/dports/audio/gstreamer1-plugins-lv2/gst-plugins-bad-1.16.2/tests/examples/webrtc/ |
H A D | webrtctransceiver.c | 12 _bus_watch (GstBus * bus, GstMessage * msg, GstElement * pipe) in _bus_watch() 64 GstElement *out; in _webrtc_pad_added() local
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/dports/audio/gstreamer1-plugins-modplug/gst-plugins-bad-1.16.2/tests/check/elements/ |
H A D | curlhttpsrc.c | 63 do_get (GioHttpServer * server, const HttpRequest * req, GOutputStream * out) in do_get() 118 send_error (GOutputStream * out, int error_code, const gchar * reason) in send_error() 135 GOutputStream *out; in server_callback() local 271 GstMessage *msg; in run_test() local 547 bus_message (GstBus * bus, GstMessage * msg, gpointer user_data) in bus_message()
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/dports/audio/gstreamer1-plugins-modplug/gst-plugins-bad-1.16.2/tests/examples/webrtc/ |
H A D | webrtc.c | 12 _bus_watch (GstBus * bus, GstMessage * msg, GstElement * pipe) in _bus_watch() 64 GstElement *out; in _webrtc_pad_added() local
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H A D | webrtcbidirectional.c | 12 _bus_watch (GstBus * bus, GstMessage * msg, GstElement * pipe) in _bus_watch() 64 GstElement *out; in _webrtc_pad_added() local
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H A D | webrtcswap.c | 12 _bus_watch (GstBus * bus, GstMessage * msg, GstElement * pipe) in _bus_watch() 64 GstElement *out = NULL; in _webrtc_pad_added() local
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H A D | webrtctransceiver.c | 12 _bus_watch (GstBus * bus, GstMessage * msg, GstElement * pipe) in _bus_watch() 64 GstElement *out; in _webrtc_pad_added() local
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/dports/audio/gstreamer1-plugins-musepack/gst-plugins-bad-1.16.2/tests/check/elements/ |
H A D | curlhttpsrc.c | 63 do_get (GioHttpServer * server, const HttpRequest * req, GOutputStream * out) in do_get() 118 send_error (GOutputStream * out, int error_code, const gchar * reason) in send_error() 135 GOutputStream *out; in server_callback() local 271 GstMessage *msg; in run_test() local 547 bus_message (GstBus * bus, GstMessage * msg, gpointer user_data) in bus_message()
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/dports/audio/gstreamer1-plugins-musepack/gst-plugins-bad-1.16.2/tests/examples/webrtc/ |
H A D | webrtc.c | 12 _bus_watch (GstBus * bus, GstMessage * msg, GstElement * pipe) in _bus_watch() 64 GstElement *out; in _webrtc_pad_added() local
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H A D | webrtcbidirectional.c | 12 _bus_watch (GstBus * bus, GstMessage * msg, GstElement * pipe) in _bus_watch() 64 GstElement *out; in _webrtc_pad_added() local
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H A D | webrtcswap.c | 12 _bus_watch (GstBus * bus, GstMessage * msg, GstElement * pipe) in _bus_watch() 64 GstElement *out = NULL; in _webrtc_pad_added() local
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H A D | webrtctransceiver.c | 12 _bus_watch (GstBus * bus, GstMessage * msg, GstElement * pipe) in _bus_watch() 64 GstElement *out; in _webrtc_pad_added() local
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/dports/audio/gstreamer1-plugins-ogg/gst-plugins-base-1.16.2/ext/alsa/ |
H A D | gstalsasink.c | 576 gchar *msg = NULL; in set_hwparams() local 899 char *msg = NULL; in gst_alsasink_prepare() local 1151 GstBuffer *out; in gst_alsasink_payload() local
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/dports/audio/gstreamer1-plugins-ogg/gst-plugins-base-1.16.2/gst-libs/gst/rtsp/ |
H A D | gstrtspconnection.c | 87 guchar out[3]; /* the size must be evenly divisible by 3 */ member 781 GstRTSPMessage *msg; in setup_tunneling() local 1357 gint out = 0; in fill_raw_bytes() local 1404 gint out = 0; in fill_bytes() local 2090 parse_request_line (guint8 * buffer, GstRTSPMessage * msg) in parse_request_line() 2149 parse_line (guint8 * buffer, GstRTSPMessage * msg) in parse_line() 2609 GstRTSPMessage *msg; in gen_tunnel_reply() local 3819 GstRTSPSerializedMessage *msg; in gst_rtsp_source_dispatch_write() local 4070 GstRTSPSerializedMessage *msg = in gst_rtsp_source_dispatch_write() local 4087 GstRTSPSerializedMessage *msg; in gst_rtsp_source_finalize() local [all …]
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/dports/audio/gstreamer1-plugins-ogg/gst-plugins-base-1.16.2/tests/check/elements/ |
H A D | audiorate.c | 181 GstMessage *msg; in do_perfect_stream_test() local 502 gint64 drop, in, out; in GST_START_TEST() local
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H A D | videorate.c | 96 guint64 in, out, dropped, duplicated; in assert_videorate_stats() local 854 GstMessage *msg; in GST_START_TEST() local
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/dports/audio/gstreamer1-plugins-ogg/gst-plugins-base-1.16.2/tests/check/pipelines/ |
H A D | tcp.c | 106 GstSample *out; in symmetry_test_assert_passthrough() local 318 GSocketControlMessage *msg; in GST_START_TEST() local 321 GstSample *out; in GST_START_TEST() local
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/dports/audio/gstreamer1-plugins-openmpt/gst-plugins-bad-1.16.2/tests/check/elements/ |
H A D | curlhttpsrc.c | 63 do_get (GioHttpServer * server, const HttpRequest * req, GOutputStream * out) in do_get() 118 send_error (GOutputStream * out, int error_code, const gchar * reason) in send_error() 135 GOutputStream *out; in server_callback() local 271 GstMessage *msg; in run_test() local 547 bus_message (GstBus * bus, GstMessage * msg, gpointer user_data) in bus_message()
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/dports/audio/gstreamer1-plugins-openmpt/gst-plugins-bad-1.16.2/tests/examples/webrtc/ |
H A D | webrtc.c | 12 _bus_watch (GstBus * bus, GstMessage * msg, GstElement * pipe) in _bus_watch() 64 GstElement *out; in _webrtc_pad_added() local
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H A D | webrtcbidirectional.c | 12 _bus_watch (GstBus * bus, GstMessage * msg, GstElement * pipe) in _bus_watch() 64 GstElement *out; in _webrtc_pad_added() local
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H A D | webrtcswap.c | 12 _bus_watch (GstBus * bus, GstMessage * msg, GstElement * pipe) in _bus_watch() 64 GstElement *out = NULL; in _webrtc_pad_added() local
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H A D | webrtctransceiver.c | 12 _bus_watch (GstBus * bus, GstMessage * msg, GstElement * pipe) in _bus_watch() 64 GstElement *out; in _webrtc_pad_added() local
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/dports/audio/gstreamer1-plugins-opus/gst-plugins-base-1.16.2/ext/alsa/ |
H A D | gstalsasink.c | 576 gchar *msg = NULL; in set_hwparams() local 899 char *msg = NULL; in gst_alsasink_prepare() local 1151 GstBuffer *out; in gst_alsasink_payload() local
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/dports/audio/gstreamer1-plugins-opus/gst-plugins-base-1.16.2/gst-libs/gst/rtsp/ |
H A D | gstrtspconnection.c | 87 guchar out[3]; /* the size must be evenly divisible by 3 */ member 781 GstRTSPMessage *msg; in setup_tunneling() local 1357 gint out = 0; in fill_raw_bytes() local 1404 gint out = 0; in fill_bytes() local 2090 parse_request_line (guint8 * buffer, GstRTSPMessage * msg) in parse_request_line() 2149 parse_line (guint8 * buffer, GstRTSPMessage * msg) in parse_line() 2609 GstRTSPMessage *msg; in gen_tunnel_reply() local 3819 GstRTSPSerializedMessage *msg; in gst_rtsp_source_dispatch_write() local 4070 GstRTSPSerializedMessage *msg = in gst_rtsp_source_dispatch_write() local 4087 GstRTSPSerializedMessage *msg; in gst_rtsp_source_finalize() local [all …]
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/dports/audio/gstreamer1-plugins-opus/gst-plugins-base-1.16.2/tests/check/elements/ |
H A D | audiorate.c | 181 GstMessage *msg; in do_perfect_stream_test() local 502 gint64 drop, in, out; in GST_START_TEST() local
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H A D | videorate.c | 96 guint64 in, out, dropped, duplicated; in assert_videorate_stats() local 854 GstMessage *msg; in GST_START_TEST() local
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