| /dports/audio/gstreamer1-plugins-opus/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-sndfile/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-sndfile/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-soundtouch/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()
|
| /dports/audio/gstreamer1-plugins-soundtouch/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
|
| /dports/audio/gstreamer1-plugins-vorbis/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-vorbis/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 …]
|
| /dports/audio/gstreamer1-plugins-vorbis/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
|
| H A D | videorate.c | 96 guint64 in, out, dropped, duplicated; in assert_videorate_stats() local
854 GstMessage *msg; in GST_START_TEST() local
|
| /dports/audio/gstreamer1-plugins-vorbis/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
|
| /dports/audio/gstreamer1-plugins-webrtcdsp/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()
|
| /dports/audio/gstreamer1-plugins-webrtcdsp/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
|
| 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
|
| 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
|
| /dports/audio/gtmixer/gtmixer-1.0.2/ |
| H A D | uthash.h | 70 #define uthash_fatal(msg) exit(-1) /* fatal error (out of memory,etc) */ argument
94 #define HASH_FIND(hh,head,keyptr,keylen,out) \ argument
241 #define HASH_FIND_STR(head,findstr,out) \ argument
245 #define HASH_FIND_INT(head,findint,out) \ argument
249 #define HASH_FIND_PTR(head,findptr,out) \ argument
580 #define HASH_FIND_IN_BKT(tbl,hh,head,keyptr,keylen_in,out) \ argument
|
| /dports/audio/guitarix-lv2/guitarix-0.43.1/src/gx_head/engine/ |
| H A D | gx_convolver.cpp | 542 boost::format msg = boost::format("failed to resample %1% -> %2%") % imprate % samplerate; in resample() local
631 float *in, *out; in compute() local
748 float *in, *in1, *out, *out1; in compute_stereo() local
|
| /dports/audio/guitarix-lv2/guitarix-0.43.1/src/gx_head/gui/ |
| H A D | gx_preset_window.cpp | 664 bool PresetWindow::run_message_dialog(Gtk::Widget& w, const Glib::ustring& msg) { in run_message_dialog()
777 FILE *out; in download_file() local
|
| /dports/audio/hexter/hexter-1.0.3/src/ |
| H A D | dx7_voice.c | 242 #define HEXTER_DEBUG_ENGINE_SLEW_CHECK(inc, msg) \ argument
245 #define HEXTER_DEBUG_ENGINE_SLEW_CHECK(inc, msg) argument
1117 int32_t phase, index, out; in dx7_lfo_update() local
|