gstreamer-audio-1.0: Fix a few binding errors

diff --git a/vapi/gstreamer-audio-1.0.vapi b/vapi/gstreamer-audio-1.0.vapi
index 4d8192f675c1a3777bb0296782abf5868e81c8a0..516b9faee88c75f2ac33d518b583c1eaa174e732 100644 (file)
--- a/vapi/gstreamer-audio-1.0.vapi
+++ b/vapi/gstreamer-audio-1.0.vapi
@@ -213,7 +213,7 @@ namespace Gst {
                        [NoWrapper]
                        public virtual bool open ();
                        [NoWrapper]
-                       public virtual Gst.FlowReturn parse (Gst.Base.Adapter adapter, int offset, int length);
+                       public virtual Gst.FlowReturn parse (Gst.Base.Adapter adapter, out int offset, out int length);
                        [NoWrapper]
                        public virtual Gst.FlowReturn pre_push (Gst.Buffer buffer);
                        [NoWrapper]
@@ -421,7 +421,7 @@ namespace Gst {
                        public void clear (int segment);
                        public virtual void clear_all ();
                        public virtual bool close_device ();
-                       public virtual uint commit (uint64 sample, [CCode (array_length_cname = "in_samples", array_length_pos = 2.5)] uint8[] data, int out_samples, ref int accum);
+                       public virtual uint commit (ref uint64 sample, [CCode (array_length_cname = "in_samples", array_length_pos = 2.5)] uint8[] data, 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.Audio.RingBufferSpec spec);
                        public static void debug_spec_caps (Gst.Audio.RingBufferSpec spec);
@@ -493,7 +493,7 @@ namespace Gst {
                        [NoWrapper]
                        public virtual bool prepare (Gst.Audio.RingBufferSpec spec);
                        [NoWrapper]
-                       public virtual uint read ([CCode (array_length_cname = "length", array_length_pos = 1.5, array_length_type = "guint", type = "gpointer")] uint8[] data, Gst.ClockTime timestamp);
+                       public virtual uint read ([CCode (array_length_cname = "length", array_length_pos = 1.5, array_length_type = "guint", type = "gpointer")] uint8[] data, out Gst.ClockTime timestamp);
                        [NoWrapper]
                        public virtual void reset ();
                        [NoWrapper]
@@ -899,7 +899,7 @@ namespace Gst {
                }
                [CCode (cheader_filename = "gst/audio/audio.h", instance_pos = 5.9)]
                [Version (since = "1.6")]
-               public delegate void BaseSinkCustomSlavingCallback (Gst.Audio.BaseSink sink, Gst.ClockTime etime, Gst.ClockTime itime, Gst.ClockTimeDiff requested_skew, Gst.Audio.BaseSinkDiscontReason discont_reason);
+               public delegate void BaseSinkCustomSlavingCallback (Gst.Audio.BaseSink sink, Gst.ClockTime etime, Gst.ClockTime itime, out Gst.ClockTimeDiff requested_skew, Gst.Audio.BaseSinkDiscontReason discont_reason);
                [CCode (cheader_filename = "gst/audio/audio.h", instance_pos = 1.9)]
                public delegate Gst.ClockTime ClockGetTimeFunc (Gst.Clock clock);
                [CCode (cheader_filename = "gst/audio/audio.h", has_target = false)]
@@ -1068,7 +1068,7 @@ namespace Gst {
                [Version (since = "1.8")]
                public static unowned Gst.Audio.ClippingMeta? buffer_add_audio_clipping_meta (Gst.Buffer buffer, Gst.Format format, uint64 start, uint64 end);
                [CCode (cheader_filename = "gst/audio/audio.h", cname = "gst_buffer_add_audio_downmix_meta")]
-               public static unowned Gst.Audio.DownmixMeta? buffer_add_audio_downmix_meta (Gst.Buffer buffer, [CCode (array_length_cname = "from_channels", array_length_pos = 2.5)] Gst.Audio.ChannelPosition[] from_position, [CCode (array_length_cname = "to_channels", array_length_pos = 3.5)] Gst.Audio.ChannelPosition[] to_position, float matrix);
+               public static unowned Gst.Audio.DownmixMeta? buffer_add_audio_downmix_meta (Gst.Buffer buffer, [CCode (array_length_cname = "from_channels", array_length_pos = 2.5)] Gst.Audio.ChannelPosition[] from_position, [CCode (array_length_cname = "to_channels", array_length_pos = 3.5)] Gst.Audio.ChannelPosition[] to_position, [CCode (array_length = false, type = "const gfloat**")] float*[] matrix);
                [CCode (cheader_filename = "gst/audio/audio.h", cname = "gst_buffer_add_audio_level_meta")]
                [Version (since = "1.20")]
                public static unowned Gst.Audio.LevelMeta? buffer_add_audio_level_meta (Gst.Buffer buffer, uint8 level, bool voice_activity);

diff --git a/vapi/metadata/GstAudio-1.0.metadata b/vapi/metadata/GstAudio-1.0.metadata
index 5d9aaa171a46fabf69a4a4cd1cdda48bf21e3830..d0b51a892167d0e96b2f4506b4eb23ca16846d9d 100644 (file)
--- a/vapi/metadata/GstAudio-1.0.metadata
+++ b/vapi/metadata/GstAudio-1.0.metadata
@@ -4,6 +4,18 @@ AUDIO_* name="AUDIO_(.+)"
 
 audio_resampler_new symbol_type="constructor" parent="Gst.Audio.Resampler" name="new"
 
+AudioBaseSinkCustomSlavingCallback
+  .requested_skew out
+AudioDecoder
+  .parse.offset out
+  .parse.length out
+AudioRingBuffer
+  .commit.sample ref
+AudioSrc
+  .read.timestamp out
+
+buffer_add_audio_downmix_meta.matrix type="float*[]"
+
 // Field ownership
 AudioBaseSink.provided_clock unowned=false
 AudioClock.*#field skip