win32: update for API changes

diff --git a/win32/common/libgstaudio.def b/win32/common/libgstaudio.def
index b660865b944b8e3079847598502191243a678e6d..991d235590268efaa9431aaf02758ded269b41ca 100644 (file)
--- a/win32/common/libgstaudio.def
+++ b/win32/common/libgstaudio.def
@@ -1,6 +1,29 @@
 EXPORTS
        _gst_audio_decoder_error
+       gst_audio_base_sink_create_ringbuffer
+       gst_audio_base_sink_get_alignment_threshold
+       gst_audio_base_sink_get_discont_wait
+       gst_audio_base_sink_get_drift_tolerance
+       gst_audio_base_sink_get_provide_clock
+       gst_audio_base_sink_get_slave_method
+       gst_audio_base_sink_get_type
+       gst_audio_base_sink_set_alignment_threshold
+       gst_audio_base_sink_set_discont_wait
+       gst_audio_base_sink_set_drift_tolerance
+       gst_audio_base_sink_set_provide_clock
+       gst_audio_base_sink_set_slave_method
+       gst_audio_base_sink_slave_method_get_type
+       gst_audio_base_src_create_ringbuffer
+       gst_audio_base_src_get_provide_clock
+       gst_audio_base_src_get_slave_method
+       gst_audio_base_src_get_type
+       gst_audio_base_src_set_provide_clock
+       gst_audio_base_src_set_slave_method
+       gst_audio_base_src_slave_method_get_type
        gst_audio_buffer_clip
+       gst_audio_cd_src_add_track
+       gst_audio_cd_src_get_type
+       gst_audio_cd_src_mode_get_type
        gst_audio_channel_position_get_type
        gst_audio_check_channel_positions
        gst_audio_clock_adjust
@@ -8,7 +31,6 @@ EXPORTS
        gst_audio_clock_get_type
        gst_audio_clock_invalidate
        gst_audio_clock_new
-       gst_audio_clock_new_full
        gst_audio_clock_reset
        gst_audio_decoder_finish_frame
        gst_audio_decoder_get_audio_info
@@ -61,69 +83,54 @@ EXPORTS
        gst_audio_format_fill_silence
        gst_audio_format_from_string
        gst_audio_format_get_info
+       gst_audio_format_info_get_type
        gst_audio_format_to_string
        gst_audio_get_channel_positions
        gst_audio_iec61937_frame_size
        gst_audio_iec61937_payload
        gst_audio_info_convert
+       gst_audio_info_copy
+       gst_audio_info_free
        gst_audio_info_from_caps
+       gst_audio_info_get_type
        gst_audio_info_init
+       gst_audio_info_new
        gst_audio_info_set_format
        gst_audio_info_to_caps
+       gst_audio_ring_buffer_acquire
+       gst_audio_ring_buffer_activate
+       gst_audio_ring_buffer_advance
+       gst_audio_ring_buffer_clear
+       gst_audio_ring_buffer_clear_all
+       gst_audio_ring_buffer_close_device
+       gst_audio_ring_buffer_commit
+       gst_audio_ring_buffer_commit_full
+       gst_audio_ring_buffer_convert
+       gst_audio_ring_buffer_debug_spec_buff
+       gst_audio_ring_buffer_debug_spec_caps
+       gst_audio_ring_buffer_delay
+       gst_audio_ring_buffer_device_is_open
+       gst_audio_ring_buffer_get_type
+       gst_audio_ring_buffer_is_acquired
+       gst_audio_ring_buffer_is_active
+       gst_audio_ring_buffer_may_start
+       gst_audio_ring_buffer_open_device
+       gst_audio_ring_buffer_parse_caps
+       gst_audio_ring_buffer_pause
+       gst_audio_ring_buffer_prepare_read
+       gst_audio_ring_buffer_read
+       gst_audio_ring_buffer_release
+       gst_audio_ring_buffer_samples_done
+       gst_audio_ring_buffer_seg_state_get_type
+       gst_audio_ring_buffer_set_callback
+       gst_audio_ring_buffer_set_flushing
+       gst_audio_ring_buffer_set_sample
+       gst_audio_ring_buffer_start
+       gst_audio_ring_buffer_state_get_type
+       gst_audio_ring_buffer_stop
        gst_audio_set_caps_channel_positions_list
        gst_audio_set_channel_positions
        gst_audio_set_structure_channel_positions_list
        gst_audio_sink_get_type
        gst_audio_src_get_type
-       gst_base_audio_sink_create_ringbuffer
-       gst_base_audio_sink_get_alignment_threshold
-       gst_base_audio_sink_get_discont_wait
-       gst_base_audio_sink_get_drift_tolerance
-       gst_base_audio_sink_get_provide_clock
-       gst_base_audio_sink_get_slave_method
-       gst_base_audio_sink_get_type
-       gst_base_audio_sink_set_alignment_threshold
-       gst_base_audio_sink_set_discont_wait
-       gst_base_audio_sink_set_drift_tolerance
-       gst_base_audio_sink_set_provide_clock
-       gst_base_audio_sink_set_slave_method
-       gst_base_audio_sink_slave_method_get_type
-       gst_base_audio_src_create_ringbuffer
-       gst_base_audio_src_get_provide_clock
-       gst_base_audio_src_get_slave_method
-       gst_base_audio_src_get_type
-       gst_base_audio_src_set_provide_clock
-       gst_base_audio_src_set_slave_method
-       gst_base_audio_src_slave_method_get_type
        gst_buffer_format_type_get_type
-       gst_ring_buffer_acquire
-       gst_ring_buffer_activate
-       gst_ring_buffer_advance
-       gst_ring_buffer_clear
-       gst_ring_buffer_clear_all
-       gst_ring_buffer_close_device
-       gst_ring_buffer_commit
-       gst_ring_buffer_commit_full
-       gst_ring_buffer_convert
-       gst_ring_buffer_debug_spec_buff
-       gst_ring_buffer_debug_spec_caps
-       gst_ring_buffer_delay
-       gst_ring_buffer_device_is_open
-       gst_ring_buffer_get_type
-       gst_ring_buffer_is_acquired
-       gst_ring_buffer_is_active
-       gst_ring_buffer_may_start
-       gst_ring_buffer_open_device
-       gst_ring_buffer_parse_caps
-       gst_ring_buffer_pause
-       gst_ring_buffer_prepare_read
-       gst_ring_buffer_read
-       gst_ring_buffer_release
-       gst_ring_buffer_samples_done
-       gst_ring_buffer_seg_state_get_type
-       gst_ring_buffer_set_callback
-       gst_ring_buffer_set_flushing
-       gst_ring_buffer_set_sample
-       gst_ring_buffer_start
-       gst_ring_buffer_state_get_type
-       gst_ring_buffer_stop

diff --git a/win32/common/libgstinterfaces.def b/win32/common/libgstinterfaces.def
index 6b974b541a39b29133abe28af1526d8e2900c5bf..fbbf7dae468a6662e4761d31627e01fdb2c4be0b 100644 (file)
--- a/win32/common/libgstinterfaces.def
+++ b/win32/common/libgstinterfaces.def
@@ -7,6 +7,11 @@ EXPORTS
        gst_color_balance_set_value
        gst_color_balance_type_get_type
        gst_color_balance_value_changed
+       gst_interfaces_marshal_VOID__OBJECT_BOOLEAN
+       gst_interfaces_marshal_VOID__OBJECT_INT
+       gst_interfaces_marshal_VOID__OBJECT_POINTER
+       gst_interfaces_marshal_VOID__OBJECT_STRING
+       gst_interfaces_marshal_VOID__OBJECT_ULONG
        gst_is_video_overlay_prepare_window_handle_message
        gst_mixer_flags_get_type
        gst_mixer_get_mixer_flags

diff --git a/win32/common/libgstrtp.def b/win32/common/libgstrtp.def
index 54d855b95fc306c410cebfe09bda3c977390b964..c1bda6a8dc6ab5524277a3eaa52d64a59b2f165c 100644 (file)
--- a/win32/common/libgstrtp.def
+++ b/win32/common/libgstrtp.def
@@ -1,23 +1,4 @@
 EXPORTS
-       gst_base_rtp_audio_payload_flush
-       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_audio_payload_set_samplebits_options
-       gst_base_rtp_depayload_get_type
-       gst_base_rtp_depayload_push
-       gst_base_rtp_depayload_push_list
-       gst_base_rtp_depayload_push_ts
-       gst_basertppayload_get_type
-       gst_basertppayload_is_filled
-       gst_basertppayload_push
-       gst_basertppayload_push_list
-       gst_basertppayload_set_options
-       gst_basertppayload_set_outcaps
        gst_rtcp_buffer_add_packet
        gst_rtcp_buffer_get_first_packet
        gst_rtcp_buffer_get_packet_count
@@ -72,6 +53,24 @@ EXPORTS
        gst_rtcp_sdes_name_to_type
        gst_rtcp_sdes_type_to_name
        gst_rtcp_unix_to_ntp
+       gst_rtp_base_audio_payload_flush
+       gst_rtp_base_audio_payload_get_adapter
+       gst_rtp_base_audio_payload_get_type
+       gst_rtp_base_audio_payload_push
+       gst_rtp_base_audio_payload_set_frame_based
+       gst_rtp_base_audio_payload_set_frame_options
+       gst_rtp_base_audio_payload_set_sample_based
+       gst_rtp_base_audio_payload_set_sample_options
+       gst_rtp_base_audio_payload_set_samplebits_options
+       gst_rtp_base_depayload_get_type
+       gst_rtp_base_depayload_push
+       gst_rtp_base_depayload_push_list
+       gst_rtp_base_payload_get_type
+       gst_rtp_base_payload_is_filled
+       gst_rtp_base_payload_push
+       gst_rtp_base_payload_push_list
+       gst_rtp_base_payload_set_options
+       gst_rtp_base_payload_set_outcaps
        gst_rtp_buffer_add_extension_onebyte_header
        gst_rtp_buffer_add_extension_twobytes_header
        gst_rtp_buffer_allocate_data

diff --git a/win32/common/libgstrtsp.def b/win32/common/libgstrtsp.def
index 5327c5b8cf5c3e5fd7949913aee860f9b7047d8c..3807869cd4f73d83719169b9b1f227410875b06a 100644 (file)
--- a/win32/common/libgstrtsp.def
+++ b/win32/common/libgstrtsp.def
@@ -1,7 +1,5 @@
 EXPORTS
        gst_rtsp_auth_method_get_type
-       gst_rtsp_base64_decode_ip
-       gst_rtsp_base64_encode
        gst_rtsp_connection_accept
        gst_rtsp_connection_clear_auth_params
        gst_rtsp_connection_close
@@ -104,8 +102,6 @@ EXPORTS
        gst_rtsp_version_get_type
        gst_rtsp_watch_attach
        gst_rtsp_watch_new
-       gst_rtsp_watch_queue_data
-       gst_rtsp_watch_queue_message
        gst_rtsp_watch_reset
        gst_rtsp_watch_send_message
        gst_rtsp_watch_unref