gstreamer-audio-0.10: Update bindings

1 files changed, 43 insertions, 13 deletions

diff --git a/vapi/gstreamer-audio-0.10.vapi b/vapi/gstreamer-audio-0.10.vapi
index fe3f7f18f..2c600429d 100644
--- a/vapi/gstreamer-audio-0.10.vapi
+++ b/vapi/gstreamer-audio-0.10.vapi
@@ -1,14 +1,16 @@
-/* gstreamer-audio-0.10.vapi generated by lt-vapigen, do not modify. */
+/* gstreamer-audio-0.10.vapi generated by vapigen, do not modify. */
 
 [CCode (cprefix = "Gst", lower_case_cprefix = "gst_")]
 namespace Gst {
 	[CCode (cheader_filename = "gst/audio/gstaudioclock.h")]
 	public class AudioClock : Gst.SystemClock {
+		public void* abidata;
 		public weak Gst.AudioClockGetTimeFunc func;
 		public Gst.ClockTime last_time;
 		public void* user_data;
 		[CCode (type = "GstClock*", has_construct_function = false)]
 		public AudioClock (string name, Gst.AudioClockGetTimeFunc func);
+		public void reset (Gst.ClockTime time);
 	}
 	[CCode (cheader_filename = "gst/audio/gstaudiofilter.h")]
 	public class AudioFilter : Gst.BaseTransform {
@@ -78,12 +80,19 @@ namespace Gst {
 		public weak Gst.RingBuffer ringbuffer;
 		public virtual unowned Gst.RingBuffer create_ringbuffer ();
 		public bool get_provide_clock ();
+		public Gst.BaseAudioSrcSlaveMethod get_slave_method ();
 		public void set_provide_clock (bool provide);
+		public void set_slave_method (Gst.BaseAudioSrcSlaveMethod method);
+		[NoAccessorMethod]
+		public int64 actual_buffer_time { get; }
+		[NoAccessorMethod]
+		public int64 actual_latency_time { get; }
 		[NoAccessorMethod]
 		public int64 buffer_time { get; set; }
 		[NoAccessorMethod]
 		public int64 latency_time { get; set; }
 		public bool provide_clock { get; set; }
+		public Gst.BaseAudioSrcSlaveMethod slave_method { get; set; }
 	}
 	[CCode (cheader_filename = "gst/audio/gstaudiofilter.h")]
 	public class RingBuffer : Gst.Object {
@@ -103,17 +112,20 @@ namespace Gst {
 		public int state;
 		public int waiting;
 		public virtual bool acquire (Gst.RingBufferSpec spec);
+		public virtual bool activate (bool active);
 		public void advance (uint advance);
 		public void clear (int segment);
 		public void clear_all ();
 		public virtual bool close_device ();
 		public uint commit (uint64 sample, uchar[] data, uint len);
-		public uint commit_full (uint64 sample, uchar[] data, int in_samples, int out_samples, int accum);
+		public uint commit_full (uint64 sample, uchar[] data, int in_samples, int out_samples, ref int accum);
+		public bool convert (Gst.Format src_fmt, int64 src_val, Gst.Format dest_fmt, out int64 dest_val);
 		public static void debug_spec_buff (Gst.RingBufferSpec spec);
 		public static void debug_spec_caps (Gst.RingBufferSpec spec);
 		public virtual uint delay ();
 		public bool device_is_open ();
 		public bool is_acquired ();
+		public bool is_active ();
 		public void may_start (bool allowed);
 		public virtual bool open_device ();
 		public static bool parse_caps (Gst.RingBufferSpec spec, Gst.Caps caps);
@@ -142,6 +154,7 @@ namespace Gst {
 		public Gst.BufferFormat format;
 		public uint64 latency_time;
 		public int rate;
+		public int seglatency;
 		public int segsize;
 		public int segtotal;
 		public bool sign;
@@ -150,7 +163,7 @@ namespace Gst {
 		public Gst.BufferFormatType type;
 		public int width;
 	}
-	[CCode (cprefix = "GST_AUDIO_CHANNEL_POSITION_", has_type_id = "0", cheader_filename = "gst/audio/multichannel.h")]
+	[CCode (cprefix = "GST_AUDIO_CHANNEL_POSITION_", cheader_filename = "gst/audio/multichannel.h")]
 	public enum AudioChannelPosition {
 		INVALID,
 		FRONT_MONO,
@@ -177,13 +190,20 @@ namespace Gst {
 		DEPTH,
 		SIGNED
 	}
-	[CCode (cprefix = "GST_BASE_AUDIO_SINK_SLAVE_", has_type_id = "0", cheader_filename = "gst/audio/gstbaseaudiosink.h")]
+	[CCode (cprefix = "", cheader_filename = "gst/audio/gstbaseaudiosink.h")]
 	public enum BaseAudioSinkSlaveMethod {
-		RESAMPLE,
-		SKEW,
-		NONE
+		Resampling slaving,
+		Skew slaving,
+		No slaving
 	}
-	[CCode (cprefix = "GST_", has_type_id = "0", cheader_filename = "gst/audio/gstringbuffer.h")]
+	[CCode (cprefix = "", cheader_filename = "gst/audio/audio.h")]
+	public enum BaseAudioSrcSlaveMethod {
+		Resampling slaving,
+		Re-timestamp,
+		Skew,
+		No slaving
+	}
+	[CCode (cprefix = "GST_", cheader_filename = "gst/audio/gstringbuffer.h")]
 	public enum BufferFormat {
 		UNKNOWN,
 		S8,
@@ -220,9 +240,13 @@ namespace Gst {
 		A_LAW,
 		IMA_ADPCM,
 		MPEG,
-		GSM
+		GSM,
+		IEC958,
+		AC3,
+		EAC3,
+		DTS
 	}
-	[CCode (cprefix = "GST_BUFTYPE_", has_type_id = "0", cheader_filename = "gst/audio/gstringbuffer.h")]
+	[CCode (cprefix = "GST_BUFTYPE_", cheader_filename = "gst/audio/gstringbuffer.h")]
 	public enum BufferFormatType {
 		LINEAR,
 		FLOAT,
@@ -230,16 +254,20 @@ namespace Gst {
 		A_LAW,
 		IMA_ADPCM,
 		MPEG,
-		GSM
+		GSM,
+		IEC958,
+		AC3,
+		EAC3,
+		DTS
 	}
-	[CCode (cprefix = "GST_SEGSTATE_", has_type_id = "0", cheader_filename = "gst/audio/gstringbuffer.h")]
+	[CCode (cprefix = "GST_SEGSTATE_", cheader_filename = "gst/audio/gstringbuffer.h")]
 	public enum RingBufferSegState {
 		INVALID,
 		EMPTY,
 		FILLED,
 		PARTIAL
 	}
-	[CCode (cprefix = "GST_RING_BUFFER_STATE_", has_type_id = "0", cheader_filename = "gst/audio/gstringbuffer.h")]
+	[CCode (cprefix = "GST_RING_BUFFER_STATE_", cheader_filename = "gst/audio/gstringbuffer.h")]
 	public enum RingBufferState {
 		STOPPED,
 		PAUSED,
@@ -263,6 +291,8 @@ namespace Gst {
 	public const string AUDIO_INT_STANDARD_PAD_TEMPLATE_CAPS;
 	[CCode (cheader_filename = "gst/audio/audio.h")]
 	public static unowned Gst.Buffer audio_buffer_clip (Gst.Buffer buffer, Gst.Segment segment, int rate, int frame_size);
+	[CCode (cheader_filename = "gst/audio/audio.h")]
+	public static bool audio_check_channel_positions (Gst.AudioChannelPosition pos, uint channels);
 	[CCode (cheader_filename = "gst/audio/mixerutils.h")]
 	public static unowned GLib.List audio_default_registry_mixer_filter (Gst.AudioMixerFilterFunc filter_func, bool first);
 	[CCode (cheader_filename = "gst/audio/audio.h")]