mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-10 03:19:40 +00:00
ec7afb6f84
Original commit message from CVS: * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_get_type): * ext/alsa/gstalsasink.c: (set_hwparams): * ext/alsa/gstalsasrc.c: (set_hwparams): * ext/gio/gstgio.c: (gst_gio_uri_handler_get_uri): * ext/ogg/gstoggmux.h: * ext/ogg/gstogmparse.c: * gst-libs/gst/audio/audio.c: * gst-libs/gst/fft/kiss_fft_f64.c: (kiss_fft_f64_alloc): * gst-libs/gst/pbutils/missing-plugins.c: (gst_missing_uri_sink_message_new), (gst_missing_element_message_new), (gst_missing_decoder_message_new), (gst_missing_encoder_message_new): * gst-libs/gst/rtp/gstbasertppayload.c: * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_bye_get_reason): * gst/audioconvert/gstaudioconvert.c: * gst/audioresample/gstaudioresample.c: * gst/ffmpegcolorspace/imgconvert.c: * gst/playback/test.c: (gen_video_element), (gen_audio_element): * gst/typefind/gsttypefindfunctions.c: * gst/videoscale/vs_4tap.c: * gst/videoscale/vs_4tap.h: * sys/v4l/gstv4lelement.c: * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get_any_caps): * sys/v4l/v4l_calls.c: * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_capture_init), (gst_v4lsrc_try_capture): * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls), (gst_ximagesink_ximage_new): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new): * tests/check/elements/audioconvert.c: * tests/check/elements/audioresample.c: (fail_unless_perfect_stream): * tests/check/elements/audiotestsrc.c: (setup_audiotestsrc): * tests/check/elements/decodebin.c: * tests/check/elements/gdpdepay.c: (setup_gdpdepay), (setup_gdpdepay_streamheader): * tests/check/elements/gdppay.c: (setup_gdppay), (GST_START_TEST), (setup_gdppay_streamheader): * tests/check/elements/gnomevfssink.c: (setup_gnomevfssink): * tests/check/elements/multifdsink.c: (setup_multifdsink): * tests/check/elements/textoverlay.c: * tests/check/elements/videorate.c: (setup_videorate): * tests/check/elements/videotestsrc.c: (setup_videotestsrc): * tests/check/elements/volume.c: (setup_volume): * tests/check/elements/vorbisdec.c: (setup_vorbisdec): * tests/check/elements/vorbistag.c: * tests/check/generic/clock-selection.c: * tests/check/generic/states.c: (setup), (teardown): * tests/check/libs/cddabasesrc.c: * tests/check/libs/video.c: * tests/check/pipelines/gio.c: * tests/check/pipelines/oggmux.c: * tests/check/pipelines/simple-launch-lines.c: (simple_launch_lines_suite): * tests/check/pipelines/streamheader.c: * tests/check/pipelines/theoraenc.c: * tests/check/pipelines/vorbisdec.c: * tests/check/pipelines/vorbisenc.c: * tests/examples/seek/scrubby.c: * tests/examples/seek/seek.c: (query_positions_elems), (query_positions_pads): * tests/icles/stress-xoverlay.c: (myclock): Correct all relevant warnings found by the sparse semantic code analyzer. This include marking several symbols static, using NULL instead of 0 for pointers and using "foo (void)" instead of "foo ()" for declarations. * win32/common/libgstrtp.def: Add gst_rtp_buffer_set_extension_data to the symbol definition file.
102 lines
3 KiB
Modula-2
102 lines
3 KiB
Modula-2
EXPORTS
|
|
gst_base_rtp_audio_payload_get_adapter
|
|
gst_base_rtp_audio_payload_get_type
|
|
gst_base_rtp_audio_payload_push
|
|
gst_base_rtp_audio_payload_set_frame_based
|
|
gst_base_rtp_audio_payload_set_frame_options
|
|
gst_base_rtp_audio_payload_set_sample_based
|
|
gst_base_rtp_audio_payload_set_sample_options
|
|
gst_base_rtp_depayload_get_type
|
|
gst_base_rtp_depayload_push
|
|
gst_base_rtp_depayload_push_ts
|
|
gst_basertppayload_get_type
|
|
gst_basertppayload_is_filled
|
|
gst_basertppayload_push
|
|
gst_basertppayload_set_options
|
|
gst_basertppayload_set_outcaps
|
|
gst_rtcp_buffer_add_packet
|
|
gst_rtcp_buffer_end
|
|
gst_rtcp_buffer_get_first_packet
|
|
gst_rtcp_buffer_get_packet_count
|
|
gst_rtcp_buffer_new
|
|
gst_rtcp_buffer_new_copy_data
|
|
gst_rtcp_buffer_new_take_data
|
|
gst_rtcp_buffer_validate
|
|
gst_rtcp_buffer_validate_data
|
|
gst_rtcp_ntp_to_unix
|
|
gst_rtcp_packet_add_rb
|
|
gst_rtcp_packet_bye_add_ssrc
|
|
gst_rtcp_packet_bye_add_ssrcs
|
|
gst_rtcp_packet_bye_get_nth_ssrc
|
|
gst_rtcp_packet_bye_get_reason
|
|
gst_rtcp_packet_bye_get_reason_len
|
|
gst_rtcp_packet_bye_get_ssrc_count
|
|
gst_rtcp_packet_bye_set_reason
|
|
gst_rtcp_packet_get_count
|
|
gst_rtcp_packet_get_length
|
|
gst_rtcp_packet_get_padding
|
|
gst_rtcp_packet_get_rb
|
|
gst_rtcp_packet_get_rb_count
|
|
gst_rtcp_packet_get_type
|
|
gst_rtcp_packet_move_to_next
|
|
gst_rtcp_packet_remove
|
|
gst_rtcp_packet_rr_get_ssrc
|
|
gst_rtcp_packet_rr_set_ssrc
|
|
gst_rtcp_packet_sdes_add_entry
|
|
gst_rtcp_packet_sdes_add_item
|
|
gst_rtcp_packet_sdes_copy_entry
|
|
gst_rtcp_packet_sdes_first_entry
|
|
gst_rtcp_packet_sdes_first_item
|
|
gst_rtcp_packet_sdes_get_entry
|
|
gst_rtcp_packet_sdes_get_item_count
|
|
gst_rtcp_packet_sdes_get_ssrc
|
|
gst_rtcp_packet_sdes_next_entry
|
|
gst_rtcp_packet_sdes_next_item
|
|
gst_rtcp_packet_set_rb
|
|
gst_rtcp_packet_sr_get_sender_info
|
|
gst_rtcp_packet_sr_set_sender_info
|
|
gst_rtcp_unix_to_ntp
|
|
gst_rtp_buffer_allocate_data
|
|
gst_rtp_buffer_calc_header_len
|
|
gst_rtp_buffer_calc_packet_len
|
|
gst_rtp_buffer_calc_payload_len
|
|
gst_rtp_buffer_compare_seqnum
|
|
gst_rtp_buffer_default_clock_rate
|
|
gst_rtp_buffer_ext_timestamp
|
|
gst_rtp_buffer_get_csrc
|
|
gst_rtp_buffer_get_csrc_count
|
|
gst_rtp_buffer_get_extension
|
|
gst_rtp_buffer_get_extension_data
|
|
gst_rtp_buffer_get_header_len
|
|
gst_rtp_buffer_get_marker
|
|
gst_rtp_buffer_get_packet_len
|
|
gst_rtp_buffer_get_padding
|
|
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_ssrc
|
|
gst_rtp_buffer_get_timestamp
|
|
gst_rtp_buffer_get_version
|
|
gst_rtp_buffer_new_allocate
|
|
gst_rtp_buffer_new_allocate_len
|
|
gst_rtp_buffer_new_copy_data
|
|
gst_rtp_buffer_new_take_data
|
|
gst_rtp_buffer_pad_to
|
|
gst_rtp_buffer_set_csrc
|
|
gst_rtp_buffer_set_extension
|
|
gst_rtp_buffer_set_extension_data
|
|
gst_rtp_buffer_set_marker
|
|
gst_rtp_buffer_set_packet_len
|
|
gst_rtp_buffer_set_padding
|
|
gst_rtp_buffer_set_payload_type
|
|
gst_rtp_buffer_set_seq
|
|
gst_rtp_buffer_set_ssrc
|
|
gst_rtp_buffer_set_timestamp
|
|
gst_rtp_buffer_set_version
|
|
gst_rtp_buffer_validate
|
|
gst_rtp_buffer_validate_data
|
|
gst_rtp_payload_info_for_name
|
|
gst_rtp_payload_info_for_pt
|