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win32: update for API changes
This commit is contained in:
Tim-Philipp Müller
parent
4b0dce5148
commit
dbb48b1de0
4 changed files with 82 additions and 75 deletions
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@ -1,6 +1,29 @@
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EXPORTS
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_gst_audio_decoder_error
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gst_audio_base_sink_create_ringbuffer
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gst_audio_base_sink_get_alignment_threshold
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gst_audio_base_sink_get_discont_wait
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gst_audio_base_sink_get_drift_tolerance
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gst_audio_base_sink_get_provide_clock
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gst_audio_base_sink_get_slave_method
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gst_audio_base_sink_get_type
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gst_audio_base_sink_set_alignment_threshold
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gst_audio_base_sink_set_discont_wait
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gst_audio_base_sink_set_drift_tolerance
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gst_audio_base_sink_set_provide_clock
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gst_audio_base_sink_set_slave_method
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gst_audio_base_sink_slave_method_get_type
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gst_audio_base_src_create_ringbuffer
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gst_audio_base_src_get_provide_clock
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gst_audio_base_src_get_slave_method
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gst_audio_base_src_get_type
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gst_audio_base_src_set_provide_clock
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gst_audio_base_src_set_slave_method
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gst_audio_base_src_slave_method_get_type
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gst_audio_buffer_clip
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gst_audio_cd_src_add_track
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gst_audio_cd_src_get_type
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gst_audio_cd_src_mode_get_type
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gst_audio_channel_position_get_type
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gst_audio_check_channel_positions
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gst_audio_clock_adjust
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@ -8,7 +31,6 @@ EXPORTS
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gst_audio_clock_get_type
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gst_audio_clock_invalidate
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gst_audio_clock_new
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gst_audio_clock_new_full
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gst_audio_clock_reset
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gst_audio_decoder_finish_frame
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gst_audio_decoder_get_audio_info
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@ -61,69 +83,54 @@ EXPORTS
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gst_audio_format_fill_silence
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gst_audio_format_from_string
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gst_audio_format_get_info
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gst_audio_format_info_get_type
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gst_audio_format_to_string
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gst_audio_get_channel_positions
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gst_audio_iec61937_frame_size
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gst_audio_iec61937_payload
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gst_audio_info_convert
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gst_audio_info_copy
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gst_audio_info_free
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gst_audio_info_from_caps
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gst_audio_info_get_type
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gst_audio_info_init
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gst_audio_info_new
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gst_audio_info_set_format
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gst_audio_info_to_caps
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gst_audio_ring_buffer_acquire
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gst_audio_ring_buffer_activate
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gst_audio_ring_buffer_advance
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gst_audio_ring_buffer_clear
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gst_audio_ring_buffer_clear_all
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gst_audio_ring_buffer_close_device
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gst_audio_ring_buffer_commit
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gst_audio_ring_buffer_commit_full
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gst_audio_ring_buffer_convert
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gst_audio_ring_buffer_debug_spec_buff
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gst_audio_ring_buffer_debug_spec_caps
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gst_audio_ring_buffer_delay
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gst_audio_ring_buffer_device_is_open
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gst_audio_ring_buffer_get_type
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gst_audio_ring_buffer_is_acquired
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gst_audio_ring_buffer_is_active
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gst_audio_ring_buffer_may_start
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gst_audio_ring_buffer_open_device
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gst_audio_ring_buffer_parse_caps
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gst_audio_ring_buffer_pause
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gst_audio_ring_buffer_prepare_read
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gst_audio_ring_buffer_read
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gst_audio_ring_buffer_release
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gst_audio_ring_buffer_samples_done
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gst_audio_ring_buffer_seg_state_get_type
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gst_audio_ring_buffer_set_callback
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gst_audio_ring_buffer_set_flushing
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gst_audio_ring_buffer_set_sample
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gst_audio_ring_buffer_start
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gst_audio_ring_buffer_state_get_type
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gst_audio_ring_buffer_stop
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gst_audio_set_caps_channel_positions_list
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gst_audio_set_channel_positions
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gst_audio_set_structure_channel_positions_list
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gst_audio_sink_get_type
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gst_audio_src_get_type
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gst_base_audio_sink_create_ringbuffer
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gst_base_audio_sink_get_alignment_threshold
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gst_base_audio_sink_get_discont_wait
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gst_base_audio_sink_get_drift_tolerance
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gst_base_audio_sink_get_provide_clock
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gst_base_audio_sink_get_slave_method
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gst_base_audio_sink_get_type
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gst_base_audio_sink_set_alignment_threshold
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gst_base_audio_sink_set_discont_wait
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gst_base_audio_sink_set_drift_tolerance
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gst_base_audio_sink_set_provide_clock
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gst_base_audio_sink_set_slave_method
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gst_base_audio_sink_slave_method_get_type
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gst_base_audio_src_create_ringbuffer
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gst_base_audio_src_get_provide_clock
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gst_base_audio_src_get_slave_method
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gst_base_audio_src_get_type
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gst_base_audio_src_set_provide_clock
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gst_base_audio_src_set_slave_method
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gst_base_audio_src_slave_method_get_type
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gst_buffer_format_type_get_type
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gst_ring_buffer_acquire
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gst_ring_buffer_activate
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gst_ring_buffer_advance
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gst_ring_buffer_clear
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gst_ring_buffer_clear_all
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gst_ring_buffer_close_device
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gst_ring_buffer_commit
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gst_ring_buffer_commit_full
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gst_ring_buffer_convert
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gst_ring_buffer_debug_spec_buff
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gst_ring_buffer_debug_spec_caps
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gst_ring_buffer_delay
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gst_ring_buffer_device_is_open
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gst_ring_buffer_get_type
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gst_ring_buffer_is_acquired
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gst_ring_buffer_is_active
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gst_ring_buffer_may_start
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gst_ring_buffer_open_device
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gst_ring_buffer_parse_caps
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gst_ring_buffer_pause
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gst_ring_buffer_prepare_read
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gst_ring_buffer_read
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gst_ring_buffer_release
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gst_ring_buffer_samples_done
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gst_ring_buffer_seg_state_get_type
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gst_ring_buffer_set_callback
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gst_ring_buffer_set_flushing
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gst_ring_buffer_set_sample
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gst_ring_buffer_start
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gst_ring_buffer_state_get_type
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gst_ring_buffer_stop
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@ -7,6 +7,11 @@ EXPORTS
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gst_color_balance_set_value
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gst_color_balance_type_get_type
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gst_color_balance_value_changed
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gst_interfaces_marshal_VOID__OBJECT_BOOLEAN
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gst_interfaces_marshal_VOID__OBJECT_INT
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gst_interfaces_marshal_VOID__OBJECT_POINTER
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gst_interfaces_marshal_VOID__OBJECT_STRING
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gst_interfaces_marshal_VOID__OBJECT_ULONG
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gst_is_video_overlay_prepare_window_handle_message
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gst_mixer_flags_get_type
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gst_mixer_get_mixer_flags
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@ -1,23 +1,4 @@
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EXPORTS
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gst_base_rtp_audio_payload_flush
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gst_base_rtp_audio_payload_get_adapter
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gst_base_rtp_audio_payload_get_type
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gst_base_rtp_audio_payload_push
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gst_base_rtp_audio_payload_set_frame_based
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gst_base_rtp_audio_payload_set_frame_options
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gst_base_rtp_audio_payload_set_sample_based
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gst_base_rtp_audio_payload_set_sample_options
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gst_base_rtp_audio_payload_set_samplebits_options
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gst_base_rtp_depayload_get_type
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gst_base_rtp_depayload_push
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gst_base_rtp_depayload_push_list
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gst_base_rtp_depayload_push_ts
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gst_basertppayload_get_type
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gst_basertppayload_is_filled
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gst_basertppayload_push
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gst_basertppayload_push_list
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gst_basertppayload_set_options
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gst_basertppayload_set_outcaps
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gst_rtcp_buffer_add_packet
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gst_rtcp_buffer_get_first_packet
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gst_rtcp_buffer_get_packet_count
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@ -72,6 +53,24 @@ EXPORTS
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gst_rtcp_sdes_name_to_type
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gst_rtcp_sdes_type_to_name
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gst_rtcp_unix_to_ntp
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gst_rtp_base_audio_payload_flush
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gst_rtp_base_audio_payload_get_adapter
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gst_rtp_base_audio_payload_get_type
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gst_rtp_base_audio_payload_push
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gst_rtp_base_audio_payload_set_frame_based
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gst_rtp_base_audio_payload_set_frame_options
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gst_rtp_base_audio_payload_set_sample_based
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gst_rtp_base_audio_payload_set_sample_options
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gst_rtp_base_audio_payload_set_samplebits_options
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gst_rtp_base_depayload_get_type
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gst_rtp_base_depayload_push
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gst_rtp_base_depayload_push_list
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gst_rtp_base_payload_get_type
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gst_rtp_base_payload_is_filled
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gst_rtp_base_payload_push
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gst_rtp_base_payload_push_list
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gst_rtp_base_payload_set_options
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gst_rtp_base_payload_set_outcaps
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gst_rtp_buffer_add_extension_onebyte_header
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gst_rtp_buffer_add_extension_twobytes_header
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gst_rtp_buffer_allocate_data
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@ -1,7 +1,5 @@
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EXPORTS
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gst_rtsp_auth_method_get_type
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gst_rtsp_base64_decode_ip
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gst_rtsp_base64_encode
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gst_rtsp_connection_accept
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gst_rtsp_connection_clear_auth_params
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gst_rtsp_connection_close
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@ -104,8 +102,6 @@ EXPORTS
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gst_rtsp_version_get_type
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gst_rtsp_watch_attach
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gst_rtsp_watch_new
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gst_rtsp_watch_queue_data
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gst_rtsp_watch_queue_message
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gst_rtsp_watch_reset
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gst_rtsp_watch_send_message
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gst_rtsp_watch_unref
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