| /third_party/gstreamer/gstplugins_good/gst/rtp/ |
| D | gstrtph263ppay.c | 302 GstStructure *new_s = gst_structure_new ("video/x-h263", in gst_rtp_h263p_pay_sink_getcaps() local
391 gst_structure_set_value (new_s, "h263version", &list); in gst_rtp_h263p_pay_sink_getcaps()
394 gst_structure_set (new_s, "h263version", G_TYPE_STRING, "h263", NULL); in gst_rtp_h263p_pay_sink_getcaps()
399 gst_structure_set (new_s, "annex-f", G_TYPE_BOOLEAN, FALSE, NULL); in gst_rtp_h263p_pay_sink_getcaps()
401 gst_structure_set (new_s, "annex-i", G_TYPE_BOOLEAN, FALSE, NULL); in gst_rtp_h263p_pay_sink_getcaps()
403 gst_structure_set (new_s, "annex-j", G_TYPE_BOOLEAN, FALSE, NULL); in gst_rtp_h263p_pay_sink_getcaps()
405 gst_structure_set (new_s, "annex-t", G_TYPE_BOOLEAN, FALSE, NULL); in gst_rtp_h263p_pay_sink_getcaps()
407 gst_structure_set (new_s, "annex-l", G_TYPE_BOOLEAN, FALSE, NULL); in gst_rtp_h263p_pay_sink_getcaps()
409 gst_structure_set (new_s, "annex-v", G_TYPE_BOOLEAN, FALSE, NULL); in gst_rtp_h263p_pay_sink_getcaps()
413 gst_structure_set (new_s, in gst_rtp_h263p_pay_sink_getcaps()
[all …]
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| D | gstrtph264pay.c | 323 GstStructure *new_s = gst_structure_new_empty ("video/x-h264"); in gst_rtp_h264_pay_getcaps() local
347 gst_structure_set (new_s, "profile", G_TYPE_STRING, profile, NULL); in gst_rtp_h264_pay_getcaps()
361 gst_structure_take_value (new_s, "profile", &profiles); in gst_rtp_h264_pay_getcaps()
365 gst_structure_set (new_s, "level", G_TYPE_STRING, level, NULL); in gst_rtp_h264_pay_getcaps()
380 gst_structure_take_value (new_s, "level", &levels); in gst_rtp_h264_pay_getcaps()
385 gst_structure_set (new_s, in gst_rtp_h264_pay_getcaps()
389 gst_structure_set (new_s, "profile", G_TYPE_STRING, profile, NULL); in gst_rtp_h264_pay_getcaps()
393 gst_structure_set (new_s, in gst_rtp_h264_pay_getcaps()
398 caps = gst_caps_merge_structure (caps, new_s); in gst_rtp_h264_pay_getcaps()
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| D | gstrtph265pay.c | 404 GstStructure *new_s = gst_structure_new_empty ("video/x-h265"); in gst_rtp_h265_pay_getcaps() local
415 gst_structure_set (new_s, "profile", G_TYPE_STRING, profile, NULL); in gst_rtp_h265_pay_getcaps()
429 gst_structure_set (new_s, "tier", G_TYPE_STRING, tier, NULL); in gst_rtp_h265_pay_getcaps()
442 gst_structure_set (new_s, "level", G_TYPE_STRING, level, NULL); in gst_rtp_h265_pay_getcaps()
457 gst_structure_take_value (new_s, "level", &levels); in gst_rtp_h265_pay_getcaps()
466 caps = gst_caps_merge_structure (caps, new_s); in gst_rtp_h265_pay_getcaps()
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| /third_party/nghttp2/third-party/url-parser/ |
| D | url_parser.c | 458 enum http_host_state new_s = http_parse_host_char(s, *p); in http_parse_host() local
460 if (new_s == s_http_host_dead) { in http_parse_host()
464 switch(new_s) { in http_parse_host()
505 s = new_s; in http_parse_host()
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| /third_party/skia/third_party/externals/harfbuzz/src/ |
| D | hb-ot-shape-complex-hangul.cc | 323 hb_codepoint_t new_s = s + new_tindex; in preprocess_text_hangul() local
324 if (font->has_glyph (new_s)) in preprocess_text_hangul()
326 (void) buffer->replace_glyphs (2, 1, &new_s); in preprocess_text_hangul()
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| /third_party/gstreamer/gstplugins_bad/ext/webrtc/ |
| D | gstwebrtcbin.c | 1367 gchar *old_s, *new_s; in _update_ice_gathering_state_task() local
1371 new_s = _enum_value_to_string (GST_TYPE_WEBRTC_ICE_GATHERING_STATE, in _update_ice_gathering_state_task()
1374 old_s, old_state, new_s, new_state); in _update_ice_gathering_state_task()
1376 g_free (new_s); in _update_ice_gathering_state_task()
1403 gchar *old_s, *new_s; in _update_ice_connection_state_task() local
1407 new_s = _enum_value_to_string (GST_TYPE_WEBRTC_ICE_CONNECTION_STATE, in _update_ice_connection_state_task()
1411 new_s, new_state); in _update_ice_connection_state_task()
1413 g_free (new_s); in _update_ice_connection_state_task()
1440 gchar *old_s, *new_s; in _update_peer_connection_state_task() local
1444 new_s = _enum_value_to_string (GST_TYPE_WEBRTC_PEER_CONNECTION_STATE, in _update_peer_connection_state_task()
[all …]
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| /third_party/gstreamer/gstplugins_base/gst-libs/gst/audio/ |
| D | gstaudioaggregator.c | 973 GstStructure *new_s = in gst_audio_aggregator_convert_sink_getcaps() local
975 gst_structure_set_value (new_s, "rate", gst_structure_get_value (s, in gst_audio_aggregator_convert_sink_getcaps()
977 sink_caps = gst_caps_merge_structure (sink_caps, new_s); in gst_audio_aggregator_convert_sink_getcaps()
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| /third_party/gstreamer/gstplugins_good/gst/matroska/ |
| D | matroska-mux.c | 1050 GstStructure *old_s, *new_s; in check_new_caps() local
1056 new_s = gst_caps_get_structure (new_caps, 0); in check_new_caps()
1059 gst_structure_filter_and_map_in_place (new_s, in check_new_caps()
1060 (GstStructureFilterMapFunc) check_field, new_s); in check_new_caps()
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| /third_party/googletest/googlemock/test/ |
| D | gmock-matchers-containers_test.cc | 625 void set_s(const std::string& new_s) { s_ = new_s; } in set_s() argument
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