using Gee; using Xmpp; public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream { public uint8 rtpid { get; private set; } public Plugin plugin { get; private set; } public Gst.Pipeline pipe { get { return plugin.pipe; }} public Gst.Element rtpbin { get { return plugin.rtpbin; }} public CodecUtil codec_util { get { return plugin.codec_util; }} private Gst.App.Sink send_rtp; private Gst.App.Sink send_rtcp; private Gst.App.Src recv_rtp; private Gst.App.Src recv_rtcp; private Gst.Element encode; private Gst.Element decode; private Gst.Element input; private Gst.Element output; private Device _input_device; public Device input_device { get { return _input_device; } set { if (!paused) { if (this._input_device != null) { this._input_device.unlink(); this._input_device = null; } set_input(value != null ? value.link_source() : null); } this._input_device = value; }} private Device _output_device; public Device output_device { get { return _output_device; } set { if (output != null) remove_output(output); if (value != null) add_output(value.link_sink()); this._output_device = value; }} public bool created { get; private set; default = false; } public bool paused { get; private set; default = false; } private bool push_recv_data = false; private string participant_ssrc = null; private Gst.Pad recv_rtcp_sink_pad; private Gst.Pad recv_rtp_sink_pad; private Gst.Pad recv_rtp_src_pad; private Gst.Pad send_rtcp_src_pad; private Gst.Pad send_rtp_sink_pad; private Gst.Pad send_rtp_src_pad; private Crypto.Srtp.Session? local_crypto_session; private Crypto.Srtp.Session? remote_crypto_session; public Stream(Plugin plugin, Xmpp.Xep.Jingle.Content content) { base(content); this.plugin = plugin; this.rtpid = plugin.next_free_id(); content.notify["senders"].connect_after(on_senders_changed); } public void on_senders_changed() { if (sending && input == null) { input_device = plugin.get_preferred_device(media, false); } if (receiving && output == null) { output_device = plugin.get_preferred_device(media, true); } } public override void create() { plugin.pause(); // Create i/o if needed if (input == null && input_device == null && sending) { input_device = plugin.get_preferred_device(media, false); } if (output == null && output_device == null && receiving && media == "audio") { output_device = plugin.get_preferred_device(media, true); } // Create app elements send_rtp = Gst.ElementFactory.make("appsink", @"rtp-sink-$rtpid") as Gst.App.Sink; send_rtp.async = false; send_rtp.caps = CodecUtil.get_caps(media, payload_type); send_rtp.emit_signals = true; send_rtp.sync = false; send_rtp.new_sample.connect(on_new_sample); pipe.add(send_rtp); send_rtcp = Gst.ElementFactory.make("appsink", @"rtcp-sink-$rtpid") as Gst.App.Sink; send_rtcp.async = false; send_rtcp.caps = new Gst.Caps.empty_simple("application/x-rtcp"); send_rtcp.emit_signals = true; send_rtcp.sync = false; send_rtcp.new_sample.connect(on_new_sample); pipe.add(send_rtcp); recv_rtp = Gst.ElementFactory.make("appsrc", @"rtp-src-$rtpid") as Gst.App.Src; recv_rtp.caps = CodecUtil.get_caps(media, payload_type); recv_rtp.do_timestamp = true; recv_rtp.format = Gst.Format.TIME; recv_rtp.is_live = true; pipe.add(recv_rtp); recv_rtcp = Gst.ElementFactory.make("appsrc", @"rtcp-src-$rtpid") as Gst.App.Src; recv_rtcp.caps = new Gst.Caps.empty_simple("application/x-rtcp"); recv_rtcp.do_timestamp = true; recv_rtcp.format = Gst.Format.TIME; recv_rtcp.is_live = true; pipe.add(recv_rtcp); // Connect RTCP send_rtcp_src_pad = rtpbin.get_request_pad(@"send_rtcp_src_$rtpid"); send_rtcp_src_pad.link(send_rtcp.get_static_pad("sink")); recv_rtcp_sink_pad = rtpbin.get_request_pad(@"recv_rtcp_sink_$rtpid"); recv_rtcp.get_static_pad("src").link(recv_rtcp_sink_pad); // Connect input encode = codec_util.get_encode_bin(media, payload_type, @"encode-$rtpid"); pipe.add(encode); send_rtp_sink_pad = rtpbin.get_request_pad(@"send_rtp_sink_$rtpid"); encode.get_static_pad("src").link(send_rtp_sink_pad); if (input != null) { input.link(encode); } // Connect output decode = codec_util.get_decode_bin(media, payload_type, @"decode-$rtpid"); pipe.add(decode); if (output != null) { decode.link(output); } // Connect RTP recv_rtp_sink_pad = rtpbin.get_request_pad(@"recv_rtp_sink_$rtpid"); recv_rtp.get_static_pad("src").link(recv_rtp_sink_pad); created = true; push_recv_data = true; plugin.unpause(); } private void prepare_local_crypto() { if (local_crypto != null && local_crypto_session == null) { local_crypto_session = new Crypto.Srtp.Session( local_crypto.crypto_suite == Xep.JingleRtp.Crypto.F8_128_HMAC_SHA1_80 ? Crypto.Srtp.Encryption.AES_F8 : Crypto.Srtp.Encryption.AES_CM, Crypto.Srtp.Authentication.HMAC_SHA1, local_crypto.crypto_suite == Xep.JingleRtp.Crypto.AES_CM_128_HMAC_SHA1_32 ? 4 : 10, Crypto.Srtp.Prf.AES_CM, 0 ); local_crypto_session.setkey(local_crypto.key, local_crypto.salt); debug("Setting up encryption with key params %s", local_crypto.key_params); } } private Gst.FlowReturn on_new_sample(Gst.App.Sink sink) { if (sink == null) { debug("Sink is null"); return Gst.FlowReturn.EOS; } Gst.Sample sample = sink.pull_sample(); Gst.Buffer buffer = sample.get_buffer(); uint8[] data; buffer.extract_dup(0, buffer.get_size(), out data); prepare_local_crypto(); if (sink == send_rtp) { if (local_crypto_session != null) { data = local_crypto_session.encrypt_rtp(data, local_crypto.crypto_suite == Xep.JingleRtp.Crypto.AES_CM_128_HMAC_SHA1_32 ? 4 : 10); } on_send_rtp_data(new Bytes.take(data)); } else if (sink == send_rtcp) { if (local_crypto_session != null) { data = local_crypto_session.encrypt_rtcp(data, local_crypto.crypto_suite == Xep.JingleRtp.Crypto.AES_CM_128_HMAC_SHA1_32 ? 4 : 10); } on_send_rtcp_data(new Bytes.take(data)); } else { warning("unknown sample"); } return Gst.FlowReturn.OK; } private static Gst.PadProbeReturn drop_probe() { return Gst.PadProbeReturn.DROP; } public override void destroy() { // Stop network communication push_recv_data = false; recv_rtp.end_of_stream(); recv_rtcp.end_of_stream(); send_rtp.new_sample.disconnect(on_new_sample); send_rtcp.new_sample.disconnect(on_new_sample); // Disconnect input device if (input != null) { input.unlink(encode); input = null; } if (this._input_device != null) { if (!paused) this._input_device.unlink(); this._input_device = null; } // Disconnect encode encode.set_locked_state(true); encode.set_state(Gst.State.NULL); encode.get_static_pad("src").unlink(send_rtp_sink_pad); pipe.remove(encode); encode = null; // Disconnect RTP sending if (send_rtp_src_pad != null) { send_rtp_src_pad.add_probe(Gst.PadProbeType.BLOCK, drop_probe); send_rtp_src_pad.unlink(send_rtp.get_static_pad("sink")); } send_rtp.set_locked_state(true); send_rtp.set_state(Gst.State.NULL); pipe.remove(send_rtp); send_rtp = null; // Disconnect decode if (recv_rtp_src_pad != null) { recv_rtp_src_pad.add_probe(Gst.PadProbeType.BLOCK, drop_probe); recv_rtp_src_pad.unlink(decode.get_static_pad("sink")); } // Disconnect RTP receiving recv_rtp.set_locked_state(true); recv_rtp.set_state(Gst.State.NULL); recv_rtp.get_static_pad("src").unlink(recv_rtp_sink_pad); pipe.remove(recv_rtp); recv_rtp = null; // Disconnect output if (output != null) { decode.unlink(output); } decode.set_locked_state(true); decode.set_state(Gst.State.NULL); pipe.remove(decode); decode = null; output = null; // Disconnect output device if (this._output_device != null) { this._output_device.unlink(); this._output_device = null; } // Disconnect RTCP receiving recv_rtcp.get_static_pad("src").unlink(recv_rtcp_sink_pad); recv_rtcp.set_locked_state(true); recv_rtcp.set_state(Gst.State.NULL); pipe.remove(recv_rtcp); recv_rtcp = null; // Disconnect RTCP sending send_rtcp_src_pad.unlink(send_rtcp.get_static_pad("sink")); send_rtcp.set_locked_state(true); send_rtcp.set_state(Gst.State.NULL); pipe.remove(send_rtcp); send_rtcp = null; // Release rtp pads rtpbin.release_request_pad(send_rtp_sink_pad); send_rtp_sink_pad = null; rtpbin.release_request_pad(recv_rtp_sink_pad); recv_rtp_sink_pad = null; rtpbin.release_request_pad(recv_rtcp_sink_pad); recv_rtcp_sink_pad = null; rtpbin.release_request_pad(send_rtcp_src_pad); send_rtcp_src_pad = null; send_rtp_src_pad = null; recv_rtp_src_pad = null; } private void prepare_remote_crypto() { if (remote_crypto != null && remote_crypto_session == null) { remote_crypto_session = new Crypto.Srtp.Session( remote_crypto.crypto_suite == Xep.JingleRtp.Crypto.F8_128_HMAC_SHA1_80 ? Crypto.Srtp.Encryption.AES_F8 : Crypto.Srtp.Encryption.AES_CM, Crypto.Srtp.Authentication.HMAC_SHA1, remote_crypto.crypto_suite == Xep.JingleRtp.Crypto.AES_CM_128_HMAC_SHA1_32 ? 4 : 10, Crypto.Srtp.Prf.AES_CM, 0 ); remote_crypto_session.setkey(remote_crypto.key, remote_crypto.salt); debug("Setting up decryption with key params %s", remote_crypto.key_params); } } public override void on_recv_rtp_data(Bytes bytes) { prepare_remote_crypto(); uint8[] data = bytes.get_data(); if (remote_crypto_session != null) { try { data = remote_crypto_session.decrypt_rtp(data); } catch (Error e) { warning("%s (%d)", e.message, e.code); } } if (push_recv_data) { recv_rtp.push_buffer(new Gst.Buffer.wrapped((owned) data)); } } public override void on_recv_rtcp_data(Bytes bytes) { prepare_remote_crypto(); uint8[] data = bytes.get_data(); if (remote_crypto_session != null) { try { data = remote_crypto_session.decrypt_rtcp(data); } catch (Error e) { warning("%s (%d)", e.message, e.code); } } if (push_recv_data) { recv_rtcp.push_buffer(new Gst.Buffer.wrapped((owned) data)); } } public override void on_rtp_ready() { // If full frame has been sent before the connection was ready, the counterpart would only display our video after the next full frame. // Send a full frame to let the counterpart display our video asap rtpbin.send_event(new Gst.Event.custom( Gst.EventType.CUSTOM_UPSTREAM, new Gst.Structure("GstForceKeyUnit", "all-headers", typeof(bool), true, null)) ); } public override void on_rtcp_ready() { int rtp_session_id = (int) rtpid; uint64 max_delay = int.MAX; Object rtp_session; bool rtp_sent; GLib.Signal.emit_by_name(rtpbin, "get-internal-session", rtp_session_id, out rtp_session); GLib.Signal.emit_by_name(rtp_session, "send-rtcp-full", max_delay, out rtp_sent); debug("RTCP is ready, resending rtcp: %s", rtp_sent.to_string()); } public void on_ssrc_pad_added(string ssrc, Gst.Pad pad) { participant_ssrc = ssrc; recv_rtp_src_pad = pad; if (decode != null) { plugin.pause(); debug("Link %s to %s decode for %s", recv_rtp_src_pad.name, media, name); recv_rtp_src_pad.link(decode.get_static_pad("sink")); plugin.unpause(); } } public void on_send_rtp_src_added(Gst.Pad pad) { send_rtp_src_pad = pad; if (send_rtp != null) { plugin.pause(); debug("Link %s to %s send_rtp for %s", send_rtp_src_pad.name, media, name); send_rtp_src_pad.link(send_rtp.get_static_pad("sink")); plugin.unpause(); } } public void set_input(Gst.Element? input) { set_input_and_pause(input, paused); } private void set_input_and_pause(Gst.Element? input, bool paused) { if (created && this.input != null) { this.input.unlink(encode); this.input = null; } this.input = input; this.paused = paused; if (created && sending && !paused && input != null) { plugin.pause(); input.link(encode); plugin.unpause(); } } public void pause() { if (paused) return; set_input_and_pause(null, true); if (input_device != null) input_device.unlink(); } public void unpause() { if (!paused) return; set_input_and_pause(input_device != null ? input_device.link_source() : null, false); } ulong block_probe_handler_id = 0; public virtual void add_output(Gst.Element element) { if (output != null) { critical("add_output() invoked more than once"); return; } this.output = element; if (created) { plugin.pause(); decode.link(element); if (block_probe_handler_id != 0) { decode.get_static_pad("src").remove_probe(block_probe_handler_id); } plugin.unpause(); } } public virtual void remove_output(Gst.Element element) { if (output != element) { critical("remove_output() invoked without prior add_output()"); return; } if (created) { block_probe_handler_id = decode.get_static_pad("src").add_probe(Gst.PadProbeType.BLOCK, drop_probe); decode.unlink(element); } if (this._output_device != null) { this._output_device.unlink(); this._output_device = null; } this.output = null; } } public class Dino.Plugins.Rtp.VideoStream : Stream { private Gee.List outputs = new ArrayList(); private Gst.Element output_tee; public VideoStream(Plugin plugin, Xmpp.Xep.Jingle.Content content) { base(plugin, content); if (media != "video") critical("VideoStream created for non-video media"); } public override void create() { plugin.pause(); output_tee = Gst.ElementFactory.make("tee", null); output_tee.@set("allow-not-linked", true); pipe.add(output_tee); add_output(output_tee); base.create(); foreach (Gst.Element output in outputs) { output_tee.link(output); } plugin.unpause(); } public override void destroy() { foreach (Gst.Element output in outputs) { output_tee.unlink(output); } base.destroy(); output_tee.set_locked_state(true); output_tee.set_state(Gst.State.NULL); pipe.remove(output_tee); output_tee = null; } public override void add_output(Gst.Element element) { if (element == output_tee) { base.add_output(element); return; } outputs.add(element); if (output_tee != null) { output_tee.link(element); } } public override void remove_output(Gst.Element element) { if (element == output_tee) { base.remove_output(element); return; } outputs.remove(element); if (output_tee != null) { output_tee.unlink(element); } } }