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Purism-Calls/plugins/sip/calls-sip-media-pipeline.c
Evangelos Ribeiro Tzaras 0e3a07aabf sip: media-pipeline: Setup socket reuse for RTP and RTCP during initialization 
Now that initialization is split per pipeline and that the OS handles port
allocation we can move setting up socket reuse into the pipeline initialization
step instead of setting it up when starting the media pipelines.

This makes the calls_sip_media_pipeline_start() method a bit simpler.

We're also now reusing sockets for RTCP.

Closes #315
2022-03-05 23:02:15 +01:00

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/*
* Copyright (C) 2021-2022 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-media-pipeline-enums.h"
#include "calls-sip-media-pipeline.h"
#include "util.h"
#include <gst/gst.h>
#include <gio/gio.h>
#define MAKE_ELEMENT(var, element, name) \
self->var = gst_element_factory_make (element, name); \
if (!self->var) { \
if (error) \
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED, \
"Could not create '%s' element of type %s", \
name ? : "unnamed", element); \
return FALSE; \
}
/**
* 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.
*/
/* The following defines are used to set/reset bitmaps of playing/paused/stop state */
#define EL_SEND_PIPELINE (1<<0)
#define EL_SEND_AUDIO_SRC (1<<1)
#define EL_SEND_RTPBIN (1<<2)
#define EL_SEND_RTP_SINK (1<<3)
#define EL_SEND_RTCP_SINK (1<<4)
#define EL_SEND_RTCP_SRC (1<<5)
#define EL_SEND_PAYLOADER (1<<6)
#define EL_SEND_ENCODER (1<<7)
#define EL_SEND_ALL_RTP EL_SEND_PIPELINE | EL_SEND_AUDIO_SRC | \
EL_SEND_RTPBIN | EL_SEND_RTP_SINK | EL_SEND_RTCP_SRC | EL_SEND_RTCP_SINK | \
EL_SEND_PAYLOADER | EL_SEND_ENCODER
#define EL_SEND_SENDING EL_SEND_AUDIO_SRC | EL_SEND_RTPBIN | EL_SEND_RTP_SINK | \
EL_SEND_PAYLOADER | EL_SEND_ENCODER
/* leave some room for more elements to be added later */
#define EL_RECV_PIPELINE (1<<16)
#define EL_RECV_AUDIO_SINK (1<<17)
#define EL_RECV_RTPBIN (1<<18)
#define EL_RECV_RTP_SRC (1<<19)
#define EL_RECV_RTCP_SINK (1<<20)
#define EL_RECV_RTCP_SRC (1<<21)
#define EL_RECV_DEPAYLOADER (1<<22)
#define EL_RECV_DECODER (1<<23)
#define EL_RECV_ALL_RTP EL_RECV_PIPELINE | EL_RECV_AUDIO_SINK | \
EL_RECV_RTPBIN | EL_RECV_RTP_SRC | EL_RECV_RTCP_SRC | EL_RECV_RTCP_SINK | \
EL_RECV_DEPAYLOADER | EL_RECV_DECODER
enum {
PROP_0,
PROP_CODEC,
PROP_REMOTE,
PROP_RPORT_RTP,
PROP_RPORT_RTCP,
PROP_DEBUG,
PROP_STATE,
PROP_LAST_PROP,
};
enum {
SENDING_STARTED,
N_SIGNALS
};
static GParamSpec *props[PROP_LAST_PROP];
static uint signals[N_SIGNALS];
struct _CallsSipMediaPipeline {
GObject parent;
MediaCodecInfo *codec;
gboolean debug;
CallsMediaPipelineState state;
uint element_map_playing;
uint element_map_paused;
uint element_map_stopped;
gboolean emitted_sending_signal;
/* Connection details */
char *remote;
gint rport_rtp;
gint rport_rtcp;
/* 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;
};
#if GLIB_CHECK_VERSION(2, 70, 0)
G_DEFINE_FINAL_TYPE (CallsSipMediaPipeline, calls_sip_media_pipeline, G_TYPE_OBJECT)
#else
G_DEFINE_TYPE (CallsSipMediaPipeline, calls_sip_media_pipeline, G_TYPE_OBJECT)
#endif
static void
set_state (CallsSipMediaPipeline *self,
CallsMediaPipelineState state)
{
g_assert (CALLS_SIP_MEDIA_PIPELINE (self));
if (self->state == state)
return;
self->state = state;
g_object_notify_by_pspec (G_OBJECT (self), props[PROP_STATE]);
self->emitted_sending_signal = FALSE;
}
static void
check_element_maps (CallsSipMediaPipeline *self)
{
g_assert (CALLS_IS_SIP_MEDIA_PIPELINE (self));
if (self->element_map_playing == (EL_SEND_ALL_RTP | EL_RECV_ALL_RTP)) {
g_debug ("All pipeline elements are playing");
set_state (self, CALLS_MEDIA_PIPELINE_STATE_PLAYING);
return;
}
if (self->element_map_paused == (EL_SEND_ALL_RTP | EL_RECV_ALL_RTP)) {
g_debug ("All pipeline elements are paused");
set_state (self, CALLS_MEDIA_PIPELINE_STATE_PAUSED);
return;
}
if (self->element_map_stopped == (EL_SEND_ALL_RTP | EL_RECV_ALL_RTP)) {
g_debug ("All pipeline elements are stopped");
set_state (self, CALLS_MEDIA_PIPELINE_STATE_STOPPED);
return;
}
if ((self->element_map_playing & (EL_SEND_SENDING)) == (EL_SEND_SENDING) &&
!self->emitted_sending_signal) {
g_debug ("Sender pipeline is sending data to %s RTP/RTCP %d/%d",
self->remote, self->rport_rtp, self->rport_rtcp);
g_signal_emit (self, signals[SENDING_STARTED], 0);
self->emitted_sending_signal = TRUE;
}
}
/* 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");
g_debug ("linking to %s", GST_PAD_NAME (sinkpad));
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 *self = 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 (self);
break;
case GST_MESSAGE_STATE_CHANGED:
{
GstState oldstate;
GstState newstate;
uint element_id = 0;
uint unset_element_id;
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));
/* Sender pipeline elements */
if (message->src == GST_OBJECT (self->send_pipeline))
element_id = EL_SEND_PIPELINE;
else if (message->src == GST_OBJECT (self->audiosrc))
element_id = EL_SEND_AUDIO_SRC;
else if (message->src == GST_OBJECT (self->send_rtpbin))
element_id = EL_SEND_RTPBIN;
else if (message->src == GST_OBJECT (self->rtp_sink))
element_id = EL_SEND_RTP_SINK;
else if (message->src == GST_OBJECT (self->rtcp_send_sink))
element_id = EL_SEND_RTCP_SINK;
else if (message->src == GST_OBJECT (self->rtcp_send_src))
element_id = EL_SEND_RTCP_SRC;
else if (message->src == GST_OBJECT (self->payloader))
element_id = EL_SEND_PAYLOADER;
else if (message->src == GST_OBJECT (self->encoder))
element_id = EL_SEND_ENCODER;
/* Receiver pipeline elements */
else if (message->src == GST_OBJECT (self->recv_pipeline))
element_id = EL_RECV_PIPELINE;
else if (message->src == GST_OBJECT (self->audiosink))
element_id = EL_RECV_AUDIO_SINK;
else if (message->src == GST_OBJECT (self->recv_rtpbin))
element_id = EL_RECV_RTPBIN;
else if (message->src == GST_OBJECT (self->rtp_src))
element_id = EL_RECV_RTP_SRC;
else if (message->src == GST_OBJECT (self->rtcp_recv_sink))
element_id = EL_RECV_RTCP_SINK;
else if (message->src == GST_OBJECT (self->rtcp_recv_src))
element_id = EL_RECV_RTCP_SRC;
else if (message->src == GST_OBJECT (self->depayloader))
element_id = EL_RECV_DEPAYLOADER;
else if (message->src == GST_OBJECT (self->decoder))
element_id = EL_RECV_DECODER;
unset_element_id = G_MAXUINT ^ element_id;
if (newstate == GST_STATE_PLAYING) {
self->element_map_playing |= element_id;
self->element_map_paused &= unset_element_id;
self->element_map_stopped &= unset_element_id;
} else if (newstate == GST_STATE_PAUSED) {
self->element_map_paused |= element_id;
self->element_map_playing &= unset_element_id;
self->element_map_stopped &= unset_element_id;
} else if (newstate == GST_STATE_NULL) {
self->element_map_stopped |= element_id;
self->element_map_playing &= unset_element_id;
self->element_map_paused &= unset_element_id;
}
check_element_maps (self);
break;
}
default:
if (self->debug)
g_debug ("Got unhandled %s message", GST_MESSAGE_TYPE_NAME (message));
break;
}
/* keep watching for messages on the bus */
return TRUE;
}
/* Setting up pipelines */
static gboolean
send_pipeline_link_elements (CallsSipMediaPipeline *self,
GError **error)
{
g_autoptr (GstPad) srcpad = NULL;
g_autoptr (GstPad) sinkpad = NULL;
g_assert (CALLS_IS_SIP_MEDIA_PIPELINE (self));
#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) {
if (error)
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Failed to link payloader to rtpbin");
return FALSE;
}
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) {
if (error)
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Failed to link rtpbin to rtpsink");
return FALSE;
}
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) {
if (error)
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Failed to link rtpbin to rtcpsink");
return FALSE;
}
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) {
if (error)
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Failed to link rtcpsrc to rtpbin");
return FALSE;
}
return TRUE;
}
static gboolean
send_pipeline_setup_codecs (CallsSipMediaPipeline *self,
MediaCodecInfo *codec,
GError **error)
{
g_assert (CALLS_IS_SIP_MEDIA_PIPELINE (self));
g_assert (codec);
MAKE_ELEMENT (encoder, codec->gst_encoder_name, "encoder");
MAKE_ELEMENT (payloader, codec->gst_payloader_name, "payloader");
gst_bin_add_many (GST_BIN (self->send_pipeline), self->payloader, self->encoder,
self->audiosrc, NULL);
if (!gst_element_link_many (self->audiosrc, self->encoder, self->payloader, NULL)) {
if (error)
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Failed to link audiosrc encoder and payloader");
return FALSE;
}
return send_pipeline_link_elements (self, error);
}
/**
* Prepare a skeleton send pipeline where we can later
* plug the codec specific elements into.
*
* In contrast to the receiver pipeline there is no need to start the
* pipeline until we actually want to establish a media session.
*
* The receiver pipeline should have been initialized at this point
* allowing us to reuse GSockets.
*/
static gboolean
send_pipeline_init (CallsSipMediaPipeline *self,
GError **error)
{
g_autoptr (GSocket) rtp_sock = NULL;
g_autoptr (GSocket) rtcp_sock = NULL;
const char *env_var;
g_assert (CALLS_SIP_MEDIA_PIPELINE (self));
self->send_pipeline = gst_pipeline_new ("rtp-send-pipeline");
if (!self->send_pipeline) {
if (error)
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Could not create send pipeline");
return FALSE;
}
gst_object_ref_sink (self->send_pipeline);
env_var = g_getenv ("CALLS_AUDIOSRC");
if (!STR_IS_NULL_OR_EMPTY (env_var)) {
MAKE_ELEMENT (audiosrc, env_var, "audiosource");
} else {
g_autoptr (GstStructure) gst_props = NULL;
MAKE_ELEMENT (audiosrc, "pulsesrc", "audiosource");
/* 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);
}
MAKE_ELEMENT (send_rtpbin, "rtpbin", "send-rtpbin");
MAKE_ELEMENT (rtp_sink, "udpsink", "rtp-udp-sink");
MAKE_ELEMENT (rtcp_send_src, "udpsrc", "rtcp-udp-send-src");
MAKE_ELEMENT (rtcp_send_sink, "udpsink", "rtcp-udp-send-sink");
g_object_set (self->rtcp_send_sink,
"async", FALSE,
"sync", FALSE,
NULL);
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, "rport-rtcp",
self->rtcp_send_sink, "port",
G_BINDING_BIDIRECTIONAL);
g_object_bind_property (self, "remote",
self->rtcp_send_sink, "host",
G_BINDING_BIDIRECTIONAL);
/* bind sockets for hole punching scheme */
g_object_get (self->rtp_src, "used-socket", &rtp_sock, NULL);
if (!rtp_sock) {
if (error)
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Could not retrieve used socket from RTP udpsrc element");
return FALSE;
}
g_object_set (self->rtp_sink,
"socket", rtp_sock,
"close-socket", FALSE,
NULL);
g_object_get (self->rtcp_recv_src, "used-socket", &rtcp_sock, NULL);
if (!rtcp_sock) {
if (error)
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Could not retrieve used socket from RTCP udpsrc element");
return FALSE;
}
g_object_set (self->rtcp_send_sink,
"socket", rtcp_sock,
"close-socket", FALSE,
NULL);
g_object_set (self->rtcp_send_src,
"socket", rtcp_sock,
"close-socket", FALSE,
NULL);
gst_bin_add (GST_BIN (self->send_pipeline), self->send_rtpbin);
gst_bin_add_many (GST_BIN (self->send_pipeline), self->rtp_sink,
self->rtcp_send_src, self->rtcp_send_sink, NULL);
self->bus_send = gst_pipeline_get_bus (GST_PIPELINE (self->send_pipeline));
self->bus_watch_send = gst_bus_add_watch (self->bus_send, on_bus_message, self);
return TRUE;
}
static gboolean
recv_pipeline_link_elements (CallsSipMediaPipeline *self,
GError **error)
{
g_autoptr (GstPad) srcpad = NULL;
g_autoptr (GstPad) sinkpad = NULL;
g_assert (CALLS_IS_SIP_MEDIA_PIPELINE (self));
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) {
if (error)
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Failed to link rtpsrc to rtpbin");
return FALSE;
}
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) {
if (error)
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Failed to link rtcpsrc to rtpbin");
return FALSE;
}
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) {
if (error)
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Failed to link rtpbin to rtcpsink");
return FALSE;
}
g_signal_connect (self->recv_rtpbin, "pad-added", G_CALLBACK (on_pad_added), self->depayloader);
return TRUE;
}
static gboolean
recv_pipeline_setup_codecs (CallsSipMediaPipeline *self,
MediaCodecInfo *codec,
GError **error)
{
g_autoptr (GstCaps) caps = NULL;
g_autofree char *caps_string = NULL;
g_assert (CALLS_IS_SIP_MEDIA_PIPELINE (self));
g_assert (codec);
MAKE_ELEMENT (decoder, codec->gst_decoder_name, "decoder");
MAKE_ELEMENT (depayloader, codec->gst_depayloader_name, "depayloader");
gst_bin_add_many (GST_BIN (self->recv_pipeline), self->depayloader, self->decoder,
self->audiosink, NULL);
if (!gst_element_link_many (self->depayloader, self->decoder, self->audiosink, NULL)) {
if (error)
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Failed to link depayloader decoder and audiosink");
return FALSE;
}
/* UDP src capabilities */
caps_string = media_codec_get_gst_capabilities (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);
return recv_pipeline_link_elements (self, error);
}
/**
* Prepares a skeleton receiver pipeline which can later be
* used to plug codec specific element in.
* This pipeline just consists of (minimally linked) rtpbin
* audio sink and two udpsrc elements, one for RTP and one for RTCP.
*
* The pipeline will be set ready to let the OS allocate sockets
* for us instead of building and providing GSockets ourselves
* by hand. These GSockets are reused for any outgoing traffic in our
* hole punching scheme as a simple NAT traversal technique.
*/
static gboolean
recv_pipeline_init (CallsSipMediaPipeline *self,
GError **error)
{
g_autoptr (GSocket) rtcp_sock = NULL;
const char *env_var;
g_assert (CALLS_SIP_MEDIA_PIPELINE (self));
self->recv_pipeline = gst_pipeline_new ("rtp-recv-pipeline");
if (!self->recv_pipeline) {
if (error)
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Could not create receiver pipeline");
return FALSE;
}
gst_object_ref_sink (self->recv_pipeline);
env_var = g_getenv ("CALLS_AUDIOSINK");
if (!STR_IS_NULL_OR_EMPTY (env_var)) {
MAKE_ELEMENT (audiosink, env_var, "audiosink");
} else {
g_autoptr (GstStructure) gst_props = NULL;
MAKE_ELEMENT (audiosink, "pulsesink", "audiosink");
/* 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);
}
MAKE_ELEMENT (recv_rtpbin, "rtpbin", "recv-rtpbin")
MAKE_ELEMENT (rtp_src, "udpsrc", "rtp-udp-src");
MAKE_ELEMENT (rtcp_recv_src, "udpsrc", "rtcp-udp-recv-src");
MAKE_ELEMENT (rtcp_recv_sink, "udpsink", "rtcp-udp-recv-sink");
g_object_set (self->rtcp_recv_sink,
"async", FALSE,
"sync", FALSE,
NULL);
/* port 0 means allocate */
g_object_set (self->rtp_src, "port", 0, NULL);
g_object_set (self->rtcp_recv_src, "port", 0, NULL);
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);
gst_bin_add (GST_BIN (self->recv_pipeline), self->recv_rtpbin);
gst_bin_add_many (GST_BIN (self->recv_pipeline), self->rtp_src,
self->rtcp_recv_src, self->rtcp_recv_sink, NULL);
self->bus_recv = gst_pipeline_get_bus (GST_PIPELINE (self->recv_pipeline));
self->bus_watch_recv = gst_bus_add_watch (self->bus_recv, on_bus_message, self);
g_object_set (self->rtp_src,
"close-socket", FALSE,
"reuse", TRUE,
NULL);
g_object_set (self->rtcp_recv_src,
"close-socket", FALSE,
"reuse", TRUE,
NULL);
/* Set pipeline to ready to get ports allocated */
gst_element_set_state (self->recv_pipeline, GST_STATE_READY);
g_object_get (self->rtcp_recv_src, "used-socket", &rtcp_sock, NULL);
if (!rtcp_sock) {
if (error)
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Could not retrieve used socket from RTCP udpsrc element");
return FALSE;
}
/* Ports are allocated. Let's reuse the socket for rtcp source in the sink for NAT traversal*/
g_object_set (self->rtcp_recv_sink,
"socket", rtcp_sock,
"close-socket", FALSE,
NULL);
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_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;
case PROP_STATE:
g_value_set_enum (value, calls_sip_media_pipeline_get_state (self));
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:
calls_sip_media_pipeline_set_codec (self, g_value_get_pointer (value));
break;
case PROP_REMOTE:
g_free (self->remote);
self->remote = g_value_dup_string (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_constructed (GObject *object)
{
g_autoptr (GError) error = NULL;
CallsSipMediaPipeline *self = CALLS_SIP_MEDIA_PIPELINE (object);
G_OBJECT_CLASS (calls_sip_media_pipeline_parent_class)->constructed (object);
set_state (self, CALLS_MEDIA_PIPELINE_STATE_INITIALIZING);
if (!recv_pipeline_init (self, &error)) {
g_warning ("Could not create receive pipeline: %s", error->message);
set_state (self, CALLS_MEDIA_PIPELINE_STATE_ERROR);
return;
}
if (!send_pipeline_init (self, &error)) {
g_warning ("Could not create send pipeline: %s", error->message);
set_state (self, CALLS_MEDIA_PIPELINE_STATE_ERROR);
return;
}
set_state (self, CALLS_MEDIA_PIPELINE_STATE_NEED_CODEC);
}
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->constructed = calls_sip_media_pipeline_constructed;
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_READWRITE);
props[PROP_REMOTE] = g_param_spec_string ("remote",
"Remote",
"Remote host",
NULL,
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);
props[PROP_STATE] = g_param_spec_enum ("state",
"State",
"The state of the media pipeline",
CALLS_TYPE_MEDIA_PIPELINE_STATE,
CALLS_MEDIA_PIPELINE_STATE_UNKNOWN,
G_PARAM_READABLE);
g_object_class_install_properties (object_class, PROP_LAST_PROP, props);
signals[SENDING_STARTED] =
g_signal_new ("sending-started",
G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST,
0, NULL, NULL, NULL,
G_TYPE_NONE, 0);
}
static void
calls_sip_media_pipeline_init (CallsSipMediaPipeline *self)
{
if (!gst_is_initialized ())
gst_init (NULL, NULL);
}
CallsSipMediaPipeline*
calls_sip_media_pipeline_new (MediaCodecInfo *codec)
{
CallsSipMediaPipeline *pipeline;
pipeline = g_object_new (CALLS_TYPE_SIP_MEDIA_PIPELINE, NULL);
if (codec)
g_object_set (pipeline, "codec", codec, NULL);
return pipeline;
}
void
calls_sip_media_pipeline_set_codec (CallsSipMediaPipeline *self,
MediaCodecInfo *codec)
{
g_autoptr (GError) error = NULL;
g_return_if_fail (CALLS_IS_SIP_MEDIA_PIPELINE (self));
g_return_if_fail (codec);
if (self->codec == codec)
return;
if (self->codec) {
g_warning ("Cannot change codec of a pipeline. Use a new pipeline instead.");
return;
}
if (!media_codec_available_in_gst (codec)) {
g_warning ("Cannot setup pipeline with codec '%s' because it's not available in GStreamer",
codec->name);
return;
}
if (!recv_pipeline_setup_codecs (self, codec, &error)) {
g_warning ("Error trying to setup codec for receive pipeline: %s",
error->message);
set_state (self, CALLS_MEDIA_PIPELINE_STATE_ERROR);
return;
}
if (!send_pipeline_setup_codecs (self, codec, &error)) {
g_warning ("Error trying to setup codec for send pipeline: %s",
error->message);
set_state (self, CALLS_MEDIA_PIPELINE_STATE_ERROR);
return;
}
self->codec = codec;
g_object_notify_by_pspec (G_OBJECT (self), props[PROP_CODEC]);
set_state (self, CALLS_MEDIA_PIPELINE_STATE_READY);
}
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));
if (self->state != CALLS_MEDIA_PIPELINE_STATE_PLAYING &&
self->state != CALLS_MEDIA_PIPELINE_STATE_PAUSED) {
g_warning ("Cannot diagnose ports when pipeline is not active");
return;
}
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)
{
g_return_if_fail (CALLS_IS_SIP_MEDIA_PIPELINE (self));
if (self->state != CALLS_MEDIA_PIPELINE_STATE_READY) {
g_warning ("Cannot start pipeline because it's not ready");
return;
}
g_debug ("Starting media pipeline");
gst_element_set_state (self->recv_pipeline, GST_STATE_PLAYING);
gst_element_set_state (self->send_pipeline, GST_STATE_PLAYING);
g_debug ("RTP/RTCP port after starting pipeline: %d/%d",
calls_sip_media_pipeline_get_rtp_port (self),
calls_sip_media_pipeline_get_rtcp_port (self));
set_state (self, CALLS_MEDIA_PIPELINE_STATE_PLAY_PENDING);
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");
/* 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);
set_state (self, CALLS_MEDIA_PIPELINE_STATE_STOP_PENDING);
}
void
calls_sip_media_pipeline_pause (CallsSipMediaPipeline *self,
gboolean pause)
{
g_return_if_fail (CALLS_IS_SIP_MEDIA_PIPELINE (self));
if (pause &&
(self->state == CALLS_MEDIA_PIPELINE_STATE_PAUSED ||
self->state == CALLS_MEDIA_PIPELINE_STATE_PAUSE_PENDING))
return;
if (!pause &&
(self->state == CALLS_MEDIA_PIPELINE_STATE_PLAYING ||
self->state == CALLS_MEDIA_PIPELINE_STATE_PLAY_PENDING))
return;
if (self->state != CALLS_MEDIA_PIPELINE_STATE_PLAYING &&
self->state != CALLS_MEDIA_PIPELINE_STATE_PLAY_PENDING &&
self->state != CALLS_MEDIA_PIPELINE_STATE_PAUSED &&
self->state != CALLS_MEDIA_PIPELINE_STATE_PAUSE_PENDING) {
g_warning ("Cannot pause or unpause pipeline because it's not currently active");
return;
}
g_debug ("%s media pipeline", pause ?
"Pausing" :
"Unpausing");
gst_element_set_state (self->recv_pipeline, pause ?
GST_STATE_PAUSED :
GST_STATE_PLAYING);
gst_element_set_state (self->send_pipeline, pause ?
GST_STATE_PAUSED :
GST_STATE_PLAYING);
set_state (self, pause ?
CALLS_MEDIA_PIPELINE_STATE_PAUSE_PENDING :
CALLS_MEDIA_PIPELINE_STATE_PLAY_PENDING);
}
int
calls_sip_media_pipeline_get_rtp_port (CallsSipMediaPipeline *self)
{
int port;
g_return_val_if_fail (CALLS_IS_SIP_MEDIA_PIPELINE (self), 0);
g_object_get (self->rtp_src, "port", &port, NULL);
return port;
}
int
calls_sip_media_pipeline_get_rtcp_port (CallsSipMediaPipeline *self)
{
int port;
g_return_val_if_fail (CALLS_IS_SIP_MEDIA_PIPELINE (self), 0);
g_object_get (self->rtcp_recv_src, "port", &port, NULL);
return port;
}
CallsMediaPipelineState
calls_sip_media_pipeline_get_state (CallsSipMediaPipeline *self)
{
g_return_val_if_fail (CALLS_IS_SIP_MEDIA_PIPELINE (self),
CALLS_MEDIA_PIPELINE_STATE_UNKNOWN);
return self->state;
}
#undef MAKE_ELEMENT
#undef EL_SEND_PIPELINE
#undef EL_SEND_AUDIO_SRC
#undef EL_SEND_RTPBIN
#undef EL_SEND_RTP_SINK
#undef EL_SEND_RTCP_SINK
#undef EL_SEND_RTCP_SRC
#undef EL_SEND_PAYLOADER
#undef EL_SEND_ENCODER
#undef EL_SEND_ALL_RTP
#undef EL_SEND_SENDING
#undef EL_RECV_PIPELINE
#undef EL_RECV_AUDIO_SINK
#undef EL_RECV_RTPBIN
#undef EL_RECV_RTP_SRC
#undef EL_RECV_RTCP_SINK
#undef EL_RECV_RTCP_SRC
#undef EL_RECV_DEPAYLOADER
#undef EL_RECV_DECODER
#undef EL_RECV_ALL_RTP
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