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Purism-Calls/plugins/sip/calls-sip-media-pipeline.c
Evangelos Ribeiro Tzaras a14b6bfbf5 sip: media-pipeline: Uncrustify struct members
2022-05-24 22:30:03 +02:00

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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 "calls-srtp-utils.h"
#include "util.h"
#include <glib-unix.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_PIPELINE (1<<0)
#define EL_RTPBIN (1<<1)
#define EL_RTP_SRC (1<<2)
#define EL_RTP_SINK (1<<3)
#define EL_RTCP_SRC (1<<4)
#define EL_RTCP_SINK (1<<5)
#define EL_SRTP_ENCODER (1<<6)
#define EL_SRTP_DECODER (1<<7)
#define EL_AUDIO_SRC (1<<8)
#define EL_AUDIO_SINK (1<<9)
#define EL_PAYLOADER (1<<10)
#define EL_DEPAYLOADER (1<<11)
#define EL_ENCODER (1<<12)
#define EL_DECODER (1<<13)
#define EL_SENDING \
(EL_AUDIO_SRC | EL_ENCODER | EL_PAYLOADER | \
EL_RTPBIN | EL_RTP_SINK | EL_RTCP_SINK)
#define EL_ALL_RTP \
(EL_PIPELINE | EL_RTPBIN | \
EL_RTP_SRC | EL_RTP_SINK | EL_RTCP_SRC | EL_RTCP_SINK | \
EL_AUDIO_SRC | EL_AUDIO_SINK | \
EL_ENCODER | EL_DECODER | EL_PAYLOADER | EL_DEPAYLOADER)
#define EL_ALL_SRTP (EL_ALL_RTP | EL_SRTP_ENCODER | EL_SRTP_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;
GstElement *pipeline;
GstElement *rtpbin;
GstElement *rtp_src;
GstElement *rtp_sink;
GstElement *rtcp_sink;
GstElement *rtcp_src;
GstElement *audio_src;
GstElement *payloader;
GstElement *encoder;
GstElement *audio_sink;
GstElement *depayloader;
GstElement *decoder;
/* SRTP */
gboolean use_srtp;
calls_srtp_crypto_attribute *crypto_own;
calls_srtp_crypto_attribute *crypto_theirs;
GstElement *srtpenc;
GstElement *srtpdec;
gulong request_rtpbin_rtp_decoder_id;
gulong request_rtpbin_rtp_encoder_id;
gulong request_rtpbin_rtcp_encoder_id;
gulong request_rtpbin_rtcp_decoder_id;
/* Gstreamer busses */
GstBus *bus;
guint bus_watch_id;
};
#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_autoptr (GEnumClass) enum_class = NULL;
GEnumValue *enum_val;
g_autofree char *fname = NULL;
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;
if (state == CALLS_MEDIA_PIPELINE_STATE_INITIALIZING)
return;
enum_class = g_type_class_ref (CALLS_TYPE_MEDIA_PIPELINE_STATE);
enum_val = g_enum_get_value (enum_class, state);
fname = g_strdup_printf ("calls-%s", enum_val->value_nick);
GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (self->pipeline),
GST_DEBUG_GRAPH_SHOW_ALL,
fname);
}
static void
check_element_maps (CallsSipMediaPipeline *self)
{
uint all_rtp_elements;
g_assert (CALLS_IS_SIP_MEDIA_PIPELINE (self));
all_rtp_elements = self->use_srtp ? EL_ALL_SRTP : EL_ALL_RTP;
if (self->element_map_playing == all_rtp_elements) {
g_debug ("All pipeline elements are playing");
set_state (self, CALLS_MEDIA_PIPELINE_STATE_PLAYING);
return;
}
if (self->element_map_paused == all_rtp_elements) {
g_debug ("All pipeline elements are paused");
set_state (self, CALLS_MEDIA_PIPELINE_STATE_PAUSED);
return;
}
if (self->element_map_stopped == all_rtp_elements) {
g_debug ("All pipeline elements are stopped");
set_state (self, CALLS_MEDIA_PIPELINE_STATE_STOPPED);
return;
}
if ((self->element_map_playing & (EL_SENDING)) == (EL_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));
if (message->src == GST_OBJECT (self->pipeline))
element_id = EL_PIPELINE;
else if (message->src == GST_OBJECT (self->rtpbin))
element_id = EL_RTPBIN;
else if (message->src == GST_OBJECT (self->rtp_src))
element_id = EL_RTP_SRC;
else if (message->src == GST_OBJECT (self->rtp_sink))
element_id = EL_RTP_SINK;
else if (message->src == GST_OBJECT (self->rtcp_src))
element_id = EL_RTCP_SRC;
else if (message->src == GST_OBJECT (self->rtcp_sink))
element_id = EL_RTCP_SINK;
else if (message->src == GST_OBJECT (self->srtpenc))
element_id = EL_SRTP_ENCODER;
else if (message->src == GST_OBJECT (self->srtpdec))
element_id = EL_SRTP_DECODER;
else if (message->src == GST_OBJECT (self->audio_src))
element_id = EL_AUDIO_SRC;
else if (message->src == GST_OBJECT (self->audio_sink))
element_id = EL_AUDIO_SINK;
else if (message->src == GST_OBJECT (self->payloader))
element_id = EL_PAYLOADER;
else if (message->src == GST_OBJECT (self->depayloader))
element_id = EL_DEPAYLOADER;
else if (message->src == GST_OBJECT (self->encoder))
element_id = EL_ENCODER;
else if (message->src == GST_OBJECT (self->decoder))
element_id = EL_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;
}
/* SRTP setup */
static GstCaps *
on_srtpdec_request_key (GstElement *srtpdec,
guint ssrc,
gpointer user_data)
{
CallsSipMediaPipeline *self = CALLS_SIP_MEDIA_PIPELINE (user_data);
GstCaps *caps;
const char *srtp_cipher = "null";
const char *srtcp_cipher = "null";
const char *srtp_auth = "null";
const char *srtcp_auth = "null";
gboolean need_mki;
if (!calls_srtp_crypto_get_srtpdec_params (self->crypto_theirs,
&srtp_cipher,
&srtp_auth,
&srtcp_cipher,
&srtcp_auth))
return NULL;
if (self->crypto_theirs->n_key_params == 0 ||
self->crypto_theirs->n_key_params > 16) {
g_warning ("Got %u key parameters, but can only handle between 1 and 16",
self->crypto_theirs->n_key_params);
return NULL;
}
need_mki = self->crypto_theirs->n_key_params > 1;
if (self->crypto_theirs->n_key_params == 1) {
/* g_autofree guchar *key_salt = NULL; */
guchar *key_salt = NULL;
gsize key_salt_length;
g_autoptr (GstBuffer) key_buffer = NULL;
key_salt = g_base64_decode (self->crypto_theirs->key_params[0].b64_keysalt,
&key_salt_length);
key_buffer = gst_buffer_new_wrapped (key_salt, key_salt_length);
/* TODO Setting up MKI buffer not implemented yet */
if (self->crypto_theirs->key_params[0].mki) {
g_warning ("Using MKI is not implemented yet");
return NULL;
}
return gst_caps_new_simple ("application/x-srtp",
"srtp-key", GST_TYPE_BUFFER, key_buffer,
"srtp-cipher", G_TYPE_STRING, srtp_cipher,
"srtcp-cipher", G_TYPE_STRING, srtcp_cipher,
"srtp-auth", G_TYPE_STRING, srtp_auth,
"srtcp-auth", G_TYPE_STRING, srtcp_auth,
NULL);
}
/* TODO Setting up MKI buffer not implemented yet */
g_warning ("Using MKI is not implemented yet");
return NULL;
caps = gst_caps_new_simple ("application/x-srtp",
"srtp-cipher", G_TYPE_STRING, srtp_cipher,
"srtcp-cipher", G_TYPE_STRING, srtcp_cipher,
"srtp-auth", G_TYPE_STRING, srtp_auth,
"srtcp-auth", G_TYPE_STRING, srtcp_auth,
NULL);
for (guint i = 0; i < self->crypto_theirs->n_key_params; i++) {
GstStructure *structure;
g_autofree char *structure_name = g_strdup_printf ("key-%u", i);
guchar *key_salt = NULL;
gsize key_salt_length;
g_autoptr (GstBuffer) key_buffer = NULL;
g_autoptr (GstBuffer) mki_buffer = NULL;
key_salt = g_base64_decode (self->crypto_theirs->key_params[0].b64_keysalt,
&key_salt_length);
key_buffer = gst_buffer_new_wrapped (key_salt, key_salt_length);
if (i == 0 && need_mki) {
structure = gst_structure_new (structure_name,
"srtp-key", GST_TYPE_BUFFER, key_buffer,
"mki", GST_TYPE_BUFFER, mki_buffer,
NULL);
} else if (i == 0 && !need_mki) {
structure = gst_structure_new (structure_name,
"srtp-key", GST_TYPE_BUFFER, key_buffer,
NULL);
} else {
g_autofree char *key_field_name = g_strdup_printf ("srtp-key%u", i+1);
g_autofree char *mki_field_name = g_strdup_printf ("mki%u", i+1);
structure = gst_structure_new (structure_name,
key_field_name, GST_TYPE_BUFFER, key_buffer,
mki_field_name, GST_TYPE_BUFFER, mki_buffer,
NULL);
}
gst_caps_append_structure (caps, structure);
}
return caps;
}
static GstElement *
on_rtpbin_request_decoder (GstElement *rtpbin,
guint session_id,
gpointer user_data)
{
CallsSipMediaPipeline *self = CALLS_SIP_MEDIA_PIPELINE (user_data);
if (!self->use_srtp)
return NULL;
return gst_object_ref (self->srtpdec);
}
static GstElement *
on_rtpbin_request_encoder (GstElement *rtpbin,
guint session_id,
gpointer user_data)
{
CallsSipMediaPipeline *self = CALLS_SIP_MEDIA_PIPELINE (user_data);
if (!self->use_srtp)
return NULL;
return gst_object_ref (self->srtpenc);
}
/* Pipeline setup */
static gboolean
setup_socket_reuse (CallsSipMediaPipeline *self,
GError **error)
{
g_autoptr (GSocket) rtp_sock = NULL;
g_autoptr (GSocket) rtcp_sock = NULL;
/* set rtp element ready and lock it's state so it doesn't get stopped */
gst_element_set_locked_state (self->rtp_src, TRUE);
gst_element_set_state (self->rtp_src, GST_STATE_READY);
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;
}
/* configure socket and don't close it, since it belongs to rtp_src */
g_object_set (self->rtp_sink,
"socket", rtp_sock,
"close-socket", FALSE,
NULL);
/* set rtcp element ready and lock it's state so it doesn't get stopped */
gst_element_set_locked_state (self->rtcp_src, TRUE);
gst_element_set_state (self->rtcp_src, GST_STATE_READY);
g_object_get (self->rtcp_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;
}
/* configure socket and don't close it, since it belongs to rtcp_src */
g_object_set (self->rtcp_sink,
"socket", rtcp_sock,
"close-socket", FALSE,
NULL);
return TRUE;
}
static gboolean
pipeline_init (CallsSipMediaPipeline *self,
GError **error)
{
GstPad *tmppad;
const char *env_var;
g_assert (CALLS_SIP_MEDIA_PIPELINE (self));
self->pipeline = gst_pipeline_new ("media-pipeline");
if (!self->pipeline) {
if (error)
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Could not create media pipeline");
return FALSE;
}
gst_object_ref_sink (self->pipeline);
/* Audio source*/
env_var = g_getenv ("CALLS_AUDIOSRC");
if (!STR_IS_NULL_OR_EMPTY (env_var)) {
MAKE_ELEMENT (audio_src, env_var, "audiosource");
} else {
g_autoptr (GstStructure) gst_props = NULL;
MAKE_ELEMENT (audio_src, "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->audio_src,
"buffer-time", (gint64) 40000,
"stream-properties", gst_props,
NULL);
}
/* Audio sink */
env_var = g_getenv ("CALLS_AUDIOSINK");
if (!STR_IS_NULL_OR_EMPTY (env_var)) {
MAKE_ELEMENT (audio_sink, env_var, "audiosink");
} else {
g_autoptr (GstStructure) gst_props = NULL;
MAKE_ELEMENT (audio_sink, "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->audio_sink,
"buffer-time", (gint64) 40000,
"stream-properties", gst_props,
NULL);
}
/* rtpbin */
MAKE_ELEMENT (rtpbin, "rtpbin", "rtpbin");
/* srtp elements */
MAKE_ELEMENT (srtpdec, "srtpdec", "srtpdec");
g_signal_connect (self->srtpdec,
"request-key",
G_CALLBACK (on_srtpdec_request_key),
self);
MAKE_ELEMENT (srtpenc, "srtpenc", "srtpenc");
#if GST_CHECK_VERSION (1, 20, 0)
tmppad = gst_element_request_pad_simple (self->srtpenc, "rtp_sink_0");
#else
tmppad = gst_element_get_request_pad (self->srtpenc, "rtp_sink_0");
#endif
gst_object_unref (tmppad);
#if GST_CHECK_VERSION (1, 20, 0)
tmppad = gst_element_request_pad_simple (self->srtpenc, "rtcp_sink_0");
#else
tmppad = gst_element_get_request_pad (self->srtpenc, "rtcp_sink_0");
#endif
gst_object_unref (tmppad);
self->request_rtpbin_rtp_encoder_id =
g_signal_connect (self->rtpbin,
"request-rtp-encoder",
G_CALLBACK (on_rtpbin_request_encoder),
self);
self->request_rtpbin_rtp_decoder_id =
g_signal_connect (self->rtpbin,
"request-rtp-decoder",
G_CALLBACK (on_rtpbin_request_decoder),
self);
self->request_rtpbin_rtcp_encoder_id =
g_signal_connect (self->rtpbin,
"request-rtcp-encoder",
G_CALLBACK (on_rtpbin_request_encoder),
self);
self->request_rtpbin_rtcp_decoder_id =
g_signal_connect (self->rtpbin,
"request-rtcp-decoder",
G_CALLBACK (on_rtpbin_request_decoder),
self);
/* UDP sources and sinks for RTP and RTCP */
MAKE_ELEMENT (rtp_src, "udpsrc", "rtp-udp-src");
MAKE_ELEMENT (rtp_sink, "udpsink", "rtp-udp-sink");
MAKE_ELEMENT (rtcp_src, "udpsrc", "rtcp-udp-src");
MAKE_ELEMENT (rtcp_sink, "udpsink", "rtcp-udp-sink");
/* port 0 means letting the OS allocate */
g_object_set (self->rtp_src, "port", 0, NULL);
g_object_set (self->rtcp_src, "port", 0, NULL);
g_object_set (self->rtp_sink, "async", FALSE, "sync", FALSE, NULL);
g_object_set (self->rtcp_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_sink, "port",
G_BINDING_BIDIRECTIONAL);
g_object_bind_property (self, "remote",
self->rtcp_sink, "host",
G_BINDING_BIDIRECTIONAL);
/* Add all elements to the pipeline */
gst_bin_add_many (GST_BIN (self->pipeline),
self->audio_src, self->audio_sink,
self->rtpbin,
self->rtp_src, self->rtp_sink,
self->rtcp_src, self->rtcp_sink,
NULL);
/* Setup bus watch */
self->bus = gst_pipeline_get_bus (GST_PIPELINE (self->pipeline));
self->bus_watch_id = gst_bus_add_watch (self->bus, on_bus_message, self);
if (!setup_socket_reuse (self, error))
return FALSE;
return TRUE;
}
static gboolean
pipeline_link_elements (CallsSipMediaPipeline *self,
GError **error)
{
g_autoptr (GstPad) srcpad = NULL;
g_autoptr (GstPad) sinkpad = NULL;
GstPadLinkReturn ret;
g_assert (CALLS_IS_SIP_MEDIA_PIPELINE (self));
/* link to payloader */
#if GST_CHECK_VERSION (1, 20, 0)
sinkpad = gst_element_request_pad_simple (self->rtpbin, "send_rtp_sink_0");
#else
sinkpad = gst_element_get_request_pad (self->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;
}
/* Transmitter 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->rtpbin, "recv_rtp_sink_0");
#else
sinkpad = gst_element_get_request_pad (self->rtpbin, "recv_rtp_sink_0");
#endif
ret = gst_pad_link (srcpad, sinkpad);
if (ret != 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->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);
srcpad = gst_element_get_static_pad (self->rtcp_src, "src");
#if GST_CHECK_VERSION (1, 20, 0)
sinkpad = gst_element_request_pad_simple (self->rtpbin, "recv_rtcp_sink_0");
#else
sinkpad = gst_element_get_request_pad (self->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->rtpbin, "send_rtcp_src_0");
#else
srcpad = gst_element_get_request_pad (self->rtpbin, "send_rtcp_src_0");
#endif
sinkpad = gst_element_get_static_pad (self->rtcp_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;
}
/* can only link to depayloader after RTP payload has been verified */
g_signal_connect (self->rtpbin, "pad-added", G_CALLBACK (on_pad_added), self->depayloader);
/* request-encoder and request-decoder signals have been emitted after linking pads from rtpbin */
if (self->request_rtpbin_rtp_decoder_id)
g_signal_handler_disconnect (self->rtpbin, self->request_rtpbin_rtp_decoder_id);
if (self->request_rtpbin_rtp_encoder_id)
g_signal_handler_disconnect (self->rtpbin, self->request_rtpbin_rtp_encoder_id);
if (self->request_rtpbin_rtcp_decoder_id)
g_signal_handler_disconnect (self->rtpbin, self->request_rtpbin_rtcp_decoder_id);
if (self->request_rtpbin_rtcp_encoder_id)
g_signal_handler_disconnect (self->rtpbin, self->request_rtpbin_rtcp_encoder_id);
return TRUE;
}
static gboolean
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");
MAKE_ELEMENT (encoder, codec->gst_encoder_name, "encoder");
MAKE_ELEMENT (payloader, codec->gst_payloader_name, "payloader");
gst_bin_add_many (GST_BIN (self->pipeline),
self->depayloader, self->decoder,
self->payloader, self->encoder,
NULL);
if (!gst_element_link_many (self->audio_src, 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;
}
if (!gst_element_link_many (self->depayloader, self->decoder, self->audio_sink, 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, self->use_srtp);
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 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 (!pipeline_init (self, &error)) {
g_warning ("Could not create 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->pipeline);
gst_bus_remove_watch (self->bus);
gst_object_unref (self->bus);
gst_object_unref (self->srtpenc);
gst_object_unref (self->srtpdec);
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 gboolean
usr2_handler (CallsSipMediaPipeline *self)
{
g_print ("playing: %d\n"
"paused: %d\n"
"stopped: %d\n"
"target map: %d\n"
"current state: %d\n",
self->element_map_playing,
self->element_map_paused,
self->element_map_stopped,
self->use_srtp ? EL_ALL_SRTP : EL_ALL_RTP,
self->state);
GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (self->pipeline),
GST_DEBUG_GRAPH_SHOW_ALL,
"usr2-debug");
return G_SOURCE_CONTINUE;
}
static void
calls_sip_media_pipeline_init (CallsSipMediaPipeline *self)
{
if (!gst_is_initialized ())
gst_init (NULL, NULL);
/* Pipeline debugging */
g_unix_signal_add (SIGUSR2,
(GSourceFunc) usr2_handler,
self);
}
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 (!pipeline_setup_codecs (self, codec, &error)) {
g_warning ("Error trying to setup codecs for pipeline: %s",
error->message);
set_state (self, CALLS_MEDIA_PIPELINE_STATE_ERROR);
return;
}
if (!pipeline_link_elements (self, &error)) {
g_warning ("Not all pads could be linked: %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);
}
void
calls_sip_media_pipeline_set_crypto (CallsSipMediaPipeline *self,
calls_srtp_crypto_attribute *crypto_own,
calls_srtp_crypto_attribute *crypto_theirs)
{
guchar *key_salt = NULL;
gsize key_salt_length;
GstSrtpCipherType srtp_cipher;
GstSrtpAuthType srtp_auth;
GstSrtpCipherType srtcp_cipher;
GstSrtpAuthType srtcp_auth;
g_autoptr (GstBuffer) key_buffer = NULL;
g_return_if_fail (CALLS_IS_SIP_MEDIA_PIPELINE (self));
g_return_if_fail (crypto_own);
g_return_if_fail (crypto_theirs);
g_return_if_fail (crypto_own->crypto_suite == crypto_theirs->crypto_suite);
g_return_if_fail (crypto_own->tag == crypto_theirs->tag);
if (self->use_srtp)
return;
self->use_srtp = TRUE;
self->crypto_own = crypto_own;
self->crypto_theirs = crypto_theirs;
if (!calls_srtp_crypto_get_srtpenc_params (crypto_own,
&srtp_cipher,
&srtp_auth,
&srtcp_cipher,
&srtcp_auth)) {
g_autofree char *attr_str =
calls_srtp_print_sdp_crypto_attribute (crypto_own, NULL);
g_warning ("Could not get srtpenc parameters from attribute: %s", attr_str);
return;
}
/* TODO MKI stuff */
key_salt = g_base64_decode (crypto_own->key_params[0].b64_keysalt,
&key_salt_length);
key_buffer = gst_buffer_new_wrapped (key_salt, key_salt_length);
g_object_set (self->srtpenc,
"key", key_buffer,
"rtp-cipher", srtp_cipher,
"rtp-auth", srtp_auth,
"rtcp-cipher", srtcp_cipher,
"rtcp-auth", srtcp_auth,
NULL);
}
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");
g_debug ("RTP/RTCP port before starting pipeline: %d/%d",
calls_sip_media_pipeline_get_rtp_port (self),
calls_sip_media_pipeline_get_rtcp_port (self));
/* unlock the state of our udp sources, see setup_socket_reuse() */
gst_element_set_locked_state (self->rtp_src, FALSE);
gst_element_set_locked_state (self->rtcp_src, FALSE);
gst_element_set_state (self->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");
gst_element_set_locked_state (self->rtp_src, FALSE);
gst_element_set_locked_state (self->rtcp_src, FALSE);
gst_element_set_locked_state (self->rtp_sink, FALSE);
gst_element_set_locked_state (self->rtcp_sink, FALSE);
gst_element_set_state (self->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");
/* leave udpsrc running to prevent timeouts */
gst_element_set_locked_state (self->rtp_src, pause);
gst_element_set_locked_state (self->rtcp_src, pause);
gst_element_set_locked_state (self->rtp_sink, pause);
gst_element_set_locked_state (self->rtcp_sink, pause);
gst_element_set_state (self->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_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;
}
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