gstreamer 3.1.6-x64-mingw32 → 3.1.7-x64-mingw32

data/vendor/local/include/gstreamer-1.0/gst/audio/audio-enumtypes.h

@@ -57,6 +57,20 @@ GType gst_audio_ring_buffer_state_get_type (void);
 #define GST_TYPE_AUDIO_RING_BUFFER_STATE (gst_audio_ring_buffer_state_get_type())
 GType gst_audio_ring_buffer_format_type_get_type (void);
 #define GST_TYPE_AUDIO_RING_BUFFER_FORMAT_TYPE (gst_audio_ring_buffer_format_type_get_type())
+
+/* enumerations from "gstaudiobasesrc.h" */
+GType gst_audio_base_src_slave_method_get_type (void);
+#define GST_TYPE_AUDIO_BASE_SRC_SLAVE_METHOD (gst_audio_base_src_slave_method_get_type())
+
+/* enumerations from "gstaudiocdsrc.h" */
+GType gst_audio_cd_src_mode_get_type (void);
+#define GST_TYPE_AUDIO_CD_SRC_MODE (gst_audio_cd_src_mode_get_type())
+
+/* enumerations from "gstaudiobasesink.h" */
+GType gst_audio_base_sink_slave_method_get_type (void);
+#define GST_TYPE_AUDIO_BASE_SINK_SLAVE_METHOD (gst_audio_base_sink_slave_method_get_type())
+GType gst_audio_base_sink_discont_reason_get_type (void);
+#define GST_TYPE_AUDIO_BASE_SINK_DISCONT_REASON (gst_audio_base_sink_discont_reason_get_type())
 G_END_DECLS
 
 #endif /* __GST_AUDIO_ENUM_TYPES_H__ */

data/vendor/local/include/gstreamer-1.0/gst/audio/audio-resampler.h

@@ -28,20 +28,20 @@ G_BEGIN_DECLS
 typedef struct _GstAudioResampler GstAudioResampler;
 
 /**
- * GST_AUDIO_RESAMPLER_OPT_CUTOFF
+ * GST_AUDIO_RESAMPLER_OPT_CUTOFF:
  *
  * G_TYPE_DOUBLE, Cutoff parameter for the filter. 0.940 is the default.
  */
 #define GST_AUDIO_RESAMPLER_OPT_CUTOFF      "GstAudioResampler.cutoff"
 /**
- * GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUTATION
+ * GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUATION:
  *
- * G_TYPE_DOUBLE, stopband attenuation in debibels. The attenutation
+ * G_TYPE_DOUBLE, stopband attenuation in decibels. The attenuation
  * after the stopband for the kaiser window. 85 dB is the default.
  */
 #define GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUATION "GstAudioResampler.stop-attenutation"
 /**
- * GST_AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH
+ * GST_AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH:
  *
  * G_TYPE_DOUBLE, transition bandwidth. The width of the
  * transition band for the kaiser window. 0.087 is the default.
@@ -137,7 +137,7 @@ typedef enum {
  */
 #define GST_AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION "GstAudioResampler.filter-interpolation"
 /**
- * GST_AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE
+ * GST_AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE:
  *
  * G_TYPE_UINT, oversampling to use when interpolating filters
  * 8 is the default.

data/vendor/local/include/gstreamer-1.0/gst/audio/gstaudiobasesink.h

@@ -99,8 +99,6 @@ typedef enum
   GST_AUDIO_BASE_SINK_SLAVE_CUSTOM
 } GstAudioBaseSinkSlaveMethod;
 
-#define GST_TYPE_AUDIO_BASE_SINK_SLAVE_METHOD (gst_audio_base_sink_slave_method_get_type ())
-
 typedef struct _GstAudioBaseSink GstAudioBaseSink;
 typedef struct _GstAudioBaseSinkClass GstAudioBaseSinkClass;
 typedef struct _GstAudioBaseSinkPrivate GstAudioBaseSinkPrivate;
@@ -223,7 +221,6 @@ struct _GstAudioBaseSinkClass {
 };
 
 GType gst_audio_base_sink_get_type(void);
-GType gst_audio_base_sink_slave_method_get_type (void);
 
 GstAudioRingBuffer *
            gst_audio_base_sink_create_ringbuffer       (GstAudioBaseSink *sink);

data/vendor/local/include/gstreamer-1.0/gst/audio/gstaudiobasesrc.h

@@ -62,10 +62,12 @@ typedef struct _GstAudioBaseSrc GstAudioBaseSrc;
 typedef struct _GstAudioBaseSrcClass GstAudioBaseSrcClass;
 typedef struct _GstAudioBaseSrcPrivate GstAudioBaseSrcPrivate;
 
+/* FIXME 2.0: Should be "retimestamp" not "re-timestamp" */
+
 /**
  * GstAudioBaseSrcSlaveMethod:
  * @GST_AUDIO_BASE_SRC_SLAVE_RESAMPLE: Resample to match the master clock.
- * @GST_AUDIO_BASE_SRC_SLAVE_RETIMESTAMP: Retimestamp output buffers with master
+ * @GST_AUDIO_BASE_SRC_SLAVE_RE_TIMESTAMP: Retimestamp output buffers with master
  * clock time.
  * @GST_AUDIO_BASE_SRC_SLAVE_SKEW: Adjust capture pointer when master clock
  * drifts too much.
@@ -77,12 +79,12 @@ typedef struct _GstAudioBaseSrcPrivate GstAudioBaseSrcPrivate;
 typedef enum
 {
   GST_AUDIO_BASE_SRC_SLAVE_RESAMPLE,
-  GST_AUDIO_BASE_SRC_SLAVE_RETIMESTAMP,
+  GST_AUDIO_BASE_SRC_SLAVE_RE_TIMESTAMP,
   GST_AUDIO_BASE_SRC_SLAVE_SKEW,
   GST_AUDIO_BASE_SRC_SLAVE_NONE
 } GstAudioBaseSrcSlaveMethod;
 
-#define GST_TYPE_AUDIO_BASE_SRC_SLAVE_METHOD (gst_audio_base_src_slave_method_get_type ())
+#define GST_AUDIO_BASE_SRC_SLAVE_RETIMESTAMP GST_AUDIO_BASE_SRC_SLAVE_RE_TIMESTAMP
 
 /**
  * GstAudioBaseSrc:
@@ -131,7 +133,6 @@ struct _GstAudioBaseSrcClass {
 };
 
 GType gst_audio_base_src_get_type(void);
-GType gst_audio_base_src_slave_method_get_type (void);
 
 GstAudioRingBuffer *
            gst_audio_base_src_create_ringbuffer        (GstAudioBaseSrc *src);

data/vendor/local/include/gstreamer-1.0/gst/audio/gstaudiocdsrc.h

@@ -54,8 +54,6 @@ typedef enum {
   GST_AUDIO_CD_SRC_MODE_CONTINUOUS       /* stream = whole disc */
 } GstAudioCdSrcMode;
 
-#define GST_TYPE_AUDIO_CD_SRC_MODE          (gst_audio_cd_src_mode_get_type ())
-
 /**
  * GstAudioCdSrcTrack:
  * @is_audio: Whether this is an audio track
@@ -134,7 +132,6 @@ struct _GstAudioCdSrcClass {
 };
 
 GType    gst_audio_cd_src_get_type (void);
-GType    gst_audio_cd_src_mode_get_type (void);
 
 gboolean gst_audio_cd_src_add_track (GstAudioCdSrc      * src,
                                      GstAudioCdSrcTrack * track);

data/vendor/local/include/gstreamer-1.0/gst/audio/gstaudioclock.h

@@ -94,10 +94,10 @@ GstClock*       gst_audio_clock_new             (const gchar *name, GstAudioCloc
                                                  gpointer user_data, GDestroyNotify destroy_notify);
 void            gst_audio_clock_reset           (GstAudioClock *clock, GstClockTime time);
 
-GstClockTime    gst_audio_clock_get_time        (GstClock * clock);
-GstClockTime    gst_audio_clock_adjust          (GstClock * clock, GstClockTime time);
+GstClockTime    gst_audio_clock_get_time        (GstAudioClock * clock);
+GstClockTime    gst_audio_clock_adjust          (GstAudioClock * clock, GstClockTime time);
 
-void            gst_audio_clock_invalidate      (GstClock * clock);
+void            gst_audio_clock_invalidate      (GstAudioClock * clock);
 
 #ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
 G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstAudioClock, gst_object_unref)

data/vendor/local/include/gstreamer-1.0/gst/audio/gstaudioringbuffer.h

@@ -83,8 +83,11 @@ typedef enum {
  * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3: samples in AC3 format
  * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3: samples in EAC3 format
  * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS: samples in DTS format
- * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC: samples in MPEG-2 AAC format
- * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC: samples in MPEG-4 AAC format
+ * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC: samples in MPEG-2 AAC ADTS format
+ * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC: samples in MPEG-4 AAC ADTS format
+ * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC_RAW: samples in MPEG-2 AAC raw format (Since 1.12)
+ * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC_RAW: samples in MPEG-4 AAC raw format (Since 1.12)
+ * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_FLAC: samples in FLAC format (Since 1.12)
  *
  * The format of the samples in the ringbuffer.
  */
@@ -101,7 +104,10 @@ typedef enum
   GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3,
   GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS,
   GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC,
-  GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC
+  GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC,
+  GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC_RAW,
+  GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC_RAW,
+  GST_AUDIO_RING_BUFFER_FORMAT_TYPE_FLAC
 } GstAudioRingBufferFormatType;
 
 /**
@@ -205,8 +211,10 @@ struct _GstAudioRingBuffer {
   gint                        may_start;
   gboolean                    active;
 
+  GDestroyNotify              cb_data_notify;
+
   /*< private >*/
-  gpointer _gst_reserved[GST_PADDING];
+  gpointer _gst_reserved[GST_PADDING - 1];
 };
 
 /**
@@ -260,9 +268,13 @@ struct _GstAudioRingBufferClass {
 GType gst_audio_ring_buffer_get_type(void);
 
 /* callback stuff */
-void            gst_audio_ring_buffer_set_callback    (GstAudioRingBuffer *buf,
-                                                       GstAudioRingBufferCallback cb,
-                                                       gpointer user_data);
+void            gst_audio_ring_buffer_set_callback      (GstAudioRingBuffer *buf,
+                                                         GstAudioRingBufferCallback cb,
+                                                         gpointer user_data);
+void            gst_audio_ring_buffer_set_callback_full (GstAudioRingBuffer *buf,
+                                                         GstAudioRingBufferCallback cb,
+                                                         gpointer user_data,
+                                                         GDestroyNotify notify);
 
 gboolean        gst_audio_ring_buffer_parse_caps      (GstAudioRingBufferSpec *spec, GstCaps *caps);
 void            gst_audio_ring_buffer_debug_spec_caps (GstAudioRingBufferSpec *spec);

data/vendor/local/include/gstreamer-1.0/gst/base/gstaggregator.h

@@ -272,9 +272,9 @@ struct _GstAggregatorClass {
  * GstAggregator methods *
  ************************/
 
-GstFlowReturn  gst_aggregator_finish_buffer         (GstAggregator                *  agg,
+GstFlowReturn  gst_aggregator_finish_buffer         (GstAggregator                *  self,
                                                      GstBuffer                    *  buffer);
-void           gst_aggregator_set_src_caps          (GstAggregator                *  agg,
+void           gst_aggregator_set_src_caps          (GstAggregator                *  self,
                                                      GstCaps                      *  caps);
 
 void           gst_aggregator_set_latency           (GstAggregator                *  self,

data/vendor/local/include/gstreamer-1.0/gst/base/gstbaseparse.h

@@ -285,6 +285,7 @@ GstBaseParseFrame * gst_base_parse_frame_new  (GstBuffer              * buffer,
 
 void            gst_base_parse_frame_init      (GstBaseParseFrame * frame);
 
+GstBaseParseFrame * gst_base_parse_frame_copy  (GstBaseParseFrame * frame);
 void            gst_base_parse_frame_free      (GstBaseParseFrame * frame);
 
 GstFlowReturn   gst_base_parse_push_frame      (GstBaseParse      * parse,
@@ -308,6 +309,8 @@ void            gst_base_parse_set_min_frame_size (GstBaseParse    * parse,
 void            gst_base_parse_set_has_timing_info (GstBaseParse   * parse,
                                                     gboolean         has_timing);
 
+void            gst_base_parse_drain           (GstBaseParse * parse);
+
 void            gst_base_parse_set_syncable    (GstBaseParse * parse,
                                                 gboolean       syncable);
 

data/vendor/local/include/gstreamer-1.0/gst/base/gstbasesink.h

@@ -122,9 +122,12 @@ struct _GstBaseSink {
  * @start: Start processing. Ideal for opening resources in the subclass
  * @stop: Stop processing. Subclasses should use this to close resources.
  * @unlock: Unlock any pending access to the resource. Subclasses should
- *     unblock any blocked function ASAP
+ *     unblock any blocked function ASAP and call gst_base_sink_wait_preroll()
  * @unlock_stop: Clear the previous unlock request. Subclasses should clear
- *     any state they set during unlock(), such as clearing command queues.
+ *     any state they set during #GstBaseSinkClass.unlock(), and be ready to
+ *     continue where they left off after gst_base_sink_wait_preroll(),
+ *     gst_base_sink_wait() or gst_wait_sink_wait_clock() return or
+ *     #GstBaseSinkClass.render() is called again.
  * @query: perform a #GstQuery on the element.
  * @event: Override this to handle events arriving on the sink pad
  * @wait_event: Override this to implement custom logic to wait for the event
@@ -207,6 +210,10 @@ GstFlowReturn   gst_base_sink_wait_preroll      (GstBaseSink *sink);
 void            gst_base_sink_set_sync          (GstBaseSink *sink, gboolean sync);
 gboolean        gst_base_sink_get_sync          (GstBaseSink *sink);
 
+/* Drop buffers which are out of segment */
+void            gst_base_sink_set_drop_out_of_segment (GstBaseSink *sink, gboolean drop_out_of_segment);
+gboolean        gst_base_sink_get_drop_out_of_segment (GstBaseSink *sink);
+
 /* dropping late buffers */
 void            gst_base_sink_set_max_lateness  (GstBaseSink *sink, gint64 max_lateness);
 gint64          gst_base_sink_get_max_lateness  (GstBaseSink *sink);

data/vendor/local/include/gstreamer-1.0/gst/base/gstbasetransform.h

@@ -208,7 +208,7 @@ struct _GstBaseTransform {
  *                   do 1-to-1 transformations on input to output buffers can either
  *                   return GST_BASE_TRANSFORM_FLOW_DROPPED or simply not generate
  *                   an output buffer until they are ready to do so. (Since 1.6)
- *                   
+ *
  * Subclasses can override any of the available virtual methods or not, as
  * needed. At minimum either @transform or @transform_ip need to be overridden.
  * If the element can overwrite the input data with the results (data is of the

data/vendor/local/include/gstreamer-1.0/gst/base/gstdataqueue.h

@@ -93,7 +93,7 @@ struct _GstDataQueueSize
  * @bytes: The amount of bytes currently in the queue.
  * @time: The accumulated duration of the items currently in the queue.
  * @checkdata: The #gpointer registered when the #GstDataQueue was created.
- * 
+ *
  * The prototype of the function used to inform the queue that it should be
  * considered as full.
  *

data/vendor/local/include/gstreamer-1.0/gst/base/gstflowcombiner.h

@@ -43,6 +43,10 @@ typedef struct _GstFlowCombiner GstFlowCombiner;
 
 GstFlowCombiner * gst_flow_combiner_new (void);
 
+GstFlowCombiner * gst_flow_combiner_ref (GstFlowCombiner * combiner);
+
+void              gst_flow_combiner_unref (GstFlowCombiner * combiner);
+
 void              gst_flow_combiner_free (GstFlowCombiner * combiner);
 
 GstFlowReturn     gst_flow_combiner_update_flow (GstFlowCombiner * combiner, GstFlowReturn fret);

data/vendor/local/include/gstreamer-1.0/gst/base/gstqueuearray.h

@@ -24,6 +24,8 @@
 #ifndef __GST_QUEUE_ARRAY_H__
 #define __GST_QUEUE_ARRAY_H__
 
+G_BEGIN_DECLS
+
 /**
  * GstQueueArray: (skip)
  */
@@ -66,4 +68,6 @@ gboolean        gst_queue_array_drop_struct      (GstQueueArray * array,
                                                   guint           idx,
                                                   gpointer        p_struct);
 
+G_END_DECLS
+
 #endif

data/vendor/local/include/gstreamer-1.0/gst/codecparsers/gsth264parser.h

@@ -178,12 +178,12 @@ typedef enum
 
 /**
  * GstH264ParserResult:
- * @GST_H264_PARSER_OK: The parsing succeded
+ * @GST_H264_PARSER_OK: The parsing succeeded
  * @GST_H264_PARSER_BROKEN_DATA: The data to parse is broken
  * @GST_H264_PARSER_BROKEN_LINK: The link to structure needed for the parsing couldn't be found
- * @GST_H264_PARSER_ERROR: An error occured when parsing
- * @GST_H264_PARSER_NO_NAL: No nal found during the parsing
- * @GST_H264_PARSER_NO_NAL_END: Start of the nal found, but not the end.
+ * @GST_H264_PARSER_ERROR: An error occurred when parsing
+ * @GST_H264_PARSER_NO_NAL: No NAL unit found during the parsing
+ * @GST_H264_PARSER_NO_NAL_END: Start of the NAL unit found, but not the end.
  *
  * The result of parsing H264 data.
  */
@@ -249,7 +249,7 @@ typedef enum
  * GstH264SEIPicStructType:
  * @GST_H264_SEI_PIC_STRUCT_FRAME: Picture is a frame
  * @GST_H264_SEI_PIC_STRUCT_TOP_FIELD: Top field of frame
- * @GST_H264_SEI_PIC_STRUCT_BOTTOM_FIELD: Botom field of frame
+ * @GST_H264_SEI_PIC_STRUCT_BOTTOM_FIELD: Bottom field of frame
  * @GST_H264_SEI_PIC_STRUCT_TOP_BOTTOM: Top bottom field of frame
  * @GST_H264_SEI_PIC_STRUCT_BOTTOM_TOP: bottom top field of frame
  * @GST_H264_SEI_PIC_STRUCT_TOP_BOTTOM_TOP: top bottom top field of frame
@@ -358,18 +358,18 @@ struct _GstH264NalUnitExtensionMVC
  *  reference picture.
  * @type: A #GstH264NalUnitType
  * @idr_pic_flag: calculated idr_pic_flag
- * @size: The size of the nal unit starting from @offset, thus
+ * @size: The size of the NAL unit starting from @offset, thus
  *  including the header bytes. e.g. @type (nal_unit_type)
- * @offset: The offset of the actual start of the nal unit, thus
+ * @offset: The offset of the actual start of the NAL unit, thus
  *  including the header bytes
- * @sc_offset: The offset of the start code of the nal unit
- * @valid: If the nal unit is valid, which means it has
+ * @sc_offset: The offset of the start code of the NAL unit
+ * @valid: If the NAL unit is valid, which means it has
  * already been parsed
- * @data: The data from which the Nalu has been parsed
+ * @data: The data from which the NAL unit has been parsed
  * @header_bytes: The size of the NALU header in bytes (Since 1.6)
  * @extension_type: the extension type (Since 1.6)
  *
- * Structure defining the Nal unit headers
+ * Structure defining the NAL unit headers
  */
 struct _GstH264NalUnit
 {
@@ -403,7 +403,7 @@ struct _GstH264NalUnit
  * SchedSelIdx-th CPB
  * @cpb_size_value_minus1: is used together with cpb_size_scale to specify the
  * SchedSelIdx-th CPB size
- * @cbr_flag: Specifies if running in itermediate bitrate mode or constant
+ * @cbr_flag: Specifies if running in constant or intermittent bit rate mode
  * @initial_cpb_removal_delay_length_minus1: specifies the length in bits of
  * the cpb_removal_delay syntax element
  * @cpb_removal_delay_length_minus1: specifies the length in bits of the
@@ -461,8 +461,8 @@ struct _GstH264HRDParams
  * @fixed_frame_rate_flag: %TRUE indicates that the temporal distance between the HRD output times
  *  of any two consecutive pictures in output order is constrained as specified in the spec, %FALSE
  *  otherwise.
- * @nal_hrd_parameters_present_flag: %TRUE if nal hrd parameters present in the bitstream
- * @vcl_hrd_parameters_present_flag: %TRUE if nal vlc hrd parameters present in the bitstream
+ * @nal_hrd_parameters_present_flag: %TRUE if NAL HRD parameters exist in the bitstream
+ * @vcl_hrd_parameters_present_flag: %TRUE if VCL HRD parameters exist in the bitstream
  * @low_delay_hrd_flag: specifies the HRD operational mode
  * @pic_struct_present_flag: %TRUE specifies that picture timing SEI messages are present or not
  * @bitstream_restriction_flag: %TRUE specifies that the following coded video sequence bitstream restriction

data/vendor/local/include/gstreamer-1.0/gst/codecparsers/gstvp8parser.h

@@ -72,7 +72,7 @@ typedef enum {
  *   index
  *
  * Dequantization indices.
- */ 
+ */
 struct _GstVp8QuantIndices
 {
   guint8 y_ac_qi;

data/vendor/local/include/gstreamer-1.0/gst/controller/controller-enumtypes.h

@@ -0,0 +1,23 @@
+
+
+
+#ifndef __GST_CONTROLLER_ENUM_TYPES_H__
+#define __GST_CONTROLLER_ENUM_TYPES_H__
+
+#include <glib-object.h>
+
+G_BEGIN_DECLS
+
+/* enumerations from "gstinterpolationcontrolsource.h" */
+GType gst_interpolation_mode_get_type (void);
+#define GST_TYPE_INTERPOLATION_MODE (gst_interpolation_mode_get_type())
+
+/* enumerations from "gstlfocontrolsource.h" */
+GType gst_lfo_waveform_get_type (void);
+#define GST_TYPE_LFO_WAVEFORM (gst_lfo_waveform_get_type())
+G_END_DECLS
+
+#endif /* __GST_CONTROLLER_ENUM_TYPES_H__ */
+
+
+

data/vendor/local/include/gstreamer-1.0/gst/controller/controller.h

@@ -24,6 +24,7 @@
 
 #include <gst/controller/gstargbcontrolbinding.h>
 #include <gst/controller/gstdirectcontrolbinding.h>
+#include <gst/controller/gstproxycontrolbinding.h>
 #include <gst/controller/gsttimedvaluecontrolsource.h>
 #include <gst/controller/gstinterpolationcontrolsource.h>
 #include <gst/controller/gsttriggercontrolsource.h>

data/vendor/local/include/gstreamer-1.0/gst/controller/gstinterpolationcontrolsource.h

@@ -28,6 +28,7 @@
 #include <gst/gst.h>
 
 #include <gst/controller/gsttimedvaluecontrolsource.h>
+#include <gst/controller/controller-enumtypes.h>
 
 G_BEGIN_DECLS
 
@@ -44,8 +45,6 @@ G_BEGIN_DECLS
 #define GST_INTERPOLATION_CONTROL_SOURCE_GET_CLASS(inst) \
   (G_TYPE_INSTANCE_GET_CLASS ((inst), GST_TYPE_INTERPOLATION_CONTROL_SOURCE, GstInterpolationControlSourceClass))
 
-#define GST_TYPE_INTERPOLATION_MODE (gst_interpolation_mode_get_type ())
-
 typedef struct _GstInterpolationControlSource GstInterpolationControlSource;
 typedef struct _GstInterpolationControlSourceClass GstInterpolationControlSourceClass;
 typedef struct _GstInterpolationControlSourcePrivate GstInterpolationControlSourcePrivate;
@@ -91,7 +90,6 @@ struct _GstInterpolationControlSourceClass {
 };
 
 GType gst_interpolation_control_source_get_type (void);
-GType gst_interpolation_mode_get_type (void);
 
 /* Functions */
 

data/vendor/local/include/gstreamer-1.0/gst/controller/gstlfocontrolsource.h

@@ -26,6 +26,7 @@
 
 #include <glib-object.h>
 #include <gst/gst.h>
+#include <gst/controller/controller-enumtypes.h>
 
 G_BEGIN_DECLS
 
@@ -42,8 +43,6 @@ G_BEGIN_DECLS
 #define GST_LFO_CONTROL_SOURCE_GET_CLASS(inst) \
   (G_TYPE_INSTANCE_GET_CLASS ((inst), GST_TYPE_LFO_CONTROL_SOURCE, GstLFOControlSourceClass))
 
-#define GST_TYPE_LFO_WAVEFORM (gst_lfo_waveform_get_type ())
-
 typedef struct _GstLFOControlSource GstLFOControlSource;
 typedef struct _GstLFOControlSourceClass GstLFOControlSourceClass;
 typedef struct _GstLFOControlSourcePrivate GstLFOControlSourcePrivate;
@@ -89,7 +88,6 @@ struct _GstLFOControlSourceClass {
 };
 
 GType gst_lfo_control_source_get_type (void);
-GType gst_lfo_waveform_get_type (void);
 
 /* Functions */