From: Sebastian Pölsterl <sebp@k-d-w.org>
Date: Tue, 7 Apr 2009 16:18:51 +0000 (+0200)
Subject: gstreamer-audio-0.10: Update bindings
X-Git-Tag: 0.6.1~6
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=ecb7da70d2fc04ec8fde1878ec1e169fd689c8c6;p=thirdparty%2Fvala.git

gstreamer-audio-0.10: Update bindings
---

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")]
diff --git a/vapi/packages/gstreamer-audio-0.10/gstreamer-audio-0.10.gi b/vapi/packages/gstreamer-audio-0.10/gstreamer-audio-0.10.gi
index b31a8d5cf..c6a6ebc78 100644
--- a/vapi/packages/gstreamer-audio-0.10/gstreamer-audio-0.10.gi
+++ b/vapi/packages/gstreamer-audio-0.10/gstreamer-audio-0.10.gi
@@ -10,6 +10,13 @@
 				<parameter name="frame_size" type="gint"/>
 			</parameters>
 		</function>
+		<function name="audio_check_channel_positions" symbol="gst_audio_check_channel_positions">
+			<return-type type="gboolean"/>
+			<parameters>
+				<parameter name="pos" type="GstAudioChannelPosition*"/>
+				<parameter name="channels" type="guint"/>
+			</parameters>
+		</function>
 		<function name="audio_default_registry_mixer_filter" symbol="gst_audio_default_registry_mixer_filter">
 			<return-type type="GList*"/>
 			<parameters>
@@ -126,9 +133,10 @@
 			<field name="segtotal" type="gint"/>
 			<field name="bytes_per_sample" type="gint"/>
 			<field name="silence_sample" type="guint8[]"/>
-			<field name="_gst_reserved" type="gpointer[]"/>
+			<field name="seglatency" type="gint"/>
+			<field name="_gst_reserved" type="guint8[]"/>
 		</struct>
-		<enum name="GstAudioChannelPosition">
+		<enum name="GstAudioChannelPosition" type-name="GstAudioChannelPosition" get-type="gst_audio_channel_position_get_type">
 			<member name="GST_AUDIO_CHANNEL_POSITION_INVALID" value="-1"/>
 			<member name="GST_AUDIO_CHANNEL_POSITION_FRONT_MONO" value="0"/>
 			<member name="GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT" value="1"/>
@@ -153,12 +161,18 @@
 			<member name="GST_AUDIO_FIELD_DEPTH" value="16"/>
 			<member name="GST_AUDIO_FIELD_SIGNED" value="32"/>
 		</enum>
-		<enum name="GstBaseAudioSinkSlaveMethod">
-			<member name="GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE" value="0"/>
-			<member name="GST_BASE_AUDIO_SINK_SLAVE_SKEW" value="1"/>
-			<member name="GST_BASE_AUDIO_SINK_SLAVE_NONE" value="2"/>
+		<enum name="GstBaseAudioSinkSlaveMethod" type-name="GstBaseAudioSinkSlaveMethod" get-type="gst_base_audio_sink_slave_method_get_type">
+			<member name="Resampling slaving" value="0"/>
+			<member name="Skew slaving" value="1"/>
+			<member name="No slaving" value="2"/>
 		</enum>
-		<enum name="GstBufferFormat">
+		<enum name="GstBaseAudioSrcSlaveMethod" type-name="GstBaseAudioSrcSlaveMethod" get-type="gst_base_audio_src_slave_method_get_type">
+			<member name="Resampling slaving" value="0"/>
+			<member name="Re-timestamp" value="1"/>
+			<member name="Skew" value="2"/>
+			<member name="No slaving" value="3"/>
+		</enum>
+		<enum name="GstBufferFormat" type-name="GstBufferFormat" get-type="gst_buffer_format_get_type">
 			<member name="GST_UNKNOWN" value="0"/>
 			<member name="GST_S8" value="1"/>
 			<member name="GST_U8" value="2"/>
@@ -195,8 +209,12 @@
 			<member name="GST_IMA_ADPCM" value="33"/>
 			<member name="GST_MPEG" value="34"/>
 			<member name="GST_GSM" value="35"/>
+			<member name="GST_IEC958" value="36"/>
+			<member name="GST_AC3" value="37"/>
+			<member name="GST_EAC3" value="38"/>
+			<member name="GST_DTS" value="39"/>
 		</enum>
-		<enum name="GstBufferFormatType">
+		<enum name="GstBufferFormatType" type-name="GstBufferFormatType" get-type="gst_buffer_format_type_get_type">
 			<member name="GST_BUFTYPE_LINEAR" value="0"/>
 			<member name="GST_BUFTYPE_FLOAT" value="1"/>
 			<member name="GST_BUFTYPE_MU_LAW" value="2"/>
@@ -204,14 +222,18 @@
 			<member name="GST_BUFTYPE_IMA_ADPCM" value="4"/>
 			<member name="GST_BUFTYPE_MPEG" value="5"/>
 			<member name="GST_BUFTYPE_GSM" value="6"/>
+			<member name="GST_BUFTYPE_IEC958" value="7"/>
+			<member name="GST_BUFTYPE_AC3" value="8"/>
+			<member name="GST_BUFTYPE_EAC3" value="9"/>
+			<member name="GST_BUFTYPE_DTS" value="10"/>
 		</enum>
-		<enum name="GstRingBufferSegState">
+		<enum name="GstRingBufferSegState" type-name="GstRingBufferSegState" get-type="gst_ring_buffer_seg_state_get_type">
 			<member name="GST_SEGSTATE_INVALID" value="0"/>
 			<member name="GST_SEGSTATE_EMPTY" value="1"/>
 			<member name="GST_SEGSTATE_FILLED" value="2"/>
 			<member name="GST_SEGSTATE_PARTIAL" value="3"/>
 		</enum>
-		<enum name="GstRingBufferState">
+		<enum name="GstRingBufferState" type-name="GstRingBufferState" get-type="gst_ring_buffer_state_get_type">
 			<member name="GST_RING_BUFFER_STATE_STOPPED" value="0"/>
 			<member name="GST_RING_BUFFER_STATE_PAUSED" value="1"/>
 			<member name="GST_RING_BUFFER_STATE_STARTED" value="2"/>
@@ -225,9 +247,17 @@
 					<parameter name="user_data" type="gpointer"/>
 				</parameters>
 			</constructor>
+			<method name="reset" symbol="gst_audio_clock_reset">
+				<return-type type="void"/>
+				<parameters>
+					<parameter name="clock" type="GstAudioClock*"/>
+					<parameter name="time" type="GstClockTime"/>
+				</parameters>
+			</method>
 			<field name="func" type="GstAudioClockGetTimeFunc"/>
 			<field name="user_data" type="gpointer"/>
 			<field name="last_time" type="GstClockTime"/>
+			<field name="abidata" type="gpointer"/>
 		</object>
 		<object name="GstAudioFilter" parent="GstBaseTransform" type-name="GstAudioFilter" get-type="gst_audio_filter_get_type">
 			<method name="class_add_pad_templates" symbol="gst_audio_filter_class_add_pad_templates">
@@ -405,6 +435,12 @@
 					<parameter name="src" type="GstBaseAudioSrc*"/>
 				</parameters>
 			</method>
+			<method name="get_slave_method" symbol="gst_base_audio_src_get_slave_method">
+				<return-type type="GstBaseAudioSrcSlaveMethod"/>
+				<parameters>
+					<parameter name="src" type="GstBaseAudioSrc*"/>
+				</parameters>
+			</method>
 			<method name="set_provide_clock" symbol="gst_base_audio_src_set_provide_clock">
 				<return-type type="void"/>
 				<parameters>
@@ -412,9 +448,19 @@
 					<parameter name="provide" type="gboolean"/>
 				</parameters>
 			</method>
+			<method name="set_slave_method" symbol="gst_base_audio_src_set_slave_method">
+				<return-type type="void"/>
+				<parameters>
+					<parameter name="src" type="GstBaseAudioSrc*"/>
+					<parameter name="method" type="GstBaseAudioSrcSlaveMethod"/>
+				</parameters>
+			</method>
+			<property name="actual-buffer-time" type="gint64" readable="1" writable="0" construct="0" construct-only="0"/>
+			<property name="actual-latency-time" type="gint64" readable="1" writable="0" construct="0" construct-only="0"/>
 			<property name="buffer-time" type="gint64" readable="1" writable="1" construct="0" construct-only="0"/>
 			<property name="latency-time" type="gint64" readable="1" writable="1" construct="0" construct-only="0"/>
 			<property name="provide-clock" type="gboolean" readable="1" writable="1" construct="0" construct-only="0"/>
+			<property name="slave-method" type="GstBaseAudioSrcSlaveMethod" readable="1" writable="1" construct="0" construct-only="0"/>
 			<vfunc name="create_ringbuffer">
 				<return-type type="GstRingBuffer*"/>
 				<parameters>
@@ -435,6 +481,13 @@
 					<parameter name="spec" type="GstRingBufferSpec*"/>
 				</parameters>
 			</method>
+			<method name="activate" symbol="gst_ring_buffer_activate">
+				<return-type type="gboolean"/>
+				<parameters>
+					<parameter name="buf" type="GstRingBuffer*"/>
+					<parameter name="active" type="gboolean"/>
+				</parameters>
+			</method>
 			<method name="advance" symbol="gst_ring_buffer_advance">
 				<return-type type="void"/>
 				<parameters>
@@ -481,6 +534,16 @@
 					<parameter name="accum" type="gint*"/>
 				</parameters>
 			</method>
+			<method name="convert" symbol="gst_ring_buffer_convert">
+				<return-type type="gboolean"/>
+				<parameters>
+					<parameter name="buf" type="GstRingBuffer*"/>
+					<parameter name="src_fmt" type="GstFormat"/>
+					<parameter name="src_val" type="gint64"/>
+					<parameter name="dest_fmt" type="GstFormat"/>
+					<parameter name="dest_val" type="gint64*"/>
+				</parameters>
+			</method>
 			<method name="debug_spec_buff" symbol="gst_ring_buffer_debug_spec_buff">
 				<return-type type="void"/>
 				<parameters>
@@ -511,6 +574,12 @@
 					<parameter name="buf" type="GstRingBuffer*"/>
 				</parameters>
 			</method>
+			<method name="is_active" symbol="gst_ring_buffer_is_active">
+				<return-type type="gboolean"/>
+				<parameters>
+					<parameter name="buf" type="GstRingBuffer*"/>
+				</parameters>
+			</method>
 			<method name="may_start" symbol="gst_ring_buffer_may_start">
 				<return-type type="void"/>
 				<parameters>
@@ -608,6 +677,13 @@
 					<parameter name="spec" type="GstRingBufferSpec*"/>
 				</parameters>
 			</vfunc>
+			<vfunc name="activate">
+				<return-type type="gboolean"/>
+				<parameters>
+					<parameter name="buf" type="GstRingBuffer*"/>
+					<parameter name="active" type="gboolean"/>
+				</parameters>
+			</vfunc>
 			<vfunc name="close_device">
 				<return-type type="gboolean"/>
 				<parameters>
diff --git a/vapi/packages/gstreamer-audio-0.10/gstreamer-audio-0.10.metadata b/vapi/packages/gstreamer-audio-0.10/gstreamer-audio-0.10.metadata
index c6a4d82dc..2d0fce168 100644
--- a/vapi/packages/gstreamer-audio-0.10/gstreamer-audio-0.10.metadata
+++ b/vapi/packages/gstreamer-audio-0.10/gstreamer-audio-0.10.metadata
@@ -21,3 +21,5 @@ gst_audio_default_registry_mixer_filter cheader_filename="gst/audio/mixerutils.h
 gst_audio_fixate_channel_positions cheader_filename="gst/audio/multichannel.h"
 gst_audio_set_caps_channel_positions_list cheader_filename="gst/audio/multichannel.h"
 gst_audio_set_structure_channel_positions_list cheader_filename="gst/audio/multichannel.h"
+gst_ring_buffer_convert.dest_val is_out="1"
+gst_ring_buffer_commit_full.accum is_ref="1"