mirror of
https://gitlab.gnome.org/GNOME/calls.git
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sip: media-pipeline: Split initialization per GstPipeline
This is the first step in getting rid of the requirement to have the codec set during object construction. The goal is to have pipelines prepared in advance so that the codec can be plugged in once negotiation is complete. Having the pipelines prepared in advance let's us grab allocated local ports of udpsrc elements for RTP and RTCP instead of setting those and hoping they're not yet in use.
This commit is contained in:
Evangelos Ribeiro Tzaras
parent
86e76380c2
commit
792e90516a
1 changed files with 419 additions and 323 deletions
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@ -1,5 +1,5 @@
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/*
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* Copyright (C) 2021 Purism SPC
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* Copyright (C) 2021-2022 Purism SPC
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*
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* This file is part of Calls.
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*
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@ -25,10 +25,22 @@
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#define G_LOG_DOMAIN "CallsSipMediaPipeline"
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#include "calls-sip-media-pipeline.h"
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#include "util.h"
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#include <gst/gst.h>
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#include <gio/gio.h>
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#define MAKE_ELEMENT(var, element, name) \
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self->var = gst_element_factory_make (element, name); \
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if (!self->var) { \
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if (error) \
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g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED, \
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"Could not create '%s' element of type %s", \
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name ? : "unnamed", element); \
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return FALSE; \
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}
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/**
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* SECTION:sip-media-pipeline
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* @short_description:
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@ -118,6 +130,8 @@ on_pad_added (GstElement *rtpbin,
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g_debug ("pad added: %s", GST_PAD_NAME (srcpad));
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sinkpad = gst_element_get_static_pad (depayloader, "sink");
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g_debug ("linking to %s", GST_PAD_NAME (sinkpad));
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if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK)
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g_warning ("Failed to link rtpbin to depayloader");
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@ -182,6 +196,402 @@ on_bus_message (GstBus *bus,
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}
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/* Setting up pipelines */
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static gboolean
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send_pipeline_link_elements (CallsSipMediaPipeline *self,
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GError **error)
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{
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g_autoptr (GstPad) srcpad = NULL;
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g_autoptr (GstPad) sinkpad = NULL;
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g_assert (CALLS_IS_SIP_MEDIA_PIPELINE (self));
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#if GST_CHECK_VERSION (1, 20, 0)
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sinkpad = gst_element_request_pad_simple (self->send_rtpbin, "send_rtp_sink_0");
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#else
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sinkpad = gst_element_get_request_pad (self->send_rtpbin, "send_rtp_sink_0");
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#endif
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srcpad = gst_element_get_static_pad (self->payloader, "src");
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if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) {
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if (error)
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g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
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"Failed to link payloader to rtpbin");
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return FALSE;
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}
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gst_object_unref (srcpad);
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gst_object_unref (sinkpad);
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/* link RTP srcpad to udpsink */
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srcpad = gst_element_get_static_pad (self->send_rtpbin, "send_rtp_src_0");
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sinkpad = gst_element_get_static_pad (self->rtp_sink, "sink");
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if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) {
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if (error)
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g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
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"Failed to link rtpbin to rtpsink");
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return FALSE;
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}
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gst_object_unref (srcpad);
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gst_object_unref (sinkpad);
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/* RTCP srcpad to udpsink */
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#if GST_CHECK_VERSION (1, 20, 0)
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srcpad = gst_element_request_pad_simple (self->send_rtpbin, "send_rtcp_src_0");
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#else
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srcpad = gst_element_get_request_pad (self->send_rtpbin, "send_rtcp_src_0");
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#endif
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sinkpad = gst_element_get_static_pad (self->rtcp_send_sink, "sink");
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if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) {
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if (error)
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g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
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"Failed to link rtpbin to rtcpsink");
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return FALSE;
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}
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gst_object_unref (srcpad);
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gst_object_unref (sinkpad);
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/* receive RTCP */
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srcpad = gst_element_get_static_pad (self->rtcp_send_src, "src");
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#if GST_CHECK_VERSION (1, 20, 0)
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sinkpad = gst_element_request_pad_simple (self->send_rtpbin, "recv_rtcp_sink_0");
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#else
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sinkpad = gst_element_get_request_pad (self->send_rtpbin, "recv_rtcp_sink_0");
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#endif
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if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) {
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if (error)
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g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
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"Failed to link rtcpsrc to rtpbin");
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return FALSE;
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}
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return TRUE;
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}
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static gboolean
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send_pipeline_setup_codecs (CallsSipMediaPipeline *self,
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MediaCodecInfo *codec,
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GError **error)
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{
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g_assert (CALLS_IS_SIP_MEDIA_PIPELINE (self));
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g_assert (codec);
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/* TODO check if codec is available */
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MAKE_ELEMENT (encoder, codec->gst_encoder_name, "encoder");
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MAKE_ELEMENT (payloader, codec->gst_payloader_name, "payloader");
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gst_bin_add_many (GST_BIN (self->send_pipeline), self->payloader, self->encoder,
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self->audiosrc, NULL);
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if (!gst_element_link_many (self->audiosrc, self->encoder, self->payloader, NULL)) {
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if (error)
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g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
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"Failed to link audiosrc encoder and payloader");
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return FALSE;
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}
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return send_pipeline_link_elements (self, error);
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}
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/** TODO: we're describing the desired state (not the current state)
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* Prepare a skeleton send pipeline where we can later
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* plug the codec specific elements into.
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*
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* In contrast to the receiver pipeline there is no need to start the
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* pipeline until we actually want to establish a media session.
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*
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* The receiver pipeline should have been initialized at this point
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* allowing us to reuse GSockets.
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*/
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static gboolean
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send_pipeline_init (CallsSipMediaPipeline *self,
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GCancellable *cancellable,
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GError **error)
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{
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const char *env_var;
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g_assert (CALLS_SIP_MEDIA_PIPELINE (self));
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self->send_pipeline = gst_pipeline_new ("rtp-send-pipeline");
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if (!self->send_pipeline) {
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if (error)
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g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
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"Could not create send pipeline");
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return FALSE;
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}
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gst_object_ref_sink (self->send_pipeline);
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env_var = g_getenv ("CALLS_AUDIOSRC");
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if (!STR_IS_NULL_OR_EMPTY (env_var)) {
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MAKE_ELEMENT (audiosrc, env_var, "audiosource");
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} else {
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g_autoptr (GstStructure) gst_props = NULL;
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MAKE_ELEMENT (audiosrc, "pulsesrc", "audiosource");
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/* enable echo cancellation and set buffer size to 40ms */
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gst_props = gst_structure_new ("props",
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"media.role", G_TYPE_STRING, "phone",
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"filter.want", G_TYPE_STRING, "echo-cancel",
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NULL);
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g_object_set (self->audiosrc,
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"buffer-time", (gint64) 40000,
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"stream-properties", gst_props,
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NULL);
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}
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MAKE_ELEMENT (send_rtpbin, "rtpbin", "send-rtpbin");
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MAKE_ELEMENT (rtp_sink, "udpsink", "rtp-udp-sink");
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MAKE_ELEMENT (rtcp_send_src, "udpsrc", "rtcp-udp-send-src");
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MAKE_ELEMENT (rtcp_send_sink, "udpsink", "rtcp-udp-send-sink");
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g_object_set (self->rtcp_send_sink,
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"async", FALSE,
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"sync", FALSE,
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NULL);
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g_object_bind_property (self, "rport-rtp",
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self->rtp_sink, "port",
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G_BINDING_BIDIRECTIONAL);
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g_object_bind_property (self, "remote",
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self->rtp_sink, "host",
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G_BINDING_BIDIRECTIONAL);
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g_object_bind_property (self, "lport-rtcp",
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self->rtcp_send_src, "port",
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G_BINDING_BIDIRECTIONAL);
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g_object_bind_property (self, "rport-rtcp",
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self->rtcp_send_sink, "port",
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G_BINDING_BIDIRECTIONAL);
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g_object_bind_property (self, "remote",
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self->rtcp_send_sink, "host",
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G_BINDING_BIDIRECTIONAL);
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gst_bin_add (GST_BIN (self->send_pipeline), self->send_rtpbin);
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gst_bin_add_many (GST_BIN (self->send_pipeline), self->rtp_sink,
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self->rtcp_send_src, self->rtcp_send_sink, NULL);
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/* TODO setting up codecs should be delayed in the future until after
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* codecs have been negotiated
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*/
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if (!send_pipeline_setup_codecs (self, self->codec, error))
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return FALSE;
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self->bus_send = gst_pipeline_get_bus (GST_PIPELINE (self->send_pipeline));
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self->bus_watch_send = gst_bus_add_watch (self->bus_send, on_bus_message, self);
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return TRUE;
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}
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static gboolean
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recv_pipeline_link_elements (CallsSipMediaPipeline *self,
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GError **error)
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{
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g_autoptr (GstPad) srcpad = NULL;
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g_autoptr (GstPad) sinkpad = NULL;
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g_assert (CALLS_IS_SIP_MEDIA_PIPELINE (self));
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srcpad = gst_element_get_static_pad (self->rtp_src, "src");
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#if GST_CHECK_VERSION (1, 20, 0)
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sinkpad = gst_element_request_pad_simple (self->recv_rtpbin, "recv_rtp_sink_0");
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#else
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sinkpad = gst_element_get_request_pad (self->recv_rtpbin, "recv_rtp_sink_0");
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#endif
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if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) {
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if (error)
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g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
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"Failed to link rtpsrc to rtpbin");
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return FALSE;
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}
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gst_object_unref (srcpad);
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gst_object_unref (sinkpad);
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srcpad = gst_element_get_static_pad (self->rtcp_recv_src, "src");
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#if GST_CHECK_VERSION (1, 20 , 0)
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sinkpad = gst_element_request_pad_simple (self->recv_rtpbin, "recv_rtcp_sink_0");
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#else
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sinkpad = gst_element_get_request_pad (self->recv_rtpbin, "recv_rtcp_sink_0");
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#endif
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if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) {
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if (error)
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g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
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"Failed to link rtcpsrc to rtpbin");
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return FALSE;
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}
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gst_object_unref (srcpad);
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gst_object_unref (sinkpad);
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#if GST_CHECK_VERSION (1, 20, 0)
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srcpad = gst_element_request_pad_simple (self->recv_rtpbin, "send_rtcp_src_0");
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#else
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srcpad = gst_element_get_request_pad (self->recv_rtpbin, "send_rtcp_src_0");
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#endif
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sinkpad = gst_element_get_static_pad (self->rtcp_recv_sink, "sink");
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if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) {
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if (error)
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g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
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"Failed to link rtpbin to rtcpsink");
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return FALSE;
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}
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g_signal_connect (self->recv_rtpbin, "pad-added", G_CALLBACK (on_pad_added), self->depayloader);
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return TRUE;
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}
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static gboolean
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recv_pipeline_setup_codecs (CallsSipMediaPipeline *self,
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MediaCodecInfo *codec,
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GError **error)
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{
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g_autoptr (GstCaps) caps = NULL;
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g_autofree char *caps_string = NULL;
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g_assert (CALLS_IS_SIP_MEDIA_PIPELINE (self));
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g_assert (codec);
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/* TODO check if codec is available */
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MAKE_ELEMENT (decoder, codec->gst_decoder_name, "decoder");
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MAKE_ELEMENT (depayloader, codec->gst_depayloader_name, "depayloader");
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gst_bin_add_many (GST_BIN (self->recv_pipeline), self->depayloader, self->decoder,
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self->audiosink, NULL);
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if (!gst_element_link_many (self->depayloader, self->decoder, self->audiosink, NULL)) {
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if (error)
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g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
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"Failed to link depayloader decoder and audiosink");
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return FALSE;
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}
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/* UDP src capabilities */
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caps_string = media_codec_get_gst_capabilities (codec);
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g_debug ("Capabilities:\n%s", caps_string);
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caps = gst_caps_from_string (caps_string);
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/* set udp sinks and sources for RTP and RTCP */
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g_object_set (self->rtp_src,
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"caps", caps,
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NULL);
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return recv_pipeline_link_elements (self, error);
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}
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/** TODO: we're describing the desired state (not the current state)
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* Prepares a skeleton receiver pipeline which can later be
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* used to plug codec specific element in.
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* This pipeline just consists of (minimally linked) rtpbin
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* audio sink and two udpsrc elements, one for RTP and one for RTCP.
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*
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* The pipeline will be started and stopped to let the OS allocate
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* sockets for us instead of building and providing GSockets ourselves
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* by hand. These GSockets will later be reused for any outgoing
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* traffic for of our hole punching scheme as a simple NAT traversal
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* technique.
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*/
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static gboolean
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recv_pipeline_init (CallsSipMediaPipeline *self,
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GCancellable *cancellable,
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GError **error)
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{
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const char *env_var;
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g_assert (CALLS_SIP_MEDIA_PIPELINE (self));
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self->recv_pipeline = gst_pipeline_new ("rtp-recv-pipeline");
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if (!self->recv_pipeline) {
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if (error)
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g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
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"Could not create receiver pipeline");
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return FALSE;
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}
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gst_object_ref_sink (self->recv_pipeline);
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env_var = g_getenv ("CALLS_AUDIOSINK");
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if (!STR_IS_NULL_OR_EMPTY (env_var)) {
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MAKE_ELEMENT (audiosink, env_var, "audiosink");
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} else {
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g_autoptr (GstStructure) gst_props = NULL;
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MAKE_ELEMENT (audiosink, "pulsesink", "audiosink");
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/* enable echo cancellation and set buffer size to 40ms */
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gst_props = gst_structure_new ("props",
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"media.role", G_TYPE_STRING, "phone",
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"filter.want", G_TYPE_STRING, "echo-cancel",
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NULL);
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g_object_set (self->audiosink,
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"buffer-time", (gint64) 40000,
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"stream-properties", gst_props,
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NULL);
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}
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MAKE_ELEMENT (recv_rtpbin, "rtpbin", "recv-rtpbin")
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MAKE_ELEMENT (rtp_src, "udpsrc", "rtp-udp-src");
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MAKE_ELEMENT (rtcp_recv_src, "udpsrc", "rtcp-udp-recv-src");
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MAKE_ELEMENT (rtcp_recv_sink, "udpsink", "rtcp-udp-recv-sink");
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g_object_set (self->rtcp_recv_sink,
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"async", FALSE,
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"sync", FALSE,
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NULL);
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g_object_bind_property (self, "lport-rtp",
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self->rtp_src, "port",
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G_BINDING_BIDIRECTIONAL);
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g_object_bind_property (self, "lport-rtcp",
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self->rtcp_recv_src, "port",
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G_BINDING_BIDIRECTIONAL);
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g_object_bind_property (self, "rport-rtcp",
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self->rtcp_recv_sink, "port",
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G_BINDING_BIDIRECTIONAL);
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g_object_bind_property (self, "remote",
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self->rtcp_recv_sink, "host",
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G_BINDING_BIDIRECTIONAL);
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gst_bin_add (GST_BIN (self->recv_pipeline), self->recv_rtpbin);
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gst_bin_add_many (GST_BIN (self->recv_pipeline), self->rtp_src,
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self->rtcp_recv_src, self->rtcp_recv_sink, NULL);
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/* TODO this should be delayed until negotiation is complete */
|
||||
if (!recv_pipeline_setup_codecs (self, self->codec, error))
|
||||
return FALSE;
|
||||
|
||||
|
||||
/* TODO use temporary bus watch for the initial pipeline start/stop */
|
||||
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);
|
||||
|
||||
|
||||
return TRUE;
|
||||
}
|
||||
|
||||
|
||||
static void
|
||||
calls_sip_media_pipeline_get_property (GObject *object,
|
||||
guint property_id,
|
||||
|
|
@ -355,342 +765,26 @@ calls_sip_media_pipeline_init (CallsSipMediaPipeline *self)
|
|||
|
||||
|
||||
static gboolean
|
||||
initable_init (GInitable *initable,
|
||||
GCancellable *cancelable,
|
||||
GError **error)
|
||||
pipelines_initable_init (GInitable *initable,
|
||||
GCancellable *cancellable,
|
||||
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");
|
||||
if (!recv_pipeline_init (self, cancellable, error))
|
||||
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");
|
||||
if (!send_pipeline_init (self, cancellable, error))
|
||||
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;
|
||||
iface->init = pipelines_initable_init;
|
||||
}
|
||||
|
||||
|
||||
|
|
@ -843,3 +937,5 @@ calls_sip_media_pipeline_pause (CallsSipMediaPipeline *self,
|
|||
self->is_running = !self->is_running;
|
||||
}
|
||||
|
||||
|
||||
#undef MAKE_ELEMENT
|
||||
|
|
|
|||
Loading…
Reference in a new issue