/third_party/gstreamer/gstplugins_good/gst/udp/ |
D | gstudpsrc.c | 512 GstUDPSrc *udpsrc; in gst_udpsrc_decide_allocation() local 518 udpsrc = GST_UDPSRC (bsrc); in gst_udpsrc_decide_allocation() 532 gst_buffer_pool_config_set_params (config, caps, udpsrc->mtu, 0, 0); in gst_udpsrc_decide_allocation() 537 gst_query_set_nth_allocation_pool (query, 0, pool, udpsrc->mtu, 0, 0); in gst_udpsrc_decide_allocation() 539 gst_query_add_allocation_pool (query, pool, udpsrc->mtu, 0, 0); in gst_udpsrc_decide_allocation() 619 GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (udpsrc, "udpsrc", GST_RANK_NONE, 803 gst_udpsrc_init (GstUDPSrc * udpsrc) in gst_udpsrc_init() argument 805 udpsrc->uri = in gst_udpsrc_init() 809 udpsrc->address = g_strdup (UDP_DEFAULT_MULTICAST_GROUP); in gst_udpsrc_init() 810 udpsrc->port = UDP_DEFAULT_PORT; in gst_udpsrc_init() [all …]
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D | gstudpelements.h | 37 GST_ELEMENT_REGISTER_DECLARE (udpsrc);
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D | gstudp.c | 35 ret |= GST_ELEMENT_REGISTER (udpsrc, plugin); in plugin_init()
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/third_party/gstreamer/gstplugins_good/tests/check/elements/ |
D | udpsrc.c | 29 udpsrc_setup (GstElement ** udpsrc, GSocket ** socket, in udpsrc_setup() argument 36 *udpsrc = gst_check_setup_element ("udpsrc"); in udpsrc_setup() 37 fail_unless (*udpsrc != NULL); in udpsrc_setup() 38 g_object_set (*udpsrc, "port", 0, NULL); in udpsrc_setup() 40 *sinkpad = gst_check_setup_sink_pad_by_name (*udpsrc, &sinktemplate, "src"); in udpsrc_setup() 44 gst_element_set_state (*udpsrc, GST_STATE_PLAYING); in udpsrc_setup() 45 g_object_get (*udpsrc, "port", &port, NULL); in udpsrc_setup() 69 GstElement *udpsrc = NULL; in GST_START_TEST() local 73 if (!udpsrc_setup (&udpsrc, &socket, &sinkpad, &sa)) in GST_START_TEST() 126 gst_element_set_state (udpsrc, GST_STATE_NULL); in GST_START_TEST() [all …]
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/third_party/gstreamer/gstplugins_bad/gst/sdp/ |
D | gstsdpdemux.c | 283 if (stream->udpsrc[0] == src) in find_stream_by_udpsrc() 285 if (stream->udpsrc[1] == src) in find_stream_by_udpsrc() 315 GstElement *udpsrc = stream->udpsrc[i]; in gst_sdp_demux_stream_free() local 318 if (udpsrc) { in gst_sdp_demux_stream_free() 319 gst_element_set_state (udpsrc, GST_STATE_NULL); in gst_sdp_demux_stream_free() 320 gst_bin_remove (GST_BIN_CAST (demux), udpsrc); in gst_sdp_demux_stream_free() 321 stream->udpsrc[i] = NULL; in gst_sdp_demux_stream_free() 542 g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", (guint64) 0, NULL); in new_session_pad() 783 stream->udpsrc[0] = in gst_sdp_demux_stream_configure_udp() 786 if (stream->udpsrc[0] == NULL) in gst_sdp_demux_stream_configure_udp() [all …]
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D | gstsdpdemux.h | 66 GstElement *udpsrc[2]; member
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/third_party/gstreamer/gstplugins_good/tests/examples/rtp/ |
D | client-H263p-AMR.sh | 16 udpsrc caps=$VIDEO_CAPS port=5000 ! rtpbin.recv_rtp_sink_0 \ 18 udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ 20 udpsrc caps=$AUDIO_CAPS port=5002 ! rtpbin.recv_rtp_sink_1 \ 22 udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \
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D | client-H263p-PCMA.sh | 21 udpsrc caps=$VIDEO_CAPS port=5000 ! rtpbin.recv_rtp_sink_0 \ 23 udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ 25 udpsrc caps=$AUDIO_CAPS port=5002 ! rtpbin.recv_rtp_sink_1 \ 27 udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \
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D | client-H264-PCMA.sh | 57 udpsrc caps=$VIDEO_CAPS port=5000 ! rtpbin.recv_rtp_sink_0 \ 59 udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ 61 udpsrc caps=$AUDIO_CAPS port=5002 ! rtpbin.recv_rtp_sink_1 \ 63 udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \
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D | client-VP8-OPUS.sh | 21 … udpsrc caps=$VIDEO_CAPS address=$SRC port=5000 ! rtpbin.recv_rtp_sink_0 \ 23 … udpsrc address=$SRC port=5001 ! rtpbin.recv_rtcp_sink_0 \ 25 udpsrc caps=$AUDIO_CAPS address=$SRC port=5002 ! rtpbin.recv_rtp_sink_1 \ 27 … udpsrc address=$SRC port=5003 ! rtpbin.recv_rtcp_sink_1 \
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D | client-PCMA.sh | 36 udpsrc caps=$AUDIO_CAPS port=5002 ! rtpbin.recv_rtp_sink_0 \ 38 udpsrc port=5003 ! rtpbin.recv_rtcp_sink_0 \
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D | client-H264.sh | 39 udpsrc caps=$VIDEO_CAPS port=5000 ! rtpbin.recv_rtp_sink_0 \ 41 udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \
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D | client-H263p.sh | 18 udpsrc caps=$VIDEO_CAPS port=5000 ! rtpbin.recv_rtp_sink_0 \ 20 udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \
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D | client-H264-rtx.sh | 39 udpsrc caps=$VIDEO_CAPS port=5000 ! rtpbin.recv_rtp_sink_0 \ 41 udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \
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D | server-VTS-H263p-ATS-PCMA.sh | 12 udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \ 16 udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1
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D | server-VTS-VP8-ATS-OPUS.sh | 14 … udpsrc address=$SRC port=5005 ! rtpbin.recv_rtcp_sink_0 \ 18 udpsrc address=$SRC port=5007 ! rtpbin.recv_rtcp_sink_1
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D | server-v4l2-H263p-alsasrc-AMR.sh | 21 … udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \ 25 udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1
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D | server-decodebin-H263p-AMR.sh | 43 … udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \ 47 udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1
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D | server-alsasrc-PCMA.sh | 35 udpsrc port=5007 ! rtpbin.recv_rtcp_sink_0
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/third_party/pulseaudio/src/modules/rtp/ |
D | rtp-gstreamer.c | 423 GstElement *udpsrc = NULL, *rtpbin = NULL, *depay = NULL, *appsink = NULL; in init_receive_pipeline() local 430 MAKE_ELEMENT(udpsrc, "udpsrc"); in init_receive_pipeline() 443 gst_bin_add_many(GST_BIN(c->pipeline), udpsrc, rtpbin, depay, appsink, NULL); in init_receive_pipeline() 461 …g_object_set(udpsrc, "socket", socket, "caps", caps, "auto-multicast" /* caller handles this */, F… in init_receive_pipeline() 481 if (!gst_element_link_pads(udpsrc, "src", rtpbin, "recv_rtp_sink_0") || in init_receive_pipeline() 490 if (!gst_element_link_pads(udpsrc, "src", rtpbin, "recv_rtp_sink_0") || in init_receive_pipeline() 504 pad = gst_element_get_static_pad(udpsrc, "src"); in init_receive_pipeline() 522 if (udpsrc) in init_receive_pipeline() 523 gst_object_unref(udpsrc); in init_receive_pipeline()
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/third_party/gstreamer/gstplugins_good/gst/rtsp/ |
D | gstrtspsrc.c | 2012 if (stream->udpsrc[0] == src) in find_stream_by_udpsrc() 2014 if (stream->udpsrc[1] == src) in find_stream_by_udpsrc() 2441 if (stream->udpsrc[i]) { in gst_rtspsrc_stream_free() 2442 gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL); in gst_rtspsrc_stream_free() 2443 if (gst_object_has_as_parent (GST_OBJECT (stream->udpsrc[i]), in gst_rtspsrc_stream_free() 2445 gst_bin_remove (GST_BIN_CAST (src), stream->udpsrc[i]); in gst_rtspsrc_stream_free() 2446 gst_object_unref (stream->udpsrc[i]); in gst_rtspsrc_stream_free() 2681 stream->udpsrc[0] = gst_object_ref_sink (udpsrc0); in gst_rtspsrc_alloc_udp_ports() 2682 stream->udpsrc[1] = gst_object_ref_sink (udpsrc1); in gst_rtspsrc_alloc_udp_ports() 2683 gst_element_set_locked_state (stream->udpsrc[0], TRUE); in gst_rtspsrc_alloc_udp_ports() [all …]
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D | gstrtspsrc.h | 117 GstElement *udpsrc[2]; member
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D | README | 314 optionally udpsrc elements can be configured to receive client RTP and 338 optionally udpsrc elements can be configured to receive client RTP and
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/third_party/gstreamer/gstplugins_good/gst/rtp/ |
D | README | 207 gst-launch-1.0 udpsrc caps="application/x-rtp, media=(string)video, 240 gst-launch-1.0 udpsrc caps="application/x-rtp, media=(string)video, 292 gst-launch-1.0 -v udpsrc caps="application/x-rtp, media=(string)video, 323 udpsrc port=5000 caps="application/x-rtp, media=(string)video, payload=(int)96, 328 udpsrc port=5002 caps="application/x-rtp, media=(string)audio, payload=(int)96, 342 recommended to use a gstrtpjitterbuffer after the udpsrc elements.
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/third_party/gstreamer/gstplugins_bad/ext/dtls/ |
D | README | 105 'udpsrc port={port}',
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