| /dports/multimedia/gstreamer1-plugins-dts/gst-plugins-bad-1.16.2/ext/webrtc/ |
| H A D | webrtcdatachannel.c | 371 pad = gst_element_get_static_pad (channel->appsrc, "src"); in _close_sctp_stream()
520 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in _parse_control_packet()
744 if (gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), in gst_webrtc_data_channel_start_negotiation()
826 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_data()
886 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_string()
1086 channel->appsrc = gst_element_factory_make ("appsrc", NULL); in gst_webrtc_data_channel_constructed()
1087 gst_object_ref_sink (channel->appsrc); in gst_webrtc_data_channel_constructed()
1088 pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_constructed()
1110 GstPad *pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_finalize()
1126 g_clear_object (&channel->appsrc); in gst_webrtc_data_channel_finalize()
|
| /dports/multimedia/gstreamer1-plugins-dash/gst-plugins-bad-1.16.2/ext/webrtc/ |
| H A D | webrtcdatachannel.c | 371 pad = gst_element_get_static_pad (channel->appsrc, "src"); in _close_sctp_stream()
520 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in _parse_control_packet()
744 if (gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), in gst_webrtc_data_channel_start_negotiation()
826 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_data()
886 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_string()
1086 channel->appsrc = gst_element_factory_make ("appsrc", NULL); in gst_webrtc_data_channel_constructed()
1087 gst_object_ref_sink (channel->appsrc); in gst_webrtc_data_channel_constructed()
1088 pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_constructed()
1110 GstPad *pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_finalize()
1126 g_clear_object (&channel->appsrc); in gst_webrtc_data_channel_finalize()
|
| /dports/multimedia/gstreamer1-plugins-assrender/gst-plugins-bad-1.16.2/ext/webrtc/ |
| H A D | webrtcdatachannel.c | 371 pad = gst_element_get_static_pad (channel->appsrc, "src"); in _close_sctp_stream()
520 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in _parse_control_packet()
744 if (gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), in gst_webrtc_data_channel_start_negotiation()
826 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_data()
886 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_string()
1086 channel->appsrc = gst_element_factory_make ("appsrc", NULL); in gst_webrtc_data_channel_constructed()
1087 gst_object_ref_sink (channel->appsrc); in gst_webrtc_data_channel_constructed()
1088 pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_constructed()
1110 GstPad *pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_finalize()
1126 g_clear_object (&channel->appsrc); in gst_webrtc_data_channel_finalize()
|
| /dports/multimedia/gstreamer1-plugins-bad/gst-plugins-bad-1.16.2/ext/webrtc/ |
| H A D | webrtcdatachannel.c | 371 pad = gst_element_get_static_pad (channel->appsrc, "src"); in _close_sctp_stream()
520 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in _parse_control_packet()
744 if (gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), in gst_webrtc_data_channel_start_negotiation()
826 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_data()
886 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_string()
1086 channel->appsrc = gst_element_factory_make ("appsrc", NULL); in gst_webrtc_data_channel_constructed()
1087 gst_object_ref_sink (channel->appsrc); in gst_webrtc_data_channel_constructed()
1088 pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_constructed()
1110 GstPad *pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_finalize()
1126 g_clear_object (&channel->appsrc); in gst_webrtc_data_channel_finalize()
|
| /dports/multimedia/gstreamer1-plugins-libde265/gst-plugins-bad-1.16.2/ext/webrtc/ |
| H A D | webrtcdatachannel.c | 371 pad = gst_element_get_static_pad (channel->appsrc, "src"); in _close_sctp_stream()
520 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in _parse_control_packet()
744 if (gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), in gst_webrtc_data_channel_start_negotiation()
826 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_data()
886 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_string()
1086 channel->appsrc = gst_element_factory_make ("appsrc", NULL); in gst_webrtc_data_channel_constructed()
1087 gst_object_ref_sink (channel->appsrc); in gst_webrtc_data_channel_constructed()
1088 pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_constructed()
1110 GstPad *pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_finalize()
1126 g_clear_object (&channel->appsrc); in gst_webrtc_data_channel_finalize()
|
| /dports/multimedia/gstreamer1-plugins-hls/gst-plugins-bad-1.16.2/ext/webrtc/ |
| H A D | webrtcdatachannel.c | 371 pad = gst_element_get_static_pad (channel->appsrc, "src"); in _close_sctp_stream()
520 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in _parse_control_packet()
744 if (gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), in gst_webrtc_data_channel_start_negotiation()
826 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_data()
886 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_string()
1086 channel->appsrc = gst_element_factory_make ("appsrc", NULL); in gst_webrtc_data_channel_constructed()
1087 gst_object_ref_sink (channel->appsrc); in gst_webrtc_data_channel_constructed()
1088 pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_constructed()
1110 GstPad *pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_finalize()
1126 g_clear_object (&channel->appsrc); in gst_webrtc_data_channel_finalize()
|
| /dports/multimedia/gstreamer1-plugins-kate/gst-plugins-bad-1.16.2/ext/webrtc/ |
| H A D | webrtcdatachannel.c | 371 pad = gst_element_get_static_pad (channel->appsrc, "src"); in _close_sctp_stream()
520 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in _parse_control_packet()
744 if (gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), in gst_webrtc_data_channel_start_negotiation()
826 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_data()
886 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_string()
1086 channel->appsrc = gst_element_factory_make ("appsrc", NULL); in gst_webrtc_data_channel_constructed()
1087 gst_object_ref_sink (channel->appsrc); in gst_webrtc_data_channel_constructed()
1088 pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_constructed()
1110 GstPad *pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_finalize()
1126 g_clear_object (&channel->appsrc); in gst_webrtc_data_channel_finalize()
|
| /dports/audio/gstreamer1-plugins-soundtouch/gst-plugins-bad-1.16.2/ext/webrtc/ |
| H A D | webrtcdatachannel.c | 371 pad = gst_element_get_static_pad (channel->appsrc, "src"); in _close_sctp_stream()
520 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in _parse_control_packet()
744 if (gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), in gst_webrtc_data_channel_start_negotiation()
826 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_data()
886 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_string()
1086 channel->appsrc = gst_element_factory_make ("appsrc", NULL); in gst_webrtc_data_channel_constructed()
1087 gst_object_ref_sink (channel->appsrc); in gst_webrtc_data_channel_constructed()
1088 pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_constructed()
1110 GstPad *pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_finalize()
1126 g_clear_object (&channel->appsrc); in gst_webrtc_data_channel_finalize()
|
| /dports/audio/gstreamer1-plugins-modplug/gst-plugins-bad-1.16.2/ext/webrtc/ |
| H A D | webrtcdatachannel.c | 371 pad = gst_element_get_static_pad (channel->appsrc, "src"); in _close_sctp_stream()
520 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in _parse_control_packet()
744 if (gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), in gst_webrtc_data_channel_start_negotiation()
826 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_data()
886 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_string()
1086 channel->appsrc = gst_element_factory_make ("appsrc", NULL); in gst_webrtc_data_channel_constructed()
1087 gst_object_ref_sink (channel->appsrc); in gst_webrtc_data_channel_constructed()
1088 pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_constructed()
1110 GstPad *pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_finalize()
1126 g_clear_object (&channel->appsrc); in gst_webrtc_data_channel_finalize()
|
| /dports/audio/gstreamer1-plugins-sndfile/gst-plugins-bad-1.16.2/ext/webrtc/ |
| H A D | webrtcdatachannel.c | 371 pad = gst_element_get_static_pad (channel->appsrc, "src"); in _close_sctp_stream()
520 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in _parse_control_packet()
744 if (gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), in gst_webrtc_data_channel_start_negotiation()
826 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_data()
886 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_string()
1086 channel->appsrc = gst_element_factory_make ("appsrc", NULL); in gst_webrtc_data_channel_constructed()
1087 gst_object_ref_sink (channel->appsrc); in gst_webrtc_data_channel_constructed()
1088 pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_constructed()
1110 GstPad *pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_finalize()
1126 g_clear_object (&channel->appsrc); in gst_webrtc_data_channel_finalize()
|
| /dports/audio/gstreamer1-plugins-musepack/gst-plugins-bad-1.16.2/ext/webrtc/ |
| H A D | webrtcdatachannel.c | 371 pad = gst_element_get_static_pad (channel->appsrc, "src"); in _close_sctp_stream()
520 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in _parse_control_packet()
744 if (gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), in gst_webrtc_data_channel_start_negotiation()
826 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_data()
886 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_string()
1086 channel->appsrc = gst_element_factory_make ("appsrc", NULL); in gst_webrtc_data_channel_constructed()
1087 gst_object_ref_sink (channel->appsrc); in gst_webrtc_data_channel_constructed()
1088 pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_constructed()
1110 GstPad *pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_finalize()
1126 g_clear_object (&channel->appsrc); in gst_webrtc_data_channel_finalize()
|
| /dports/audio/gstreamer1-plugins-ladspa/gst-plugins-bad-1.16.2/ext/webrtc/ |
| H A D | webrtcdatachannel.c | 371 pad = gst_element_get_static_pad (channel->appsrc, "src"); in _close_sctp_stream()
520 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in _parse_control_packet()
744 if (gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), in gst_webrtc_data_channel_start_negotiation()
826 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_data()
886 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_string()
1086 channel->appsrc = gst_element_factory_make ("appsrc", NULL); in gst_webrtc_data_channel_constructed()
1087 gst_object_ref_sink (channel->appsrc); in gst_webrtc_data_channel_constructed()
1088 pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_constructed()
1110 GstPad *pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_finalize()
1126 g_clear_object (&channel->appsrc); in gst_webrtc_data_channel_finalize()
|
| /dports/audio/gstreamer1-plugins-openmpt/gst-plugins-bad-1.16.2/ext/webrtc/ |
| H A D | webrtcdatachannel.c | 371 pad = gst_element_get_static_pad (channel->appsrc, "src"); in _close_sctp_stream()
520 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in _parse_control_packet()
744 if (gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), in gst_webrtc_data_channel_start_negotiation()
826 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_data()
886 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_string()
1086 channel->appsrc = gst_element_factory_make ("appsrc", NULL); in gst_webrtc_data_channel_constructed()
1087 gst_object_ref_sink (channel->appsrc); in gst_webrtc_data_channel_constructed()
1088 pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_constructed()
1110 GstPad *pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_finalize()
1126 g_clear_object (&channel->appsrc); in gst_webrtc_data_channel_finalize()
|
| /dports/graphics/gstreamer1-plugins-zbar/gst-plugins-bad-1.16.2/ext/webrtc/ |
| H A D | webrtcdatachannel.c | 371 pad = gst_element_get_static_pad (channel->appsrc, "src"); in _close_sctp_stream()
520 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in _parse_control_packet()
744 if (gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), in gst_webrtc_data_channel_start_negotiation()
826 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_data()
886 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_string()
1086 channel->appsrc = gst_element_factory_make ("appsrc", NULL); in gst_webrtc_data_channel_constructed()
1087 gst_object_ref_sink (channel->appsrc); in gst_webrtc_data_channel_constructed()
1088 pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_constructed()
1110 GstPad *pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_finalize()
1126 g_clear_object (&channel->appsrc); in gst_webrtc_data_channel_finalize()
|
| /dports/graphics/gstreamer1-plugins-rsvg/gst-plugins-bad-1.16.2/ext/webrtc/ |
| H A D | webrtcdatachannel.c | 371 pad = gst_element_get_static_pad (channel->appsrc, "src"); in _close_sctp_stream()
520 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in _parse_control_packet()
744 if (gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), in gst_webrtc_data_channel_start_negotiation()
826 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_data()
886 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_string()
1086 channel->appsrc = gst_element_factory_make ("appsrc", NULL); in gst_webrtc_data_channel_constructed()
1087 gst_object_ref_sink (channel->appsrc); in gst_webrtc_data_channel_constructed()
1088 pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_constructed()
1110 GstPad *pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_finalize()
1126 g_clear_object (&channel->appsrc); in gst_webrtc_data_channel_finalize()
|
| /dports/graphics/gstreamer1-plugins-vulkan/gst-plugins-bad-1.16.2/ext/webrtc/ |
| H A D | webrtcdatachannel.c | 371 pad = gst_element_get_static_pad (channel->appsrc, "src"); in _close_sctp_stream()
520 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in _parse_control_packet()
744 if (gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), in gst_webrtc_data_channel_start_negotiation()
826 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_data()
886 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_string()
1086 channel->appsrc = gst_element_factory_make ("appsrc", NULL); in gst_webrtc_data_channel_constructed()
1087 gst_object_ref_sink (channel->appsrc); in gst_webrtc_data_channel_constructed()
1088 pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_constructed()
1110 GstPad *pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_finalize()
1126 g_clear_object (&channel->appsrc); in gst_webrtc_data_channel_finalize()
|
| /dports/graphics/gstreamer1-plugins-webp/gst-plugins-bad-1.16.2/ext/webrtc/ |
| H A D | webrtcdatachannel.c | 371 pad = gst_element_get_static_pad (channel->appsrc, "src"); in _close_sctp_stream()
520 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in _parse_control_packet()
744 if (gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), in gst_webrtc_data_channel_start_negotiation()
826 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_data()
886 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_string()
1086 channel->appsrc = gst_element_factory_make ("appsrc", NULL); in gst_webrtc_data_channel_constructed()
1087 gst_object_ref_sink (channel->appsrc); in gst_webrtc_data_channel_constructed()
1088 pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_constructed()
1110 GstPad *pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_finalize()
1126 g_clear_object (&channel->appsrc); in gst_webrtc_data_channel_finalize()
|
| /dports/audio/gstreamer1-plugins-webrtcdsp/gst-plugins-bad-1.16.2/ext/webrtc/ |
| H A D | webrtcdatachannel.c | 371 pad = gst_element_get_static_pad (channel->appsrc, "src"); in _close_sctp_stream()
520 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in _parse_control_packet()
744 if (gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), in gst_webrtc_data_channel_start_negotiation()
826 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_data()
886 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_string()
1086 channel->appsrc = gst_element_factory_make ("appsrc", NULL); in gst_webrtc_data_channel_constructed()
1087 gst_object_ref_sink (channel->appsrc); in gst_webrtc_data_channel_constructed()
1088 pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_constructed()
1110 GstPad *pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_finalize()
1126 g_clear_object (&channel->appsrc); in gst_webrtc_data_channel_finalize()
|
| /dports/comms/gstreamer1-plugins-spandsp/gst-plugins-bad-1.16.2/ext/webrtc/ |
| H A D | webrtcdatachannel.c | 371 pad = gst_element_get_static_pad (channel->appsrc, "src"); in _close_sctp_stream()
520 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in _parse_control_packet()
744 if (gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), in gst_webrtc_data_channel_start_negotiation()
826 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_data()
886 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_string()
1086 channel->appsrc = gst_element_factory_make ("appsrc", NULL); in gst_webrtc_data_channel_constructed()
1087 gst_object_ref_sink (channel->appsrc); in gst_webrtc_data_channel_constructed()
1088 pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_constructed()
1110 GstPad *pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_finalize()
1126 g_clear_object (&channel->appsrc); in gst_webrtc_data_channel_finalize()
|
| /dports/audio/gstreamer1-plugins-bs2b/gst-plugins-bad-1.16.2/ext/webrtc/ |
| H A D | webrtcdatachannel.c | 371 pad = gst_element_get_static_pad (channel->appsrc, "src"); in _close_sctp_stream()
520 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in _parse_control_packet()
744 if (gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), in gst_webrtc_data_channel_start_negotiation()
826 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_data()
886 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_string()
1086 channel->appsrc = gst_element_factory_make ("appsrc", NULL); in gst_webrtc_data_channel_constructed()
1087 gst_object_ref_sink (channel->appsrc); in gst_webrtc_data_channel_constructed()
1088 pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_constructed()
1110 GstPad *pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_finalize()
1126 g_clear_object (&channel->appsrc); in gst_webrtc_data_channel_finalize()
|
| /dports/audio/gstreamer1-plugins-chromaprint/gst-plugins-bad-1.16.2/ext/webrtc/ |
| H A D | webrtcdatachannel.c | 371 pad = gst_element_get_static_pad (channel->appsrc, "src"); in _close_sctp_stream()
520 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in _parse_control_packet()
744 if (gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), in gst_webrtc_data_channel_start_negotiation()
826 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_data()
886 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_string()
1086 channel->appsrc = gst_element_factory_make ("appsrc", NULL); in gst_webrtc_data_channel_constructed()
1087 gst_object_ref_sink (channel->appsrc); in gst_webrtc_data_channel_constructed()
1088 pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_constructed()
1110 GstPad *pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_finalize()
1126 g_clear_object (&channel->appsrc); in gst_webrtc_data_channel_finalize()
|
| /dports/audio/gstreamer1-plugins-gme/gst-plugins-bad-1.16.2/ext/webrtc/ |
| H A D | webrtcdatachannel.c | 371 pad = gst_element_get_static_pad (channel->appsrc, "src"); in _close_sctp_stream()
520 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in _parse_control_packet()
744 if (gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), in gst_webrtc_data_channel_start_negotiation()
826 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_data()
886 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_string()
1086 channel->appsrc = gst_element_factory_make ("appsrc", NULL); in gst_webrtc_data_channel_constructed()
1087 gst_object_ref_sink (channel->appsrc); in gst_webrtc_data_channel_constructed()
1088 pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_constructed()
1110 GstPad *pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_finalize()
1126 g_clear_object (&channel->appsrc); in gst_webrtc_data_channel_finalize()
|
| /dports/audio/gstreamer1-plugins-flite/gst-plugins-bad-1.16.2/ext/webrtc/ |
| H A D | webrtcdatachannel.c | 371 pad = gst_element_get_static_pad (channel->appsrc, "src"); in _close_sctp_stream()
520 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in _parse_control_packet()
744 if (gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), in gst_webrtc_data_channel_start_negotiation()
826 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_data()
886 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_string()
1086 channel->appsrc = gst_element_factory_make ("appsrc", NULL); in gst_webrtc_data_channel_constructed()
1087 gst_object_ref_sink (channel->appsrc); in gst_webrtc_data_channel_constructed()
1088 pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_constructed()
1110 GstPad *pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_finalize()
1126 g_clear_object (&channel->appsrc); in gst_webrtc_data_channel_finalize()
|
| /dports/audio/gstreamer1-plugins-gsm/gst-plugins-bad-1.16.2/ext/webrtc/ |
| H A D | webrtcdatachannel.c | 371 pad = gst_element_get_static_pad (channel->appsrc, "src"); in _close_sctp_stream()
520 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in _parse_control_packet()
744 if (gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), in gst_webrtc_data_channel_start_negotiation()
826 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_data()
886 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_string()
1086 channel->appsrc = gst_element_factory_make ("appsrc", NULL); in gst_webrtc_data_channel_constructed()
1087 gst_object_ref_sink (channel->appsrc); in gst_webrtc_data_channel_constructed()
1088 pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_constructed()
1110 GstPad *pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_finalize()
1126 g_clear_object (&channel->appsrc); in gst_webrtc_data_channel_finalize()
|
| /dports/audio/gstreamer1-plugins-faad/gst-plugins-bad-1.16.2/ext/webrtc/ |
| H A D | webrtcdatachannel.c | 371 pad = gst_element_get_static_pad (channel->appsrc, "src"); in _close_sctp_stream()
520 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in _parse_control_packet()
744 if (gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), in gst_webrtc_data_channel_start_negotiation()
826 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_data()
886 ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer); in gst_webrtc_data_channel_send_string()
1086 channel->appsrc = gst_element_factory_make ("appsrc", NULL); in gst_webrtc_data_channel_constructed()
1087 gst_object_ref_sink (channel->appsrc); in gst_webrtc_data_channel_constructed()
1088 pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_constructed()
1110 GstPad *pad = gst_element_get_static_pad (channel->appsrc, "src"); in gst_webrtc_data_channel_finalize()
1126 g_clear_object (&channel->appsrc); in gst_webrtc_data_channel_finalize()
|