gstreamer-audio-0.10: Update bindings

diff --git a/vapi/gstreamer-audio-0.10.vapi b/vapi/gstreamer-audio-0.10.vapi
index fe3f7f18fc341167e315ba97c47285287747f9aa..2c600429dae6c7861b2c3ae1a9024b305d5711e4 100644 (file)
--- 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 b31a8d5cf52e0085ea0c5ad5edf0bc17ba773865..c6a6ebc780d6acf33aed636929471346b3a0bd58 100644 (file)
--- 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 c6a4d82dc438a2c47f6b0746dea4965e866f56c7..2d0fce168ba472ff09cab865f32a74c24396a6cb 100644 (file)
--- 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"