAD_NOLOCK(, softmix_translate_helper_entry) entries;
+};
+
+static struct softmix_translate_helper_entry *softmix_translate_helper_entry_alloc(struct ast_format *dst)
+{
+ struct softmix_translate_helper_entry *entry;
+ if (!(entry = ast_calloc(1, sizeof(*entry)))) {
+ return NULL;
+ }
+ ast_format_copy(&entry->dst_format, dst);
+ return entry;
+}
+
+static void *softmix_translate_helper_free_entry(struct softmix_translate_helper_entry *entry)
+{
+ if (entry->trans_pvt) {
+ ast_translator_free_path(entry->trans_pvt);
+ }
+ if (entry->out_frame) {
+ ast_frfree(entry->out_frame);
+ }
+ ast_free(entry);
+ return NULL;
+}
+
+static void softmix_translate_helper_init(struct softmix_translate_helper *trans_helper, unsigned int sample_rate)
+{
+ memset(trans_helper, 0, sizeof(*trans_helper));
+ ast_format_set(&trans_helper->slin_src, ast_format_slin_by_rate(sample_rate), 0);
+}
+
+static void softmix_translate_helper_destroy(struct softmix_translate_helper *trans_helper)
+{
+ struct softmix_translate_helper_entry *entry;
+
+ while ((entry = AST_LIST_REMOVE_HEAD(&trans_helper->entries, entry))) {
+ softmix_translate_helper_free_entry(entry);
+ }
+}
+
+static void softmix_translate_helper_change_rate(struct softmix_translate_helper *trans_helper, unsigned int sample_rate)
+{
+ struct softmix_translate_helper_entry *entry;
+
+ ast_format_set(&trans_helper->slin_src, ast_format_slin_by_rate(sample_rate), 0);
+ AST_LIST_TRAVERSE_SAFE_BEGIN(&trans_helper->entries, entry, entry) {
+ if (entry->trans_pvt) {
+ ast_translator_free_path(entry->trans_pvt);
+ if (!(entry->trans_pvt = ast_translator_build_path(&entry->dst_format, &trans_helper->slin_src))) {
+ AST_LIST_REMOVE_CURRENT(entry);
+ entry = softmix_translate_helper_free_entry(entry);
+ }
+ }
+ }
+ AST_LIST_TRAVERSE_SAFE_END;
+}
+
+/*!
+ * \internal
+ * \brief Get the next available audio on the softmix channel's read stream
+ * and determine if it should be mixed out or not on the write stream.
+ *
+ * \retval pointer to buffer containing the exact number of samples requested on success.
+ * \retval NULL if no samples are present
+ */
+static int16_t *softmix_process_read_audio(struct softmix_channel *sc, unsigned int num_samples)
+{
+ if ((ast_slinfactory_available(&sc->factory) >= num_samples) &&
+ ast_slinfactory_read(&sc->factory, sc->our_buf, num_samples)) {
+ sc->have_audio = 1;
+ return sc->our_buf;
+ }
+ sc->have_audio = 0;
+ return NULL;
+}
+
+/*!
+ * \internal
+ * \brief Process a softmix channel's write audio
+ *
+ * \details This function will remove the channel's talking from its own audio if present and
+ * possibly even do the channel's write translation for it depending on how many other
+ * channels use the same write format.
+ */
+static void softmix_process_write_audio(struct softmix_translate_helper *trans_helper,
+ struct ast_format *raw_write_fmt,
+ struct softmix_channel *sc)
+{
+ struct softmix_translate_helper_entry *entry = NULL;
+ int i;
+
+ /* If we provided audio that was not determined to be silence,
+ * then take it out while in slinear format. */
+ if (sc->have_audio && sc->talking) {
+ for (i = 0; i < sc->write_frame.samples; i++) {
+ ast_slinear_saturated_subtract(&sc->final_buf[i], &sc->our_buf[i]);
+ }
+ /* do not do any special write translate optimization if we had to make
+ * a special mix for them to remove their own audio. */
+ return;
+ }
+
+ AST_LIST_TRAVERSE(&trans_helper->entries, entry, entry) {
+ if (ast_format_cmp(&entry->dst_format, raw_write_fmt) == AST_FORMAT_CMP_EQUAL) {
+ entry->num_times_requested++;
+ } else {
+ continue;
+ }
+ if (!entry->trans_pvt && (entry->num_times_requested > 1)) {
+ entry->trans_pvt = ast_translator_build_path(&entry->dst_format, &trans_helper->slin_src);
+ }
+ if (entry->trans_pvt && !entry->out_frame) {
+ entry->out_frame = ast_translate(entry->trans_pvt, &sc->write_frame, 0);
+ }
+ if (entry->out_frame && (entry->out_frame->datalen < MAX_DATALEN)) {
+ ast_format_copy(&sc->write_frame.subclass.format, &entry->out_frame->subclass.format);
+ memcpy(sc->final_buf, entry->out_frame->data.ptr, entry->out_frame->datalen);
+ sc->write_frame.datalen = entry->out_frame->datalen;
+ sc->write_frame.samples = entry->out_frame->samples;
+ }
+ break;
+ }
+
+ /* add new entry into list if this format destination was not matched. */
+ if (!entry && (entry = softmix_translate_helper_entry_alloc(raw_write_fmt))) {
+ AST_LIST_INSERT_HEAD(&trans_helper->entries, entry, entry);
+ }
+}
+
+static void softmix_translate_helper_cleanup(struct softmix_translate_helper *trans_helper)
+{
+ struct softmix_translate_helper_entry *entry = NULL;
+ AST_LIST_TRAVERSE(&trans_helper->entries, entry, entry) {
+ if (entry->out_frame) {
+ ast_frfree(entry->out_frame);
+ entry->out_frame = NULL;
+ }
+ entry->num_times_requested = 0;
+ }
+}
+
+static void softmix_bridge_data_destroy(void *obj)
+{
+ struct softmix_bridge_data *softmix_data = obj;
+ ast_timer_close(softmix_data->timer);
+}
+
/*! \brief Function called when a bridge is created */
static int softmix_bridge_create(struct ast_bridge *bridge)
{
- struct softmix_bridge_data *bridge_data;
+ struct softmix_bridge_data *softmix_data;
- if (!(bridge_data = ast_calloc(1, sizeof(*bridge_data)))) {
+ if (!(softmix_data = ao2_alloc(sizeof(*softmix_data), softmix_bridge_data_destroy))) {
return -1;
}
- if (!(bridge_data->timer = ast_timer_open())) {
- ast_free(bridge_data);
+ if (!(softmix_data->timer = ast_timer_open())) {
+ ao2_ref(softmix_data, -1);
return -1;
}
/* start at 8khz, let it grow from there */
- bridge_data->internal_rate = 8000;
+ softmix_data->internal_rate = 8000;
+ softmix_data->internal_mixing_interval = DEFAULT_SOFTMIX_INTERVAL;
- bridge->bridge_pvt = bridge_data;
+ bridge->bridge_pvt = softmix_data;
return 0;
}
/*! \brief Function called when a bridge is destroyed */
static int softmix_bridge_destroy(struct ast_bridge *bridge)
{
- struct softmix_bridge_data *bridge_data = bridge->bridge_pvt;
+ struct softmix_bridge_data *softmix_data = bridge->bridge_pvt;
if (!bridge->bridge_pvt) {
return -1;
}
- ast_timer_close(bridge_data->timer);
- ast_free(bridge_data);
+ ao2_ref(softmix_data, -1);
+ bridge->bridge_pvt = NULL;
return 0;
}
-static void set_softmix_bridge_data(int rate, struct ast_bridge_channel *bridge_channel, int reset)
+static void set_softmix_bridge_data(int rate, int interval, struct ast_bridge_channel *bridge_channel, int reset)
{
struct softmix_channel *sc = bridge_channel->bridge_pvt;
+ unsigned int channel_read_rate = ast_format_rate(&bridge_channel->chan->rawreadformat);
+
+ ast_mutex_lock(&sc->lock);
if (reset) {
ast_slinfactory_destroy(&sc->factory);
+ ast_dsp_free(sc->dsp);
}
- /* Setup frame parameters */
- sc->frame.frametype = AST_FRAME_VOICE;
-
- ast_format_set(&sc->frame.subclass.format, ast_format_slin_by_rate(rate), 0);
- sc->frame.data.ptr = sc->final_buf;
- sc->frame.datalen = SOFTMIX_DATALEN(rate);
- sc->frame.samples = SOFTMIX_SAMPLES(rate);
+ /* Setup read/write frame parameters */
+ sc->write_frame.frametype = AST_FRAME_VOICE;
+ ast_format_set(&sc->write_frame.subclass.format, ast_format_slin_by_rate(rate), 0);
+ sc->write_frame.data.ptr = sc->final_buf;
+ sc->write_frame.datalen = SOFTMIX_DATALEN(rate, interval);
+ sc->write_frame.samples = SOFTMIX_SAMPLES(rate, interval);
+
+ sc->read_frame.frametype = AST_FRAME_VOICE;
+ ast_format_set(&sc->read_frame.subclass.format, ast_format_slin_by_rate(channel_read_rate), 0);
+ sc->read_frame.data.ptr = sc->our_buf;
+ sc->read_frame.datalen = SOFTMIX_DATALEN(channel_read_rate, interval);
+ sc->read_frame.samples = SOFTMIX_SAMPLES(channel_read_rate, interval);
/* Setup smoother */
- ast_slinfactory_init_with_format(&sc->factory, &sc->frame.subclass.format);
+ ast_slinfactory_init_with_format(&sc->factory, &sc->write_frame.subclass.format);
+
+ /* set new read and write formats on channel. */
+ ast_set_read_format(bridge_channel->chan, &sc->read_frame.subclass.format);
+ ast_set_write_format(bridge_channel->chan, &sc->write_frame.subclass.format);
+
+ /* set up new DSP. This is on the read side only right before the read frame enters the smoother. */
+ sc->dsp = ast_dsp_new_with_rate(channel_read_rate);
+ /* we want to aggressively detect silence to avoid feedback */
+ if (bridge_channel->tech_args.talking_threshold) {
+ ast_dsp_set_threshold(sc->dsp, bridge_channel->tech_args.talking_threshold);
+ } else {
+ ast_dsp_set_threshold(sc->dsp, DEFAULT_SOFTMIX_TALKING_THRESHOLD);
+ }
- ast_set_read_format(bridge_channel->chan, &sc->frame.subclass.format);
- ast_set_write_format(bridge_channel->chan, &sc->frame.subclass.format);
+ ast_mutex_unlock(&sc->lock);
}
/*! \brief Function called when a channel is joined into the bridge */
static int softmix_bridge_join(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel)
{
struct softmix_channel *sc = NULL;
- struct softmix_bridge_data *bridge_data = bridge->bridge_pvt;
+ struct softmix_bridge_data *softmix_data = bridge->bridge_pvt;
/* Create a new softmix_channel structure and allocate various things on it */
if (!(sc = ast_calloc(1, sizeof(*sc)))) {
@@ -159,7 +372,9 @@ static int softmix_bridge_join(struct ast_bridge *bridge, struct ast_bridge_chan
/* Can't forget to record our pvt structure within the bridged channel structure */
bridge_channel->bridge_pvt = sc;
- set_softmix_bridge_data(bridge_data->internal_rate, bridge_channel, 0);
+ set_softmix_bridge_data(softmix_data->internal_rate,
+ softmix_data->internal_mixing_interval ? softmix_data->internal_mixing_interval : DEFAULT_SOFTMIX_INTERVAL,
+ bridge_channel, 0);
return 0;
}
@@ -169,44 +384,102 @@ static int softmix_bridge_leave(struct ast_bridge *bridge, struct ast_bridge_cha
{
struct softmix_channel *sc = bridge_channel->bridge_pvt;
+ if (!(bridge_channel->bridge_pvt)) {
+ return 0;
+ }
+ bridge_channel->bridge_pvt = NULL;
+
/* Drop mutex lock */
ast_mutex_destroy(&sc->lock);
/* Drop the factory */
ast_slinfactory_destroy(&sc->factory);
+ /* Drop the DSP */
+ ast_dsp_free(sc->dsp);
+
/* Eep! drop ourselves */
ast_free(sc);
return 0;
}
+/*!
+ * \internal
+ * \brief If the bridging core passes DTMF to us, then they want it to be distributed out to all memebers. Do that here.
+ */
+static void softmix_pass_dtmf(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel, struct ast_frame *frame)
+{
+ struct ast_bridge_channel *tmp;
+ AST_LIST_TRAVERSE(&bridge->channels, tmp, entry) {
+ if (tmp == bridge_channel) {
+ 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 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. */
/* Only accept audio frames, all others are unsupported */
- if (frame->frametype != AST_FRAME_VOICE) {
+ if (frame->frametype == AST_FRAME_DTMF_END || frame->frametype == AST_FRAME_DTMF_BEGIN) {
+ softmix_pass_dtmf(bridge, bridge_channel, frame);
+ return AST_BRIDGE_WRITE_SUCCESS;
+ } else if (frame->frametype != AST_FRAME_VOICE) {
return AST_BRIDGE_WRITE_UNSUPPORTED;
}
ast_mutex_lock(&sc->lock);
- /* If a frame was provided add it to the smoother */
- if (frame->frametype == AST_FRAME_VOICE && ast_format_is_slinear(&frame->subclass.format)) {
+ ast_dsp_silence(sc->dsp, frame, &totalsilence);
+ 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->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 AST_BRIDGE_WRITE_SUCCESS;
}
@@ -218,7 +491,7 @@ static int softmix_bridge_poke(struct ast_bridge *bridge, struct ast_bridge_chan
ast_mutex_lock(&sc->lock);
if (sc->have_frame) {
- ast_write(bridge_channel->chan, &sc->frame);
+ ast_write(bridge_channel->chan, &sc->write_frame);
sc->have_frame = 0;
}
@@ -227,167 +500,306 @@ static int softmix_bridge_poke(struct ast_bridge *bridge, struct ast_bridge_chan
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 {
- /*! Each index represents a sample rate used above the internal rate. */
- unsigned int sample_rates[8];
- /*! Each index represents the number of channels using the same index in the sample_rates array. */
- unsigned int num_channels[8];
- /*! the number of channels above the internal sample rate */
- unsigned int num_above_internal_rate;
- /*! the number of channels at the internal sample rate */
- unsigned int num_at_internal_rate;
- /*! the absolute highest sample rate supported by any channel in the bridge */
- unsigned int highest_supported_rate;
- } stats;
- struct softmix_bridge_data *bridge_data = bridge->bridge_pvt;
- struct ast_timer *timer = bridge_data->timer;
- int timingfd = ast_timer_fd(timer);
+ 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;
+ int i, x;
+ int res = -1;
- ast_timer_set_rate(timer, (1000 / SOFTMIX_INTERVAL));
+ 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;
- short buf[MAX_DATALEN] = {0, };
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;
+ }
- /* these variables help determine if a rate change is required */
- memset(&stats, 0, sizeof(stats));
- stats.highest_supported_rate = 8000;
+ /* 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;
- int channel_native_rate;
-
- ast_mutex_lock(&sc->lock);
+ /* Update the sample rate to match the bridge's native sample rate if necessary. */
if (update_all_rates) {
- set_softmix_bridge_data(bridge_data->internal_rate, bridge_channel, 1);
+ set_softmix_bridge_data(softmix_data->internal_rate, softmix_data->internal_mixing_interval, bridge_channel, 1);
}
- /* Try to get audio from the factory if available */
- if (ast_slinfactory_available(&sc->factory) >= SOFTMIX_SAMPLES(bridge_data->internal_rate) &&
- ast_slinfactory_read(&sc->factory, sc->our_buf, SOFTMIX_SAMPLES(bridge_data->internal_rate))) {
- short *data1, *data2;
- int i;
-
- /* Put into the local final buffer */
- for (i = 0, data1 = buf, data2 = sc->our_buf; i < SOFTMIX_DATALEN(bridge_data->internal_rate); i++, data1++, data2++)
- ast_slinear_saturated_add(data1, data2);
- /* Yay we have our own audio */
- sc->have_audio = 1;
- } else {
- /* Awww we don't have audio ;( */
- sc->have_audio = 0;
+ /* 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);
}
- /* 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 > bridge_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 == bridge_data->internal_rate) {
- stats.num_at_internal_rate++;
+ /* 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;
- int i = 0;
- /* Copy from local final buffer to our final buffer */
- memcpy(sc->final_buf, buf, sizeof(sc->final_buf));
+ if (bridge_channel->suspended) {
+ continue;
+ }
- /* If we provided audio then take it out */
- if (sc->have_audio) {
- for (i = 0; i < SOFTMIX_DATALEN(bridge_data->internal_rate); i++) {
- ast_slinear_saturated_subtract(&sc->final_buf[i], &sc->our_buf[i]);
- }
+ 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);
}
- /* Re-adjust the internal bridge sample rate if
- * 1. two or more channels support a higher sample rate
- * 2. no channels support the current sample rate or a higher rate
- */
- 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;
-
- /* 1. pick the best sample rate two or more channels support
- * 2. if two or more channels do not support the same rate, pick the
- * lowest sample rate that is still above the internal rate. */
- for (i = 0; ((i < ARRAY_LEN(stats.num_channels)) && stats.num_channels[i]); i++) {
- if ((stats.num_channels[i] >= 2 && (best_index == -1)) ||
- ((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;
- } else if (best_index == -1) {
- best_rate = MIN(best_rate, stats.sample_rates[i]);
- }
- }
-
- ast_debug(1, " Bridge changed from %d To %d\n", bridge_data->internal_rate, best_rate);
- bridge_data->internal_rate = best_rate;
- update_all_rates = 1;
- } else if (!stats.num_at_internal_rate && !stats.num_above_internal_rate) {
- update_all_rates = 1;
- /* in this case, the highest supported rate is actually lower than the internal rate */
- bridge_data->internal_rate = stats.highest_supported_rate;
- ast_debug(1, " Bridge changed from %d to %d\n", bridge_data->internal_rate, stats.highest_supported_rate);
- update_all_rates = 1;
- } else {
- update_all_rates = 0;
+ 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.*/
+ }
}
- return 0;
+ 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,
+ .capabilities = AST_BRIDGE_CAPABILITY_MULTIMIX | AST_BRIDGE_CAPABILITY_THREAD | AST_BRIDGE_CAPABILITY_MULTITHREADED | AST_BRIDGE_CAPABILITY_OPTIMIZE,
.preference = AST_BRIDGE_PREFERENCE_LOW,
.create = softmix_bridge_create,
.destroy = softmix_bridge_destroy,
@@ -410,11 +822,7 @@ static int load_module(void)
if (!(softmix_bridge.format_capabilities = ast_format_cap_alloc())) {
return AST_MODULE_LOAD_DECLINE;
}
-#ifdef SOFTMIX_16_SUPPORT
- ast_format_cap_add(softmix_bridge.format_capabilities, ast_format_set(&tmp, AST_FORMAT_SLINEAR16, 0));
-#else
ast_format_cap_add(softmix_bridge.format_capabilities, ast_format_set(&tmp, AST_FORMAT_SLINEAR, 0));
-#endif
return ast_bridge_technology_register(&softmix_bridge);
}