mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-10-22 10:23:47 +00:00
9caee48ed4
Original commit message from CVS: * gst-libs/gst/rtp/gstbasertpaudiopayload.c:(gst_base_rtp_audio_payload_handle_frame_based_buffer): Use gst_gdouble_to_guint64 for conversions. * win32/common/config.h.in: Add a define for GST_INSTALL_PLUGINS_HELPER * win32/common/libgstaudio.def: * win32/common/libgstcdda.def: * win32/common/libgstnetbuffer.def: * win32/common/libgstrtp.def: * win32/common/libgutils.def: Add new exported functions. * win32/vs6/gst_plugins_base.dsw: * win32/vs6/libgstdecodebin.dsp: * win32/vs6/libgstnetbuffer.dsp: * win32/vs6/libgstplaybin.dsp: * win32/vs6/libgstrtp.dsp: * win32/vs6/libgstvorbis.dsp: * win32/vs6/libgstcdda.dsp: * win32/vs6/libgstgdp.dsp: * win32/vs6/libgstutils.dsp: Update and add new project files.
29 lines
899 B
Modula-2
29 lines
899 B
Modula-2
EXPORTS
|
|
gst_basertppayload_is_filled
|
|
gst_basertppayload_get_type
|
|
gst_basertppayload_push
|
|
gst_basertppayload_set_options
|
|
gst_basertppayload_set_outcaps
|
|
gst_base_rtp_audio_payload_get_type
|
|
gst_base_rtp_audio_payload_set_frame_based
|
|
gst_base_rtp_audio_payload_set_frame_options
|
|
gst_base_rtp_depayload_get_type
|
|
gst_base_rtp_depayload_push
|
|
gst_base_rtp_depayload_push_ts
|
|
gst_rtp_buffer_calc_header_len
|
|
gst_rtp_buffer_calc_packet_len
|
|
gst_rtp_buffer_calc_payload_len
|
|
gst_rtp_buffer_get_marker
|
|
gst_rtp_buffer_get_payload
|
|
gst_rtp_buffer_get_payload_buffer
|
|
gst_rtp_buffer_get_payload_len
|
|
gst_rtp_buffer_get_payload_subbuffer
|
|
gst_rtp_buffer_get_payload_type
|
|
gst_rtp_buffer_get_seq
|
|
gst_rtp_buffer_get_timestamp
|
|
gst_rtp_buffer_new_allocate
|
|
gst_rtp_buffer_new_allocate_len
|
|
gst_rtp_buffer_new_take_data
|
|
gst_rtp_buffer_set_marker
|
|
gst_rtp_buffer_set_payload_type
|
|
gst_rtp_buffer_validate
|