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Purism-Calls/plugins/sip/calls-sip-media-pipeline.c

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/*
* Copyright (C) 2021 Purism SPC
*
* This file is part of Calls.
*
* Calls is free software: you can redistribute it and/or modify it
* under the terms of the GNU General Public License as published by
* the Free Software Foundation, either version 3 of the License, or
* (at your option) any later version.
*
* Calls is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with Calls. If not, see <http://www.gnu.org/licenses/>.
*
* Author: Evangelos Ribeiro Tzaras <evangelos.tzaras@puri.sm>
*
* SPDX-License-Identifier: GPL-3.0-or-later
*
*/
#define G_LOG_DOMAIN "CallsSipMediaPipeline"
#include "calls-sip-media-pipeline.h"
#include <gst/gst.h>
#include <gio/gio.h>
/**
* SECTION:sip-media-pipeline
* @short_description:
* @Title:
*
* #CallsSipMediaPipeline is responsible for building Gstreamer pipelines.
* Usually a sender and receiver pipeline is employed.
*
* The sender pipeline records audio and uses RTP to send it out over the network
* to the specified host.
* The receiver pipeline receives RTP from the network and plays the audio
* on the system.
*
* Both pipelines are using RTCP.
*/
enum {
PROP_0,
PROP_CODEC,
PROP_REMOTE,
PROP_LPORT_RTP,
PROP_RPORT_RTP,
PROP_LPORT_RTCP,
PROP_RPORT_RTCP,
PROP_DEBUG,
PROP_LAST_PROP,
};
static GParamSpec *props[PROP_LAST_PROP];
struct _CallsSipMediaPipeline {
GObject parent;
MediaCodecInfo *codec;
gboolean debug;
/* Connection details */
char *remote;
gint rport_rtp;
gint lport_rtp;
gint rport_rtcp;
gint lport_rtcp;
gboolean is_running;
/* Gstreamer Elements (sending) */
GstElement *send_pipeline;
GstElement *audiosrc;
GstElement *send_rtpbin;
GstElement *rtp_sink; /* UDP out */
GstElement *payloader;
GstElement *encoder;
GstElement *rtcp_send_sink;
GstElement *rtcp_send_src;
/* Gstreamer elements (receiving) */
GstElement *recv_pipeline;
GstElement *audiosink;
GstElement *recv_rtpbin;
GstElement *rtp_src; /* UDP in */
GstElement *depayloader;
GstElement *decoder;
GstElement *rtcp_recv_sink;
GstElement *rtcp_recv_src;
/* Gstreamer busses */
GstBus *bus_send;
GstBus *bus_recv;
guint bus_watch_send;
guint bus_watch_recv;
};
static void initable_iface_init (GInitableIface *iface);
G_DEFINE_TYPE_WITH_CODE (CallsSipMediaPipeline, calls_sip_media_pipeline, G_TYPE_OBJECT,
G_IMPLEMENT_INTERFACE (G_TYPE_INITABLE, initable_iface_init));
/* rtpbin adds a pad once the payload is verified */
static void
on_pad_added (GstElement *rtpbin,
GstPad *srcpad,
GstElement *depayloader)
{
GstPad *sinkpad;
g_debug ("pad added: %s", GST_PAD_NAME (srcpad));
sinkpad = gst_element_get_static_pad (depayloader, "sink");
if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK)
g_warning ("Failed to link rtpbin to depayloader");
gst_object_unref (sinkpad);
}
static gboolean
on_bus_message (GstBus *bus,
GstMessage *message,
gpointer data)
{
CallsSipMediaPipeline *pipeline = CALLS_SIP_MEDIA_PIPELINE (data);
switch (GST_MESSAGE_TYPE (message)) {
case GST_MESSAGE_ERROR:
{
g_autoptr (GError) error = NULL;
g_autofree char *msg = NULL;
gst_message_parse_error (message, &error, &msg);
g_warning ("Error on the message bus: %s (%s)", error->message, msg);
break;
}
case GST_MESSAGE_WARNING:
{
g_autoptr (GError) error = NULL;
g_autofree char *msg = NULL;
gst_message_parse_warning (message, &error, &msg);
g_warning ("Warning on the message bus: %s (%s)", error->message, msg);
break;
}
case GST_MESSAGE_EOS:
g_debug ("Received end of stream");
calls_sip_media_pipeline_stop (pipeline);
break;
case GST_MESSAGE_STATE_CHANGED:
{
GstState oldstate;
GstState newstate;
gst_message_parse_state_changed (message, &oldstate, &newstate, NULL);
g_debug ("Element %s has changed state from %s to %s",
GST_OBJECT_NAME (message->src),
gst_element_state_get_name (oldstate),
gst_element_state_get_name (newstate));
break;
}
default:
if (pipeline->debug)
g_debug ("Got unhandled %s message", GST_MESSAGE_TYPE_NAME (message));
break;
}
/* keep watching for messages on the bus */
return TRUE;
}
static void
calls_sip_media_pipeline_get_property (GObject *object,
guint property_id,
GValue *value,
GParamSpec *pspec)
{
CallsSipMediaPipeline *self = CALLS_SIP_MEDIA_PIPELINE (object);
switch (property_id) {
case PROP_CODEC:
g_value_set_pointer (value, self->codec);
break;
case PROP_REMOTE:
g_value_set_string (value, self->remote);
break;
case PROP_LPORT_RTP:
g_value_set_uint (value, self->lport_rtp);
break;
case PROP_LPORT_RTCP:
g_value_set_uint (value, self->lport_rtcp);
break;
case PROP_RPORT_RTP:
g_value_set_uint (value, self->rport_rtp);
break;
case PROP_RPORT_RTCP:
g_value_set_uint (value, self->rport_rtcp);
break;
case PROP_DEBUG:
g_value_set_boolean (value, self->debug);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
}
}
static void
calls_sip_media_pipeline_set_property (GObject *object,
guint property_id,
const GValue *value,
GParamSpec *pspec)
{
CallsSipMediaPipeline *self = CALLS_SIP_MEDIA_PIPELINE (object);
switch (property_id) {
case PROP_CODEC:
self->codec = g_value_get_pointer (value);
break;
case PROP_REMOTE:
g_free (self->remote);
self->remote = g_value_dup_string (value);
break;
case PROP_LPORT_RTP:
self->lport_rtp = g_value_get_uint (value);
break;
case PROP_LPORT_RTCP:
self->lport_rtcp = g_value_get_uint (value);
break;
case PROP_RPORT_RTP:
self->rport_rtp = g_value_get_uint (value);
break;
case PROP_RPORT_RTCP:
self->rport_rtcp = g_value_get_uint (value);
break;
case PROP_DEBUG:
self->debug = g_value_get_boolean (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
}
}
static void
calls_sip_media_pipeline_finalize (GObject *object)
{
CallsSipMediaPipeline *self = CALLS_SIP_MEDIA_PIPELINE (object);
calls_sip_media_pipeline_stop (self);
gst_object_unref (self->send_pipeline);
gst_object_unref (self->recv_pipeline);
gst_bus_remove_watch (self->bus_send);
gst_object_unref (self->bus_send);
gst_bus_remove_watch (self->bus_recv);
gst_object_unref (self->bus_recv);
g_free (self->remote);
G_OBJECT_CLASS (calls_sip_media_pipeline_parent_class)->finalize (object);
}
static void
calls_sip_media_pipeline_class_init (CallsSipMediaPipelineClass *klass)
{
GObjectClass *object_class = G_OBJECT_CLASS (klass);
object_class->set_property = calls_sip_media_pipeline_set_property;
object_class->get_property = calls_sip_media_pipeline_get_property;
object_class->finalize = calls_sip_media_pipeline_finalize;
/* Maybe we want to turn Codec into a GObject later */
props[PROP_CODEC] = g_param_spec_pointer ("codec",
"Codec",
"Media codec",
G_PARAM_CONSTRUCT_ONLY | G_PARAM_READWRITE);
props[PROP_REMOTE] = g_param_spec_string ("remote",
"Remote",
"Remote host",
NULL,
G_PARAM_READWRITE);
props[PROP_LPORT_RTP] = g_param_spec_uint ("lport-rtp",
"lport-rtp",
"local rtp port",
1025, 65535, 5002,
G_PARAM_READWRITE);
props[PROP_LPORT_RTCP] = g_param_spec_uint ("lport-rtcp",
"lport-rtcp",
"local rtcp port",
1025, 65535, 5003,
G_PARAM_READWRITE);
props[PROP_RPORT_RTP] = g_param_spec_uint ("rport-rtp",
"rport-rtp",
"remote rtp port",
1025, 65535, 5002,
G_PARAM_READWRITE);
props[PROP_RPORT_RTCP] = g_param_spec_uint ("rport-rtcp",
"rport-rtcp",
"remote rtcp port",
1025, 65535, 5003,
G_PARAM_READWRITE);
props[PROP_DEBUG] = g_param_spec_boolean ("debug",
"Debug",
"Enable debugging information",
FALSE,
G_PARAM_READWRITE);
g_object_class_install_properties (object_class, PROP_LAST_PROP, props);
}
static void
calls_sip_media_pipeline_init (CallsSipMediaPipeline *self)
{
if (!gst_is_initialized ())
gst_init (NULL, NULL);
}
static gboolean
initable_init (GInitable *initable,
GCancellable *cancelable,
GError **error)
{
CallsSipMediaPipeline *self = CALLS_SIP_MEDIA_PIPELINE (initable);
g_autoptr (GstCaps) caps = NULL;
g_autofree char *caps_string = NULL;
GstPad *srcpad, *sinkpad;
GstStructure *gst_props = NULL;
const char *env_var;
env_var = g_getenv ("CALLS_AUDIOSINK");
if (env_var) {
self->audiosink = gst_element_factory_make (env_var, "sink");
} else {
/* could also use autoaudiosink instead of pulsesink */
self->audiosink = gst_element_factory_make ("pulsesink", "sink");
/* enable echo cancellation and set buffer size to 40ms */
gst_props = gst_structure_new ("props",
"media.role", G_TYPE_STRING, "phone",
"filter.want", G_TYPE_STRING, "echo-cancel",
NULL);
g_object_set (self->audiosink,
"buffer-time", (gint64) 40000,
"stream-properties", gst_props,
NULL);
gst_structure_free (gst_props);
}
env_var = g_getenv ("CALLS_AUDIOSRC");
if (env_var) {
self->audiosrc = gst_element_factory_make (env_var, "source");
} else {
/* could also use autoaudiosrc instead of pulsesrc */
self->audiosrc = gst_element_factory_make ("pulsesrc", "source");
/* enable echo cancellation and set buffer size to 40ms */
gst_props = gst_structure_new ("props",
"media.role", G_TYPE_STRING, "phone",
"filter.want", G_TYPE_STRING, "echo-cancel",
NULL);
g_object_set (self->audiosrc,
"buffer-time", (gint64) 40000,
"stream-properties", gst_props,
NULL);
gst_structure_free (gst_props);
}
if (!self->audiosrc || !self->audiosink) {
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Could not create audiosink or audiosrc");
return FALSE;
}
/* maybe we need to also explicitly add audioconvert and audioresample elements */
self->send_rtpbin = gst_element_factory_make ("rtpbin", "send-rtpbin");
self->recv_rtpbin = gst_element_factory_make ("rtpbin", "recv-rtpbin");
if (!self->send_rtpbin || !self->recv_rtpbin) {
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Could not create send/receive rtpbin");
return FALSE;
}
self->decoder = gst_element_factory_make (self->codec->gst_decoder_name, "decoder");
if (!self->decoder) {
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Could not create decoder %s", self->codec->gst_decoder_name);
return FALSE;
}
self->depayloader = gst_element_factory_make (self->codec->gst_depayloader_name, "depayloader");
if (!self->depayloader) {
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Could not create depayloader %s", self->codec->gst_depayloader_name);
return FALSE;
}
self->encoder = gst_element_factory_make (self->codec->gst_encoder_name, "encoder");
if (!self->encoder) {
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Could not create encoder %s", self->codec->gst_encoder_name);
return FALSE;
}
self->payloader = gst_element_factory_make (self->codec->gst_payloader_name, "payloader");
if (!self->encoder) {
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Could not create payloader %s", self->codec->gst_payloader_name);
return FALSE;
}
self->rtp_src = gst_element_factory_make ("udpsrc", "rtp-udp-src");
self->rtp_sink = gst_element_factory_make ("udpsink", "rtp-udp-sink");
self->rtcp_recv_sink = gst_element_factory_make ("udpsink", "rtcp-udp-recv-sink");
self->rtcp_recv_src = gst_element_factory_make ("udpsrc", "rtcp-udp-recv-src");
self->rtcp_send_sink = gst_element_factory_make ("udpsink", "rtcp-udp-send-sink");
self->rtcp_send_src = gst_element_factory_make ("udpsrc", "rtcp-udp-send-src");
if (!self->rtp_src || !self->rtp_sink ||
!self->rtcp_recv_sink || !self->rtcp_recv_src ||
!self->rtcp_send_sink || !self->rtcp_send_src) {
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Could not create udp sinks or sources");
return FALSE;
}
self->send_pipeline = gst_pipeline_new ("rtp-send-pipeline");
self->recv_pipeline = gst_pipeline_new ("rtp-recv-pipeline");
if (!self->send_pipeline || !self->recv_pipeline) {
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Could not create send or receiver pipeline");
return FALSE;
}
gst_object_ref_sink (self->send_pipeline);
gst_object_ref_sink (self->recv_pipeline);
/* get the busses and establish watches */
self->bus_send = gst_pipeline_get_bus (GST_PIPELINE (self->send_pipeline));
self->bus_recv = gst_pipeline_get_bus (GST_PIPELINE (self->recv_pipeline));
self->bus_watch_send = gst_bus_add_watch (self->bus_send, on_bus_message, self);
self->bus_watch_recv = gst_bus_add_watch (self->bus_recv, on_bus_message, self);
gst_bin_add_many (GST_BIN (self->recv_pipeline), self->depayloader, self->decoder,
self->audiosink, NULL);
gst_bin_add_many (GST_BIN (self->send_pipeline), self->payloader, self->encoder,
self->audiosrc, NULL);
if (!gst_element_link_many (self->depayloader, self->decoder, self->audiosink, NULL)) {
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Failed to link depayloader decoder and audiosink");
return FALSE;
}
if (!gst_element_link_many (self->audiosrc, self->encoder, self->payloader, NULL)) {
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Failed to link audiosrc encoder and payloader");
return FALSE;
}
gst_bin_add (GST_BIN (self->send_pipeline), self->send_rtpbin);
gst_bin_add (GST_BIN (self->recv_pipeline), self->recv_rtpbin);
gst_bin_add_many (GST_BIN (self->send_pipeline), self->rtp_sink,
self->rtcp_send_src, self->rtcp_send_sink, NULL);
gst_bin_add_many (GST_BIN (self->recv_pipeline), self->rtp_src,
self->rtcp_recv_src, self->rtcp_recv_sink, NULL);
caps_string = media_codec_get_gst_capabilities (self->codec);
g_debug ("Capabilities:\n%s", caps_string);
caps = gst_caps_from_string (caps_string);
/* set udp sinks and sources for RTP and RTCP */
g_object_set (self->rtp_src,
"caps", caps,
NULL);
g_object_set (self->rtcp_recv_sink,
"async", FALSE,
"sync", FALSE,
NULL);
g_object_set (self->rtcp_send_sink,
"async", FALSE,
"sync", FALSE,
NULL);
/* bind to properties of udp sinks and sources */
/* Receiver side */
if (self->remote == NULL)
self->remote = g_strdup ("localhost");
g_object_bind_property (self, "lport-rtp",
self->rtp_src, "port",
G_BINDING_BIDIRECTIONAL);
g_object_bind_property (self, "lport-rtcp",
self->rtcp_recv_src, "port",
G_BINDING_BIDIRECTIONAL);
g_object_bind_property (self, "rport-rtcp",
self->rtcp_recv_sink, "port",
G_BINDING_BIDIRECTIONAL);
g_object_bind_property (self, "remote",
self->rtcp_recv_sink, "host",
G_BINDING_BIDIRECTIONAL);
/* Sender side */
g_object_bind_property (self, "rport-rtp",
self->rtp_sink, "port",
G_BINDING_BIDIRECTIONAL);
g_object_bind_property (self, "remote",
self->rtp_sink, "host",
G_BINDING_BIDIRECTIONAL);
g_object_bind_property (self, "lport-rtcp",
self->rtcp_send_src, "port",
G_BINDING_BIDIRECTIONAL);
g_object_bind_property (self, "rport-rtcp",
self->rtcp_send_sink, "port",
G_BINDING_BIDIRECTIONAL);
g_object_bind_property (self, "remote",
self->rtcp_send_sink, "host",
G_BINDING_BIDIRECTIONAL);
/* Link pads */
/* in/receive direction */
/* request and link the pads */
srcpad = gst_element_get_static_pad (self->rtp_src, "src");
#if GST_CHECK_VERSION (1, 20, 0)
sinkpad = gst_element_request_pad_simple (self->recv_rtpbin, "recv_rtp_sink_0");
#else
sinkpad = gst_element_get_request_pad (self->recv_rtpbin, "recv_rtp_sink_0");
#endif
if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) {
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Failed to link rtpsrc to rtpbin");
goto err;
}
gst_object_unref (srcpad);
gst_object_unref (sinkpad);
srcpad = gst_element_get_static_pad (self->rtcp_recv_src, "src");
#if GST_CHECK_VERSION (1, 20 , 0)
sinkpad = gst_element_request_pad_simple (self->recv_rtpbin, "recv_rtcp_sink_0");
#else
sinkpad = gst_element_get_request_pad (self->recv_rtpbin, "recv_rtcp_sink_0");
#endif
if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) {
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Failed to link rtcpsrc to rtpbin");
goto err;
}
gst_object_unref (srcpad);
gst_object_unref (sinkpad);
#if GST_CHECK_VERSION (1, 20, 0)
srcpad = gst_element_request_pad_simple (self->recv_rtpbin, "send_rtcp_src_0");
#else
srcpad = gst_element_get_request_pad (self->recv_rtpbin, "send_rtcp_src_0");
#endif
sinkpad = gst_element_get_static_pad (self->rtcp_recv_sink, "sink");
if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) {
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Failed to link rtpbin to rtcpsink");
goto err;
}
gst_object_unref (srcpad);
gst_object_unref (sinkpad);
/* need to link RTP pad to the depayloader */
g_signal_connect (self->recv_rtpbin, "pad-added", G_CALLBACK (on_pad_added), self->depayloader);
/* out/send direction */
/* link payloader src to RTP sink pad */
#if GST_CHECK_VERSION (1, 20, 0)
sinkpad = gst_element_request_pad_simple (self->send_rtpbin, "send_rtp_sink_0");
#else
sinkpad = gst_element_get_request_pad (self->send_rtpbin, "send_rtp_sink_0");
#endif
srcpad = gst_element_get_static_pad (self->payloader, "src");
if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) {
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Failed to link payloader to rtpbin");
goto err;
}
gst_object_unref (srcpad);
gst_object_unref (sinkpad);
/* link RTP srcpad to udpsink */
srcpad = gst_element_get_static_pad (self->send_rtpbin, "send_rtp_src_0");
sinkpad = gst_element_get_static_pad (self->rtp_sink, "sink");
if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) {
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Failed to link rtpbin to rtpsink");
goto err;
}
gst_object_unref (srcpad);
gst_object_unref (sinkpad);
/* RTCP srcpad to udpsink */
#if GST_CHECK_VERSION (1, 20, 0)
srcpad = gst_element_request_pad_simple (self->send_rtpbin, "send_rtcp_src_0");
#else
srcpad = gst_element_get_request_pad (self->send_rtpbin, "send_rtcp_src_0");
#endif
sinkpad = gst_element_get_static_pad (self->rtcp_send_sink, "sink");
if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) {
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Failed to link rtpbin to rtcpsink");
goto err;
}
gst_object_unref (srcpad);
gst_object_unref (sinkpad);
/* receive RTCP */
srcpad = gst_element_get_static_pad (self->rtcp_send_src, "src");
#if GST_CHECK_VERSION (1, 20, 0)
sinkpad = gst_element_request_pad_simple (self->send_rtpbin, "recv_rtcp_sink_0");
#else
sinkpad = gst_element_get_request_pad (self->send_rtpbin, "recv_rtcp_sink_0");
#endif
if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) {
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Failed to link rtcpsrc to rtpbin");
goto err;
}
gst_object_unref (srcpad);
gst_object_unref (sinkpad);
return TRUE;
err:
gst_object_unref (srcpad);
gst_object_unref (sinkpad);
return FALSE;
}
static void
initable_iface_init (GInitableIface *iface)
{
iface->init = initable_init;
}
CallsSipMediaPipeline*
calls_sip_media_pipeline_new (MediaCodecInfo *codec)
{
CallsSipMediaPipeline *pipeline;
g_autoptr (GError) error = NULL;
pipeline = g_initable_new (CALLS_TYPE_SIP_MEDIA_PIPELINE, NULL, &error,
"codec", codec,
NULL);
if (pipeline == NULL)
g_warning ("Media pipeline could not be initialized: %s", error->message);
return pipeline;
}
static void
diagnose_used_ports_in_socket (GSocket *socket)
{
g_autoptr (GSocketAddress) local_addr = NULL;
g_autoptr (GSocketAddress) remote_addr = NULL;
guint16 local_port;
guint16 remote_port;
local_addr = g_socket_get_local_address (socket, NULL);
remote_addr = g_socket_get_remote_address (socket, NULL);
if (!local_addr) {
g_warning ("Could not get local address of socket");
return;
}
g_assert (G_IS_INET_SOCKET_ADDRESS (local_addr));
local_port = g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (local_addr));
g_debug ("Using local port %d", local_port);
if (!remote_addr) {
g_warning ("Could not get remote address of socket");
return;
}
g_assert (G_IS_INET_SOCKET_ADDRESS (remote_addr));
remote_port = g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (remote_addr));
g_debug ("Using remote port %d", remote_port);
}
static void
diagnose_ports_in_use (CallsSipMediaPipeline *self)
{
GSocket *socket_in;
GSocket *socket_out;
gboolean same_socket = FALSE;
g_assert (CALLS_IS_SIP_MEDIA_PIPELINE (self));
g_assert (self->is_running);
g_object_get (self->rtp_src, "used-socket", &socket_in, NULL);
g_object_get (self->rtp_sink, "used-socket", &socket_out, NULL);
if (socket_in == NULL || socket_out == NULL) {
g_warning ("Could not get used socket");
return;
}
same_socket = socket_in == socket_out;
if (same_socket) {
g_debug ("Diagnosing bidirectional socket...");
diagnose_used_ports_in_socket (socket_in);
}
else {
g_debug ("Diagnosing server socket...");
diagnose_used_ports_in_socket (socket_in);
g_debug ("Diagnosing client socket...");
diagnose_used_ports_in_socket (socket_out);
}
}
void
calls_sip_media_pipeline_start (CallsSipMediaPipeline *self)
{
GSocket *socket;
g_return_if_fail (CALLS_IS_SIP_MEDIA_PIPELINE (self));
g_debug ("Starting media pipeline");
self->is_running = TRUE;
/* First start the receiver pipeline so that
we may reuse the socket in the sender pipeline */
/* TODO can we do something similar for RTCP? */
gst_element_set_state (self->recv_pipeline, GST_STATE_PLAYING);
g_object_get (self->rtp_src, "used-socket", &socket, NULL);
if (socket) {
g_object_set (self->rtp_sink,
"close-socket", FALSE,
"socket", socket,
NULL);
}
else
g_warning ("Could not get used socket of udpsrc element");
/* Now start the sender pipeline */
gst_element_set_state (self->send_pipeline, GST_STATE_PLAYING);
if (self->debug)
diagnose_ports_in_use (self);
}
void
calls_sip_media_pipeline_stop (CallsSipMediaPipeline *self)
{
g_return_if_fail (CALLS_IS_SIP_MEDIA_PIPELINE (self));
g_debug ("Stopping media pipeline");
self->is_running = FALSE;
/* Stop the pipelines in reverse order (compared to the starting) */
gst_element_set_state (self->send_pipeline, GST_STATE_NULL);
gst_element_set_state (self->recv_pipeline, GST_STATE_NULL);
}
void
calls_sip_media_pipeline_pause (CallsSipMediaPipeline *self)
{
g_return_if_fail (CALLS_IS_SIP_MEDIA_PIPELINE (self));
g_debug ("Pause/unpause media pipeline");
self->is_running = FALSE;
}
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