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Adding mod_gstreamer 

Add gstreamer to environment variables.
This commit is contained in:
networkedaudio 2023-03-20 14:36:59 -05:00
parent 1c4a48595c
commit 569de5382e
8 changed files with 3385 additions and 2 deletions
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19
Freeswitch.2017.sln
View File

@ -1,6 +1,6 @@
Microsoft Visual Studio Solution File, Format Version 12.00
# Visual Studio 15
VisualStudioVersion = 15.0.27130.2010
# Visual Studio Version 17
VisualStudioVersion = 17.5.33502.453
MinimumVisualStudioVersion = 10.0.40219.1
Project("{2150E333-8FDC-42A3-9474-1A3956D46DE8}") = "Codecs", "Codecs", "{F881ADA2-2F1A-4046-9FEB-191D9422D781}"
EndProject
@ -554,6 +554,8 @@ Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "test_switch_core_db", "test
EndProject
Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "test_switch_ivr_originate", "tests\unit\test_switch_ivr_originate.2017.vcxproj", "{69A7464A-9B0D-4804-A108-835229DACF58}"
EndProject
Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "mod_gstreamer", "src\mod\endpoints\mod_gstreamer\mod_gstreamer.vcxproj", "{5AC15CAA-B153-4B45-804A-0CFDF6FA3422}"
EndProject
Global
GlobalSection(SolutionConfigurationPlatforms) = preSolution
All|Win32 = All|Win32
@ -2516,6 +2518,18 @@ Global
{69A7464A-9B0D-4804-A108-835229DACF58}.Release|Win32.Build.0 = Release|Win32
{69A7464A-9B0D-4804-A108-835229DACF58}.Release|x64.ActiveCfg = Release|x64
{69A7464A-9B0D-4804-A108-835229DACF58}.Release|x64.Build.0 = Release|x64
{5AC15CAA-B153-4B45-804A-0CFDF6FA3422}.All|Win32.ActiveCfg = Debug|Win32
{5AC15CAA-B153-4B45-804A-0CFDF6FA3422}.All|Win32.Build.0 = Debug|Win32
{5AC15CAA-B153-4B45-804A-0CFDF6FA3422}.All|x64.ActiveCfg = Debug|x64
{5AC15CAA-B153-4B45-804A-0CFDF6FA3422}.All|x64.Build.0 = Debug|x64
{5AC15CAA-B153-4B45-804A-0CFDF6FA3422}.Debug|Win32.ActiveCfg = Debug|Win32
{5AC15CAA-B153-4B45-804A-0CFDF6FA3422}.Debug|Win32.Build.0 = Debug|Win32
{5AC15CAA-B153-4B45-804A-0CFDF6FA3422}.Debug|x64.ActiveCfg = Debug|x64
{5AC15CAA-B153-4B45-804A-0CFDF6FA3422}.Debug|x64.Build.0 = Debug|x64
{5AC15CAA-B153-4B45-804A-0CFDF6FA3422}.Release|Win32.ActiveCfg = Release|Win32
{5AC15CAA-B153-4B45-804A-0CFDF6FA3422}.Release|Win32.Build.0 = Release|Win32
{5AC15CAA-B153-4B45-804A-0CFDF6FA3422}.Release|x64.ActiveCfg = Release|x64
{5AC15CAA-B153-4B45-804A-0CFDF6FA3422}.Release|x64.Build.0 = Release|x64
EndGlobalSection
GlobalSection(SolutionProperties) = preSolution
HideSolutionNode = FALSE
@ -2714,6 +2728,7 @@ Global
{0B612F84-7533-4DEC-AEDD-5C9CBCF15EAC} = {31C2761D-20E0-4BF8-98B9-E32F0D8DD6E1}
{580675D7-C1C9-4197-AAC5-00F64FAFDE78} = {9388C266-C3FC-468A-92EF-0CBC35941412}
{69A7464A-9B0D-4804-A108-835229DACF58} = {9388C266-C3FC-468A-92EF-0CBC35941412}
{5AC15CAA-B153-4B45-804A-0CFDF6FA3422} = {9460B5F1-0A95-41C4-BEB7-9C2C96459A7C}
EndGlobalSection
GlobalSection(ExtensibilityGlobals) = postSolution
SolutionGuid = {09840DE7-9208-45AA-9667-1A71EE93BD1E}

46
conf/vanilla/autoload_configs/gstreamer.conf.xml Normal file
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@ -0,0 +1,46 @@
<configuration name="gstreamer.conf" description="Gstreamer module">
<settings>
<!--file to play when calls are on hold-->
<param name="hold-file" value="$${hold_music}"/>
<!--Default dialplan and caller-id info -->
<param name="dialplan" value="XML"/>
<param name="cid-name" value="$${outbound_caller_name}"/>
<param name="cid-num" value="$${outbound_caller_id}"/>
<!--audio sample rate and interval -->
<param name="sample-rate" value="48000"/>
<param name="codec-ms" value="20"/>
<param name="channels" value="1"/>
<param name="bit-depth" value="S16LE"/>
<!-- Tx params for the gstreamer pipeline -->
<param name="tx-address" value="239.69.0.119"/>
<param name="tx-port" value="5004" />
<!-- Rx params for the in gstreamer pipeline -->
<param name="rx-address" value="239.69.161.58"/>
<param name="rx-port" value="5004" />
</settings>
<streams>
<stream name="udp1">
<!-- Tx params for the gstreamer pipeline -->
<param name="tx-address" value="239.69.0.119"/>
<param name="tx-port" value="5004" />
<!-- Rx params for the in gstreamer pipeline -->
<param name="rx-address" value="239.69.161.58"/>
<param name="rx-port" value="5004" />
<param name="sample-rate" value="48000"/>
<param name="codec-ms" value="20"/>
<param name="channels" value="1" />
</stream>
</streams>
<endpoints>
<endpoint name="udp1-tx">
<param name="outstream" value="udp1:0" />
</endpoint>
<endpoint name="udp1-rx">
<param name="instream" value="udp1:0" />
</endpoint>
<endpoint name="udp1-txrx">
<param name="outstream" value="udp1:0" />
<param name="instream" value="udp1:0" />
</endpoint>
</endpoints>
</configuration>

19
src/mod/endpoints/mod_gstreamer/Makefile.am Normal file
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@ -0,0 +1,19 @@
include $(top_srcdir)/build/modmake.rulesam
MODNAME=mod_gstreamer
if HAVE_GST
mod_LTLIBRARIES = mod_gstreamer.la
mod_gstreamer_la_SOURCES = mod_gstreamer.c gstreamer_api.c
mod_gstreamer_la_CFLAGS = $(AM_CFLAGS)
mod_gstreamer_la_CPPFLAGS = -I. $(GST_CFLAGS) $(AM_CPPFLAGS)
mod_gstreamer_la_LIBADD = $(GST_LIBS) $(switch_builddir)/libfreeswitch.la
mod_gstreamer_la_LDFLAGS = -avoid-version -module -no-undefined -shared
if ISMAC
#mod_gstreamer_la_LDFLAGS +=
endif
else
install: error
all: error
error:
$(error You must install gstreamer-1.0 and its subprojects to build $(MODNAME))
endif

46
src/mod/endpoints/mod_gstreamer/conf/autoload_configs/gstreamer.conf.xml Normal file
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@ -0,0 +1,46 @@
<configuration name="gstreamer.conf" description="Gstreamer module">
<settings>
<!--file to play when calls are on hold-->
<param name="hold-file" value="$${hold_music}"/>
<!--Default dialplan and caller-id info -->
<param name="dialplan" value="XML"/>
<param name="cid-name" value="$${outbound_caller_name}"/>
<param name="cid-num" value="$${outbound_caller_id}"/>
<!--audio sample rate and interval -->
<param name="sample-rate" value="48000"/>
<param name="codec-ms" value="20"/>
<param name="channels" value="1"/>
<param name="bit-depth" value="S16LE"/>
<!-- Tx params for the gstreamer pipeline -->
<param name="tx-address" value="239.69.0.119"/>
<param name="tx-port" value="5004" />
<!-- Rx params for the in gstreamer pipeline -->
<param name="rx-address" value="239.69.161.58"/>
<param name="rx-port" value="5004" />
</settings>
<streams>
<stream name="udp1">
<!-- Tx params for the gstreamer pipeline -->
<param name="tx-address" value="239.69.0.119"/>
<param name="tx-port" value="5004" />
<!-- Rx params for the in gstreamer pipeline -->
<param name="rx-address" value="239.69.161.58"/>
<param name="rx-port" value="5004" />
<param name="sample-rate" value="48000"/>
<param name="codec-ms" value="20"/>
<param name="channels" value="1" />
</stream>
</streams>
<endpoints>
<endpoint name="udp1-tx">
<param name="outstream" value="udp1:0" />
</endpoint>
<endpoint name="udp1-rx">
<param name="instream" value="udp1:0" />
</endpoint>
<endpoint name="udp1-txrx">
<param name="outstream" value="udp1:0" />
<param name="instream" value="udp1:0" />
</endpoint>
</endpoints>
</configuration>

650
src/mod/endpoints/mod_gstreamer/gstreamer_api.c Normal file
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@ -0,0 +1,650 @@
#include <switch.h>
#include <gst/app/gstappsink.h>
#include <gst/audio/audio-channels.h>
#include <gst/net/net.h>
#include "gstreamer_api.h"
#define ELEMENT_NAME_SIZE 20
#define STR_SIZE 15
#define NAME_ELEMENT(name, element, ch_idx) \
g_snprintf(name, ELEMENT_NAME_SIZE, "%s-ch%u", element, ch_idx)
#define RTP_DEPAY "rx-depay"
#ifdef _WIN32
#define SYNTHETIC_CLOCK_INTERVAL_MS 1000
#else
#define SYNTHETIC_CLOCK_INTERVAL_MS 100
#endif
static gboolean
bus_callback (GstBus * bus, GstMessage * msg, gpointer data)
{
g_stream_t *stream = (g_stream_t *) data;
GstElement *pipeline = (GstElement *) stream->pipeline;
switch (GST_MESSAGE_TYPE (msg)) {
case GST_MESSAGE_EOS:
switch_log_printf (SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO,
"End of stream\n");
gst_element_set_state (pipeline, GST_STATE_NULL);
break;
case GST_MESSAGE_ERROR:{
gchar *debug;
GError *error;
gst_message_parse_error (msg, &error, &debug);
g_free (debug);
switch_log_printf (SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error: %s\n",
error->message);
if (stream->error_cb)
stream->error_cb (error->message, stream);
g_error_free (error);
gst_element_set_state (pipeline, GST_STATE_NULL);
break;
}
case GST_MESSAGE_STATE_CHANGED:{
GstState old, new, pending;
GstObject *pipe = (GstObject *) stream->pipeline;
gst_message_parse_state_changed (msg, &old, &new, &pending);
if (msg->src == (GstObject *) pipe) {
gchar *old_state, *new_state, *transition;
guint len = 0;
old_state = g_strdup (gst_element_state_get_name (old));
new_state = g_strdup (gst_element_state_get_name (new));
len = strlen (old_state) + strlen (new_state) + strlen ("_to_") + 5;
transition = g_malloc0 (len);
switch_log_printf (SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO,
"Pipeline %s changed state from %s to %s\n",
GST_OBJECT_NAME (msg->src), old_state, new_state);
g_snprintf (transition, len, "%s_to_%s", old_state, new_state);
GST_DEBUG_BIN_TO_DOT_FILE (GST_BIN (pipe),
GST_DEBUG_GRAPH_SHOW_ALL, transition);
g_free (old_state);
g_free (new_state);
g_free (transition);
}
break;
}
default:
break;
}
return TRUE;
}
static void
deinterleave_pad_added (GstElement * deinterleave, GstPad * pad,
gpointer userdata)
{
GstElement *pipeline =
GST_ELEMENT (gst_element_get_parent (deinterleave)), *queue;
GstPad *queue_sink_pad;
gchar name[ELEMENT_NAME_SIZE];
gchar *pad_name;
guint ch_idx;
pad_name = gst_pad_get_name (pad);
sscanf (pad_name, "src_%u", &ch_idx);
NAME_ELEMENT (name, "rx-queue", ch_idx);
queue = gst_bin_get_by_name (GST_BIN (pipeline), name);
g_assert_nonnull (queue);
queue_sink_pad = gst_element_get_static_pad (queue, "sink");
if (gst_pad_link (pad, queue_sink_pad) != GST_PAD_LINK_OK) {
switch_log_printf (SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR,
"Failed to link deinterleave %s pad in the rx pipeline", pad_name);
}
gst_object_unref (queue_sink_pad);
gst_object_unref(queue);
GST_DEBUG_BIN_TO_DOT_FILE (GST_BIN (pipeline), GST_DEBUG_GRAPH_SHOW_ALL,
pad_name);
g_free (pad_name);
}
gboolean update_clock (gpointer userdata) {
g_stream_t *stream = (g_stream_t *) userdata;
GstStructure *stats = NULL;
guint32 rtp_timestamp;
GstElement *pipeline;
GstClockTime internal, external;
gdouble r_sq;
GstElement *rtpdepay;
pipeline = (GstElement *) stream->pipeline;
rtpdepay = gst_bin_get_by_name (GST_BIN (pipeline), RTP_DEPAY);
g_object_get (G_OBJECT(rtpdepay), "stats", &stats, NULL);
if (gst_structure_get_uint(stats, "timestamp", &rtp_timestamp) ) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "rtp timestamp in rtpdepay %u\n", rtp_timestamp);
internal = gst_clock_get_internal_time(stream->clock);
external = gst_util_uint64_scale (rtp_timestamp, GST_SECOND, stream->sample_rate);
if (gst_clock_add_observation(stream->clock, internal, external, &r_sq) &&
!g_atomic_int_get (&stream->clock_sync)) {
g_atomic_int_set(&stream->clock_sync, 1);
gst_pipeline_use_clock (GST_PIPELINE (pipeline), stream->clock);
gst_pipeline_set_clock (GST_PIPELINE (pipeline), stream->clock);
}
}
gst_structure_free(stats);
gst_object_unref (rtpdepay);
return G_SOURCE_CONTINUE;
}
g_stream_t *
create_pipeline (pipeline_data_t *data, event_callback_t * error_cb)
{
GstBus *bus;
GstElement *pipeline, *rtp_pay = NULL, *rtpdepay = NULL;
g_stream_t *stream = g_new (g_stream_t, 1);
char fixed_name[25] = { "pipeline" };
char *pipeline_name;
if (data->name)
pipeline_name = data->name;
else
pipeline_name = fixed_name;
pipeline = gst_pipeline_new (pipeline_name);
if (!pipeline) {
switch_log_printf (SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR,
"Failed to create the pipeline\n");
return NULL;
}
if (data->direction & DIRECTION_RX) {
GstElement *udp_source, *appsink, *deinterleave, *rx_audioconv,
*capsfilter, *queue, *identity;
GstCaps *udp_caps = NULL, *rx_caps = NULL;
udp_source = gst_element_factory_make ("udpsrc", "rx-src");
if (data->rx_codec == L16) {
rtpdepay = gst_element_factory_make ("rtpL16depay", RTP_DEPAY);
udp_caps = gst_caps_new_simple ("application/x-rtp",
"clock-rate", G_TYPE_INT, data->sample_rate,
"channels", G_TYPE_INT, data->channels,
"encoding-name", G_TYPE_STRING, "L16",
"media", G_TYPE_STRING, "audio", NULL);
} else {
rtpdepay = gst_element_factory_make ("rtpL24depay", RTP_DEPAY);
udp_caps = gst_caps_new_simple ("application/x-rtp",
"clock-rate", G_TYPE_INT, data->sample_rate,
"channels", G_TYPE_INT, data->channels,
"encoding-name", G_TYPE_STRING, "L24",
"media", G_TYPE_STRING, "audio", NULL);
}
// rtpjitbuf = gst_element_factory_make("rtpjitterbuffer", "rx-jitbuf");
rx_audioconv = gst_element_factory_make ("audioconvert", "rx-aconv");
capsfilter = gst_element_factory_make ("capsfilter", "rx-caps");
/*Always feed S16LE to the FS*/
rx_caps = gst_caps_new_simple ("audio/x-raw",
"channels", G_TYPE_INT, data->channels,
"format", G_TYPE_STRING, "S16LE",
"layout", G_TYPE_STRING, "interleaved", NULL);
g_object_set (capsfilter, "caps", rx_caps, NULL);
gst_caps_unref (rx_caps);
deinterleave = gst_element_factory_make ("deinterleave", "rx-deinterleave");
for (guint ch = 0; ch < data->channels; ch++) {
gchar name[ELEMENT_NAME_SIZE];
NAME_ELEMENT (name, "rx-queue", ch);
queue = gst_element_factory_make ("queue", name);
NAME_ELEMENT (name, "appsink", ch);
appsink = gst_element_factory_make ("appsink", name);
NAME_ELEMENT (name, "identity", ch);
identity = gst_element_factory_make ("identity", name);
if (!queue || !appsink || !identity) {
switch_log_printf (SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR,
"Failed to create receive elements");
continue;
}
g_object_set (appsink, "emit-signals", FALSE, "sync", FALSE,"drop", TRUE,
"max-buffers", 1, NULL);
g_object_set (queue, "max-size-time", 100000000 /* 100 ms */ , "leaky",
2 /* downstream */ , NULL);
g_object_set (identity, "drop-probability", 1.0, NULL);
gst_bin_add (GST_BIN (pipeline), queue);
gst_bin_add (GST_BIN (pipeline), identity);
gst_bin_add (GST_BIN (pipeline), appsink);
gst_element_link_many (queue, identity, appsink, NULL);
// The deinterleave will be linked to the queue dynamically
}
g_signal_connect (deinterleave, "pad-added",
G_CALLBACK (deinterleave_pad_added), NULL);
g_object_set (udp_source, "address", data->rx_ip_addr, "port", data->rx_port,
NULL);
g_object_set (udp_source, "caps", udp_caps, NULL);
gst_caps_unref (udp_caps);
if (!udp_source || !rtpdepay || !rx_audioconv || !capsfilter
|| !deinterleave) {
switch_log_printf (SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR,
"Failed to create rx elements\n");
goto error;
}
gst_bin_add_many (GST_BIN (pipeline), udp_source, rtpdepay, rx_audioconv,
capsfilter, deinterleave, NULL);
if (!gst_element_link_many (udp_source, rtpdepay, rx_audioconv, capsfilter,
deinterleave, NULL)) {
switch_log_printf (SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR,
"Failed to link elements in the rx pipeline");
goto error;
}
}
if (data->direction & DIRECTION_TX) {
GstElement *udpsink, *tx_audioconv, *audiointerleave, *capsfilter;
GstElement *appsrc;
GstCaps *caps = NULL;
audiointerleave =
gst_element_factory_make ("audiointerleave", "audiointerleave");
gst_bin_add (GST_BIN (pipeline), audiointerleave);
g_object_set(audiointerleave, "start-time-selection", GST_AGGREGATOR_START_TIME_SELECTION_FIRST, NULL);
if (data->tx_codec == L16) {
rtp_pay = gst_element_factory_make ("rtpL16pay", "rtp-pay");
} else {
rtp_pay = gst_element_factory_make ("rtpL24pay", "rtp-pay");
}
if (data->ptime_ms != -1.0) {
g_object_set(rtp_pay, "max-ptime", (gint64) (data->ptime_ms * 1000000),
"min-ptime", (gint64) (data->ptime_ms * 1000000), NULL);
}
for (guint ch = 0; ch < data->channels; ch++) {
gchar name[ELEMENT_NAME_SIZE];
gchar pad_name[STR_SIZE];
NAME_ELEMENT (name, "appsrc", ch);
g_snprintf (pad_name, STR_SIZE, "sink_%u", ch);
appsrc = gst_element_factory_make ("appsrc", name);
if (!appsrc) {
switch_log_printf (SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR,
"Failed to create %s \n", name);
continue;
}
/*Always accept S16LE from the FS*/
caps = gst_caps_new_simple ("audio/x-raw",
"rate", G_TYPE_INT, data->sample_rate,
"channels", G_TYPE_INT, 1,
"format", G_TYPE_STRING, "S16LE",
"layout", G_TYPE_STRING, "interleaved",
"channel-mask", GST_TYPE_BITMASK, (guint64) (1 << ch), NULL);
g_object_set (appsrc, "format", GST_FORMAT_TIME, NULL);
g_object_set (appsrc, "do-timestamp", TRUE, NULL);
g_object_set (appsrc, "is-live", TRUE, NULL);
/* Second * 3 allows a little bit of headroom */
g_object_set (appsrc, "max-bytes", data->codec_ms * data->sample_rate * 2 * 3, NULL);
g_object_set (appsrc, "caps", caps, NULL);
gst_caps_unref (caps);
gst_bin_add (GST_BIN (pipeline), appsrc);
if (!gst_element_link_pads (appsrc, "src", audiointerleave, pad_name)) {
switch_log_printf (SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR,
"Failed to link pads of %s with audiointerleave\n", name);
goto error;
}
}
capsfilter = gst_element_factory_make ("capsfilter", "tx-capsf");
tx_audioconv = gst_element_factory_make ("audioconvert", "tx-audioconv");
udpsink = gst_element_factory_make ("udpsink", "tx-sink");
caps = gst_caps_new_simple ("audio/x-raw",
"rate", G_TYPE_INT, data->sample_rate,
"channels", G_TYPE_INT, data->channels,
"format", G_TYPE_STRING, "S16LE",
"layout", G_TYPE_STRING, "interleaved",
"channel-mask", GST_TYPE_BITMASK, (guint64) (1 << data->channels) - 1,
NULL);
g_object_set (capsfilter, "caps", caps, NULL);
gst_caps_unref (caps);
g_object_set (udpsink, "host", data->tx_ip_addr, "port", data->tx_port, NULL);
g_object_set (udpsink, "sync", TRUE, "async", FALSE, NULL);
g_object_set (udpsink, "qos", TRUE, "qos-dscp", 34, NULL);
if (!audiointerleave || !tx_audioconv || !rtp_pay || !udpsink) {
switch_log_printf (SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR,
"Failed to create tx elements\n");
goto error;
}
gst_bin_add_many (GST_BIN (pipeline), capsfilter, tx_audioconv, rtp_pay,
udpsink, NULL);
if (!gst_element_link_many (audiointerleave, capsfilter, tx_audioconv,
rtp_pay, udpsink, NULL)) {
switch_log_printf (SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR,
"Failed to link elements");
goto error;
}
}
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
gst_bus_add_watch (bus, bus_callback, stream);
gst_object_unref (bus);
stream->error_cb = error_cb;
stream->pipeline = GST_PIPELINE (pipeline);
stream->mainloop = g_main_loop_new (NULL, FALSE);
stream->thread = g_thread_new (pipeline_name, start_pipeline, stream);
stream->sample_rate = data->sample_rate;
g_atomic_int_set (&stream->clock_sync, 0);
gst_element_set_start_time(pipeline, GST_CLOCK_TIME_NONE);
gst_element_set_base_time(pipeline, 0);
if (rtp_pay) {
g_object_set (rtp_pay, "timestamp-offset",
gst_util_uint64_scale_int (data->rtp_ts_offset * GST_MSECOND, data->sample_rate, GST_SECOND)
% G_MAXUINT32,
NULL);
}
if (rtpdepay && data->synthetic_ptp) {
stream->clock = g_object_new (GST_TYPE_SYSTEM_CLOCK, "name", "SyntheticPtpClock", NULL);
stream->cb_rx_stats_id =
g_timeout_add_full(G_PRIORITY_DEFAULT, SYNTHETIC_CLOCK_INTERVAL_MS, update_clock, stream, NULL);
/* We'll set the pipeline clock once it's synced */
} else {
stream->clock = NULL;
gst_pipeline_use_clock (GST_PIPELINE(pipeline), data->clock);
g_atomic_int_set (&stream->clock_sync, 1);
}
for (guint ch = 0; ch < MAX_IO_CHANNELS; ch++)
stream->leftover_bytes[ch] = 0;
return stream;
error:
gst_object_unref (pipeline);
g_free (stream);
return NULL;
}
void *
start_pipeline (void *data)
{
g_stream_t *stream = (g_stream_t *) data;
gst_element_set_state (GST_ELEMENT (stream->pipeline), GST_STATE_PLAYING);
GST_DEBUG_BIN_TO_DOT_FILE (GST_BIN (stream->pipeline),
GST_DEBUG_GRAPH_SHOW_ALL, "start-pipeline");
start_mainloop (stream->mainloop);
return NULL;
}
void
stop_pipeline (g_stream_t * stream)
{
GstBus *bus;
GST_DEBUG_BIN_TO_DOT_FILE (GST_BIN (stream->pipeline),
GST_DEBUG_GRAPH_SHOW_ALL, "pipeline-stop");
gst_element_set_state (GST_ELEMENT (stream->pipeline), GST_STATE_NULL);
/* cb_rx_stats_id will be non zero only when
Rx is operational and pipeline clock is not ptp*/
if (stream->cb_rx_stats_id)
g_source_remove(stream->cb_rx_stats_id);
bus = gst_pipeline_get_bus (GST_PIPELINE (stream->pipeline));
gst_bus_remove_watch (bus);
gst_object_unref (bus);
gst_object_unref (stream->pipeline);
if (stream->clock)
gst_object_unref (stream->clock);
teardown_mainloop (stream->mainloop);
g_thread_join (stream->thread);
g_free (stream);
switch_log_printf (SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG,
"Pipeline and mainloop cleaned up\n");
}
void
teardown_mainloop (GMainLoop * mainloop)
{
g_main_loop_quit (mainloop);
g_main_loop_unref (mainloop);
}
void
start_mainloop (GMainLoop * mainloop)
{
switch_log_printf (SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "Running mainloop\n");
g_main_loop_run (mainloop);
}
gboolean
push_buffer (g_stream_t *stream, unsigned char *payload, guint len,
guint ch_idx, switch_timer_t * timer)
{
GstState cur_state = GST_STATE_NULL, pending_state;
GstBuffer *buf;
GstMapInfo info;
GstFlowReturn ret;
gchar name[ELEMENT_NAME_SIZE];
GstElement *appsrc = NULL;
GstPipeline *pipeline = stream->pipeline;
gboolean res = FALSE;
NAME_ELEMENT (name, "appsrc", ch_idx);
appsrc = gst_bin_get_by_name (GST_BIN (pipeline), name);
switch_core_timer_next (timer);
if (appsrc == NULL) {
switch_log_printf (SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG,
"Failed to find appsrc in the pipeline\n");
return FALSE;
}
if (!g_atomic_int_get(&stream->clock_sync)) {
ret = TRUE;
goto done;
}
gst_element_get_state (GST_ELEMENT (pipeline), &cur_state, &pending_state, 0);
if (cur_state != GST_STATE_PAUSED && cur_state != GST_STATE_PLAYING) {
ret = TRUE;
goto done;
}
buf = gst_buffer_new_allocate (NULL, len, NULL);
if (buf == NULL) {
switch_log_printf (SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR,
"Failed to allocate buffer\n");
goto done;
}
if (!gst_buffer_map (buf, &info, GST_MAP_WRITE)) {
switch_log_printf (SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR,
"Failed to get buffer map\n");
goto done;
}
memcpy (info.data, payload, len);
gst_buffer_unmap (buf, &info);
g_signal_emit_by_name (appsrc, "push-buffer", buf, &ret);
// switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "Pushed buffer\n");
gst_buffer_unref (buf);
if (ret == GST_FLOW_ERROR) {
switch_log_printf (SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR,
"Failed to do 'push-buffer' \n");
goto done;
}
res = TRUE;
done:
gst_object_unref (GST_OBJECT (appsrc));
return res;
}
int
pull_buffers (g_stream_t * stream, unsigned char *payload, guint needed_bytes,
guint ch_idx, switch_timer_t * timer)
{
GstState cur_state = GST_STATE_NULL, pending_state;
GstBuffer *buf;
GstSample *sample;
GstMapInfo info;
int total_bytes = 0;
gchar name[ELEMENT_NAME_SIZE];
GstElement *appsink;
NAME_ELEMENT (name, "appsink", ch_idx);
appsink = gst_bin_get_by_name (GST_BIN (stream->pipeline), name);
if (appsink == NULL) {
switch_log_printf (SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR,
"Failed to find appsink in the pipeline\n");
return 0;
}
gst_element_get_state (GST_ELEMENT (stream->pipeline), &cur_state,
&pending_state, 0);
if (cur_state != GST_STATE_PAUSED && cur_state != GST_STATE_PLAYING)
goto out;
if (gst_app_sink_is_eos (GST_APP_SINK (appsink)))
goto out;
// Note: assumes leftover_bytes will never be more than buflen, which is
// likely true (packet is limited to MTU, while buflen is 8192)
if (stream->leftover_bytes[ch_idx]) {
int copy =
stream->leftover_bytes[ch_idx] <=
needed_bytes ? stream->leftover_bytes[ch_idx] : needed_bytes;
memcpy (payload, stream->leftover[ch_idx], copy);
total_bytes += copy;
stream->leftover_bytes[ch_idx] -= copy;
}
while (total_bytes < needed_bytes) {
// switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "pulling buffer\n");
sample =
gst_app_sink_try_pull_sample (GST_APP_SINK (appsink),
10 * GST_MSECOND);
if (!sample) {
// switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "Failed to pull sample\n");
switch_cond_next ();
break;
}
buf = gst_sample_get_buffer (sample);
if (!buf)
continue;
if (gst_buffer_map (buf, &info, GST_MAP_READ)) {
if (total_bytes + info.size > needed_bytes) {
int want = needed_bytes - total_bytes;
stream->leftover_bytes[ch_idx] = info.size - want;
memcpy (stream->leftover[ch_idx], info.data + want,
stream->leftover_bytes[ch_idx]);
info.size = want;
}
memcpy (payload + total_bytes, info.data, info.size);
total_bytes += info.size;
}
gst_buffer_unmap (buf, &info);
gst_sample_unref (sample);
// switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "Got %d\n", total_bytes);
}
#if 0
{
// Dump data to file
char name[100];
int fd;
NAME_ELEMENT (name, "/tmp/raw", ch_idx);
fd = open (name, O_WRONLY | O_CREAT | O_APPEND);
write (fd, payload, total_bytes);
close (fd);
}
#endif
// switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "%u Returning needed %d, total_bytes: %d\n", ch_idx, needed_bytes, total_bytes);
// switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "Leftover %lu\n", stream->leftover_bytes[ch_idx]);
out:
gst_object_unref(appsink);
return total_bytes;
}
void
drop_input_buffers (gboolean drop, g_stream_t * stream, guint32 ch_idx)
{
gchar name[ELEMENT_NAME_SIZE];
GstElement *identity;
NAME_ELEMENT (name, "identity", ch_idx);
identity = gst_bin_get_by_name (GST_BIN (stream->pipeline), name);
if (identity == NULL) {
switch_log_printf (SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR,
"Failed to get identity element in the pipeline\n");
return;
}
g_object_set (identity, "drop-probability", drop ? 1.0 : 0.0, NULL);
g_snprintf (name, STR_SIZE, "drop-ch%d-%d", ch_idx, drop);
GST_DEBUG_BIN_TO_DOT_FILE (GST_BIN (stream->pipeline),
GST_DEBUG_GRAPH_SHOW_ALL, name);
gst_object_unref(identity);
}

67
src/mod/endpoints/mod_gstreamer/gstreamer_api.h Normal file
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@ -0,0 +1,67 @@
#ifndef __GSTREAMER_API__
#define __GSTREAMER_API__
#include <gst/gst.h>
#include <switch.h>
#define DIRECTION_TX 1 << 0
#define DIRECTION_RX 1 << 1
#define IP_ADDR_MAX_LEN 20
#define AUDIO_FMT_STR_LEN 10
#define MAX_IO_CHANNELS 256
typedef enum
{ L16, L24 } aes67_codec_t;
typedef struct g_stream g_stream_t;
typedef void event_callback_t (gchar * error_msg, g_stream_t * stream);
typedef struct
{
char rx_ip_addr[IP_ADDR_MAX_LEN];
int rx_port;
char tx_ip_addr[IP_ADDR_MAX_LEN];
int tx_port;
int direction;
int sample_rate;
char bit_depth[AUDIO_FMT_STR_LEN];
int channels;
aes67_codec_t tx_codec;
aes67_codec_t rx_codec;
int codec_ms;
char *name;
double ptime_ms;
GstClock *clock;
gint synthetic_ptp;
double rtp_ts_offset;
} pipeline_data_t;
struct g_stream
{
GstPipeline *pipeline;
GMainLoop *mainloop;
GThread *thread;
unsigned char leftover[MAX_IO_CHANNELS][SWITCH_RECOMMENDED_BUFFER_SIZE];
size_t leftover_bytes[MAX_IO_CHANNELS];
event_callback_t *error_cb;
guint cb_rx_stats_id;
volatile gint clock_sync;
GstClock *clock;
gint sample_rate;
};
g_stream_t *create_pipeline (pipeline_data_t *data, event_callback_t * error_cb);
void *start_pipeline (void *data);
void stop_pipeline (g_stream_t * pipeline);
void teardown_mainloop (GMainLoop * loop);
void start_mainloop (GMainLoop * loop);
gboolean push_buffer (g_stream_t *stream, unsigned char *payload, guint len,
guint ch_idx, switch_timer_t * timer);
int pull_buffers (g_stream_t * stream, unsigned char *payload, guint buflen,
guint ch_idx, switch_timer_t * timer);
void drop_input_buffers (gboolean drop, g_stream_t * stream, guint32 ch_idx);
#endif /*__GSTREAMER_API__*/

2333
src/mod/endpoints/mod_gstreamer/mod_gstreamer.c Normal file
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File diff suppressed because it is too large Load Diff

207
src/mod/endpoints/mod_gstreamer/mod_gstreamer.vcxproj Normal file
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