[vala/0.6] gstreamer-audio-0.10: Update bindings
- From: Jürg Billeter <juergbi src gnome org>
- To: svn-commits-list gnome org
- Subject: [vala/0.6] gstreamer-audio-0.10: Update bindings
- Date: Sun, 12 Apr 2009 12:14:18 -0400 (EDT)
commit ecb7da70d2fc04ec8fde1878ec1e169fd689c8c6
Author: Sebastian Pölsterl <sebp k-d-w org>
Date: Tue Apr 7 18:18:51 2009 +0200
gstreamer-audio-0.10: Update bindings
---
vapi/gstreamer-audio-0.10.vapi | 56 +++++++++---
.../gstreamer-audio-0.10/gstreamer-audio-0.10.gi | 96 ++++++++++++++++++--
.../gstreamer-audio-0.10.metadata | 2 +
3 files changed, 131 insertions(+), 23 deletions(-)
diff --git a/vapi/gstreamer-audio-0.10.vapi b/vapi/gstreamer-audio-0.10.vapi
index fe3f7f1..2c60042 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 b31a8d5..c6a6ebc 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 c6a4d82..2d0fce1 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"
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