/usr/share/vala-0.40/vapi/gstreamer-audio-1.0.vapi is in valac-0.40-vapi 0.40.4-1.
This file is owned by root:root, with mode 0o644.
The actual contents of the file can be viewed below.
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[CCode (cprefix = "Gst", gir_namespace = "GstAudio", gir_version = "1.0", lower_case_cprefix = "gst_")]
namespace Gst {
namespace Audio {
[CCode (cheader_filename = "gst/audio/audio.h", type_id = "gst_audio_aggregator_get_type ()")]
[GIR (name = "AudioAggregator")]
public abstract class Aggregator : Gst.Base.Aggregator {
public weak Gst.Caps current_caps;
[CCode (has_construct_function = false)]
protected Aggregator ();
[NoWrapper]
public virtual bool aggregate_one_buffer (Gst.Audio.AggregatorPad pad, Gst.Buffer inbuf, uint in_offset, Gst.Buffer outbuf, uint out_offset, uint num_frames);
[NoWrapper]
public virtual Gst.Buffer create_output_buffer (uint num_frames);
public void set_sink_caps (Gst.Audio.AggregatorPad pad, Gst.Caps caps);
[NoAccessorMethod]
public uint64 alignment_threshold { get; set; }
[NoAccessorMethod]
public uint64 discont_wait { get; set; }
[NoAccessorMethod]
public uint64 output_buffer_duration { get; set; }
}
[CCode (cheader_filename = "gst/audio/audio.h", type_id = "gst_audio_aggregator_convert_pad_get_type ()")]
[GIR (name = "AudioAggregatorConvertPad")]
public class AggregatorConvertPad : Gst.Audio.AggregatorPad {
[CCode (has_construct_function = false)]
protected AggregatorConvertPad ();
[NoAccessorMethod]
public Gst.Structure converter_config { owned get; set; }
}
[CCode (cheader_filename = "gst/audio/audio.h", type_id = "gst_audio_aggregator_pad_get_type ()")]
[GIR (name = "AudioAggregatorPad")]
public class AggregatorPad : Gst.Base.AggregatorPad {
public weak Gst.Audio.Info info;
[CCode (has_construct_function = false)]
protected AggregatorPad ();
[NoWrapper]
public virtual Gst.Buffer convert_buffer (Gst.Audio.Info in_info, Gst.Audio.Info out_info, Gst.Buffer buffer);
[NoWrapper]
public virtual void update_conversion_info ();
}
[CCode (cheader_filename = "gst/audio/audio.h", type_id = "gst_audio_base_sink_get_type ()")]
[GIR (name = "AudioBaseSink")]
public class BaseSink : Gst.Base.Sink {
public bool eos_rendering;
public uint64 next_sample;
public Gst.Clock provided_clock;
public weak Gst.Audio.RingBuffer ringbuffer;
[CCode (has_construct_function = false)]
protected BaseSink ();
public virtual unowned Gst.Audio.RingBuffer create_ringbuffer ();
public Gst.ClockTime get_alignment_threshold ();
public Gst.ClockTime get_discont_wait ();
public int64 get_drift_tolerance ();
public bool get_provide_clock ();
public Gst.Audio.BaseSinkSlaveMethod get_slave_method ();
[NoWrapper]
public virtual Gst.Buffer payload (Gst.Buffer buffer);
[Version (since = "1.6")]
public void report_device_failure ();
public void set_alignment_threshold (Gst.ClockTime alignment_threshold);
[Version (since = "1.6")]
public void set_custom_slaving_callback (owned Gst.Audio.BaseSinkCustomSlavingCallback callback);
public void set_discont_wait (Gst.ClockTime discont_wait);
public void set_drift_tolerance (int64 drift_tolerance);
public void set_provide_clock (bool provide);
public void set_slave_method (Gst.Audio.BaseSinkSlaveMethod method);
public uint64 alignment_threshold { get; set; }
[NoAccessorMethod]
public int64 buffer_time { get; set; }
[NoAccessorMethod]
public bool can_activate_pull { get; set; }
public uint64 discont_wait { get; set; }
public int64 drift_tolerance { get; set; }
[NoAccessorMethod]
public int64 latency_time { get; set; }
public bool provide_clock { get; set; }
public Gst.Audio.BaseSinkSlaveMethod slave_method { get; set; }
}
[CCode (cheader_filename = "gst/audio/audio.h", type_id = "gst_audio_base_src_get_type ()")]
[GIR (name = "AudioBaseSrc")]
public class BaseSrc : Gst.Base.PushSrc {
public weak Gst.Clock clock;
public uint64 next_sample;
public weak Gst.Audio.RingBuffer ringbuffer;
[CCode (has_construct_function = false)]
protected BaseSrc ();
public virtual unowned Gst.Audio.RingBuffer create_ringbuffer ();
public bool get_provide_clock ();
public Gst.Audio.BaseSrcSlaveMethod get_slave_method ();
public void set_provide_clock (bool provide);
public void set_slave_method (Gst.Audio.BaseSrcSlaveMethod 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.Audio.BaseSrcSlaveMethod slave_method { get; set; }
}
[CCode (cheader_filename = "gst/audio/audio.h", type_id = "gst_audio_cd_src_get_type ()")]
[GIR (name = "AudioCdSrc")]
public class CdSrc : Gst.Base.PushSrc, Gst.URIHandler {
public weak Gst.TagList tags;
[CCode (has_construct_function = false)]
protected CdSrc ();
public bool add_track (Gst.Audio.CdSrcTrack track);
[NoWrapper]
public virtual void close ();
[NoWrapper]
public virtual bool open (string device);
[NoWrapper]
public virtual Gst.Buffer read_sector (int sector);
[NoAccessorMethod]
public string device { owned get; set; }
[NoAccessorMethod]
public Gst.Audio.CdSrcMode mode { get; set; }
[NoAccessorMethod]
public uint track { get; set; }
}
[CCode (cheader_filename = "gst/audio/audio.h", has_type_id = false)]
[Compact]
[GIR (name = "AudioChannelMixer")]
public class ChannelMixer {
public void free ();
public bool is_passthrough ();
public void samples (void* @in, void* @out, int samples);
}
[CCode (cheader_filename = "gst/audio/audio.h", type_id = "gst_audio_clock_get_type ()")]
[GIR (name = "AudioClock")]
public class Clock : Gst.SystemClock {
[CCode (has_construct_function = false, type = "GstClock*")]
public Clock (string name, owned Gst.Audio.ClockGetTimeFunc func);
public Gst.ClockTime adjust (Gst.ClockTime time);
public Gst.ClockTime get_time ();
public void invalidate ();
public void reset (Gst.ClockTime time);
}
[CCode (cheader_filename = "gst/audio/audio.h", copy_function = "g_boxed_copy", free_function = "g_boxed_free", type_id = "gst_audio_converter_get_type ()")]
[Compact]
[GIR (name = "AudioConverter")]
public class Converter {
[CCode (has_construct_function = false)]
public Converter (Gst.Audio.ConverterFlags flags, Gst.Audio.Info in_info, Gst.Audio.Info out_info, owned Gst.Structure? config);
[Version (since = "1.14")]
public bool convert (Gst.Audio.ConverterFlags flags, [CCode (array_length_cname = "in_size", array_length_pos = 2.5, array_length_type = "gsize")] uint8[] @in, [CCode (array_length_cname = "out_size", array_length_pos = 3.1, array_length_type = "gsize")] out uint8[] @out);
public void free ();
public unowned Gst.Structure get_config (int in_rate, int out_rate);
public size_t get_in_frames (size_t out_frames);
public size_t get_max_latency ();
public size_t get_out_frames (size_t in_frames);
public void reset ();
public bool samples (Gst.Audio.ConverterFlags flags, void* @in, size_t in_frames, void* @out, size_t out_frames);
public bool supports_inplace ();
public bool update_config (int in_rate, int out_rate, owned Gst.Structure? config);
}
[CCode (cheader_filename = "gst/audio/audio.h", type_id = "gst_audio_decoder_get_type ()")]
[GIR (name = "AudioDecoder")]
public abstract class Decoder : Gst.Element {
public weak Gst.Segment input_segment;
public weak Gst.Segment output_segment;
public weak Gst.Pad sinkpad;
public weak Gst.Pad srcpad;
public GLib.RecMutex stream_lock;
[CCode (has_construct_function = false)]
protected Decoder ();
public Gst.Buffer allocate_output_buffer (size_t size);
[NoWrapper]
public virtual bool close ();
[NoWrapper]
public virtual bool decide_allocation (Gst.Query query);
public Gst.FlowReturn finish_frame (Gst.Buffer buf, int frames);
[NoWrapper]
public virtual void flush (bool hard);
public void get_allocator (out Gst.Allocator allocator, out Gst.AllocationParams @params);
public Gst.Audio.Info get_audio_info ();
public int get_delay ();
public bool get_drainable ();
public int get_estimate_rate ();
public void get_latency (out Gst.ClockTime min, out Gst.ClockTime max);
public int get_max_errors ();
public Gst.ClockTime get_min_latency ();
public bool get_needs_format ();
public void get_parse_state (bool sync, bool eos);
public bool get_plc ();
public int get_plc_aware ();
public Gst.ClockTime get_tolerance ();
[NoWrapper]
public virtual Gst.Caps getcaps (Gst.Caps filter);
[NoWrapper]
public virtual Gst.FlowReturn handle_frame (Gst.Buffer buffer);
public void merge_tags (Gst.TagList? tags, Gst.TagMergeMode mode);
public virtual bool negotiate ();
[NoWrapper]
public virtual bool open ();
[NoWrapper]
public virtual Gst.FlowReturn parse (Gst.Base.Adapter adapter, int offset, int length);
[NoWrapper]
public virtual Gst.FlowReturn pre_push (Gst.Buffer buffer);
[NoWrapper]
public virtual bool propose_allocation (Gst.Query query);
[Version (since = "1.6")]
public Gst.Caps proxy_getcaps (Gst.Caps? caps, Gst.Caps? filter);
[Version (since = "1.10")]
public void set_allocation_caps (Gst.Caps? allocation_caps);
public void set_drainable (bool enabled);
public void set_estimate_rate (bool enabled);
[NoWrapper]
public virtual bool set_format (Gst.Caps caps);
public void set_latency (Gst.ClockTime min, Gst.ClockTime max);
public void set_max_errors (int num);
public void set_min_latency (Gst.ClockTime num);
public void set_needs_format (bool enabled);
public bool set_output_format (Gst.Audio.Info info);
public void set_plc (bool enabled);
public void set_plc_aware (bool plc);
public void set_tolerance (Gst.ClockTime tolerance);
[Version (since = "1.6")]
public void set_use_default_pad_acceptcaps (bool use);
[NoWrapper]
public virtual bool sink_event (Gst.Event event);
[NoWrapper]
public virtual bool sink_query (Gst.Query query);
[NoWrapper]
public virtual bool src_event (Gst.Event event);
[NoWrapper]
public virtual bool src_query (Gst.Query query);
[NoWrapper]
public virtual bool start ();
[NoWrapper]
public virtual bool stop ();
[NoWrapper]
public virtual bool transform_meta (Gst.Buffer outbuf, Gst.Meta meta, Gst.Buffer inbuf);
public int64 min_latency { get; set; }
public bool plc { get; set; }
public int64 tolerance { get; set; }
}
[CCode (cheader_filename = "gst/audio/audio.h", type_id = "gst_audio_encoder_get_type ()")]
[GIR (name = "AudioEncoder")]
public abstract class Encoder : Gst.Element, Gst.Preset {
public weak Gst.Segment input_segment;
public weak Gst.Segment output_segment;
public weak Gst.Pad sinkpad;
public weak Gst.Pad srcpad;
public GLib.RecMutex stream_lock;
[CCode (has_construct_function = false)]
protected Encoder ();
public Gst.Buffer allocate_output_buffer (size_t size);
[NoWrapper]
public virtual bool close ();
[NoWrapper]
public virtual bool decide_allocation (Gst.Query query);
public Gst.FlowReturn finish_frame (Gst.Buffer buffer, int samples);
[NoWrapper]
public virtual void flush ();
public void get_allocator (out Gst.Allocator allocator, out Gst.AllocationParams @params);
public Gst.Audio.Info get_audio_info ();
public bool get_drainable ();
public int get_frame_max ();
public int get_frame_samples_max ();
public int get_frame_samples_min ();
public bool get_hard_min ();
public bool get_hard_resync ();
public void get_latency (out Gst.ClockTime min, out Gst.ClockTime max);
public int get_lookahead ();
public bool get_mark_granule ();
public bool get_perfect_timestamp ();
public Gst.ClockTime get_tolerance ();
[NoWrapper]
public virtual Gst.Caps getcaps (Gst.Caps filter);
[NoWrapper]
public virtual Gst.FlowReturn handle_frame (Gst.Buffer buffer);
public void merge_tags (Gst.TagList? tags, Gst.TagMergeMode mode);
public virtual bool negotiate ();
[NoWrapper]
public virtual bool open ();
[NoWrapper]
public virtual Gst.FlowReturn pre_push (Gst.Buffer buffer);
[NoWrapper]
public virtual bool propose_allocation (Gst.Query query);
public Gst.Caps proxy_getcaps (Gst.Caps? caps, Gst.Caps? filter);
[Version (since = "1.10")]
public void set_allocation_caps (Gst.Caps? allocation_caps);
public void set_drainable (bool enabled);
[NoWrapper]
public virtual bool set_format (Gst.Audio.Info info);
public void set_frame_max (int num);
public void set_frame_samples_max (int num);
public void set_frame_samples_min (int num);
public void set_hard_min (bool enabled);
public void set_hard_resync (bool enabled);
public void set_headers (owned GLib.List<Gst.Buffer> headers);
public void set_latency (Gst.ClockTime min, Gst.ClockTime max);
public void set_lookahead (int num);
public void set_mark_granule (bool enabled);
public bool set_output_format (Gst.Caps caps);
public void set_perfect_timestamp (bool enabled);
public void set_tolerance (Gst.ClockTime tolerance);
[NoWrapper]
public virtual bool sink_event (Gst.Event event);
[NoWrapper]
public virtual bool sink_query (Gst.Query query);
[NoWrapper]
public virtual bool src_event (Gst.Event event);
[NoWrapper]
public virtual bool src_query (Gst.Query query);
[NoWrapper]
public virtual bool start ();
[NoWrapper]
public virtual bool stop ();
[NoWrapper]
public virtual bool transform_meta (Gst.Buffer outbuf, Gst.Meta meta, Gst.Buffer inbuf);
public bool hard_resync { get; set; }
public bool mark_granule { get; }
public bool perfect_timestamp { get; set; }
public int64 tolerance { get; set; }
}
[CCode (cheader_filename = "gst/audio/audio.h", type_id = "gst_audio_filter_get_type ()")]
[GIR (name = "AudioFilter")]
public abstract class Filter : Gst.Base.Transform {
public weak Gst.Audio.Info info;
[CCode (has_construct_function = false)]
protected Filter ();
[NoWrapper]
public virtual bool setup (Gst.Audio.Info info);
}
[CCode (cheader_filename = "gst/audio/audio.h", copy_function = "g_boxed_copy", free_function = "g_boxed_free", type_id = "gst_audio_info_get_type ()")]
[Compact]
[GIR (name = "AudioInfo")]
public class Info {
public int bpf;
public int channels;
public Gst.Audio.FormatInfo finfo;
public Gst.Audio.Flags flags;
public Gst.Audio.Layout layout;
[CCode (array_length = false)]
public weak Gst.Audio.ChannelPosition position[64];
public int rate;
[CCode (has_construct_function = false)]
public Info ();
public bool convert (Gst.Format src_fmt, int64 src_val, Gst.Format dest_fmt, int64 dest_val);
public Gst.Audio.Info copy ();
public void free ();
public bool from_caps (Gst.Caps caps);
public void init ();
[Version (since = "1.2")]
public bool is_equal (Gst.Audio.Info other);
public void set_format (Gst.Audio.Format format, int rate, int channels, [CCode (array_length = false)] Gst.Audio.ChannelPosition position[64]);
public Gst.Caps to_caps ();
}
[CCode (cheader_filename = "gst/audio/audio.h", has_type_id = false)]
[Compact]
[GIR (name = "AudioQuantize")]
public class Quantize {
public void free ();
public void reset ();
public void samples (void* @in, void* @out, uint samples);
}
[CCode (cheader_filename = "gst/audio/audio.h", has_type_id = false)]
[Compact]
[GIR (name = "AudioResampler")]
public class Resampler {
[CCode (has_construct_function = false)]
public Resampler (Gst.Audio.ResamplerMethod method, Gst.Audio.ResamplerFlags flags, Gst.Audio.Format format, int channels, int in_rate, int out_rate, Gst.Structure options);
[Version (since = "1.6")]
public void free ();
public size_t get_in_frames (size_t out_frames);
public size_t get_max_latency ();
public size_t get_out_frames (size_t in_frames);
public void resample (void* @in, size_t in_frames, void* @out, size_t out_frames);
public void reset ();
public bool update (int in_rate, int out_rate, Gst.Structure options);
}
[CCode (cheader_filename = "gst/audio/audio.h", type_id = "gst_audio_ring_buffer_get_type ()")]
[GIR (name = "AudioRingBuffer")]
public abstract class RingBuffer : Gst.Object {
public bool acquired;
public GLib.Cond cond;
public uint8 empty_seg;
public uint8 memory;
public bool open;
public int samples_per_seg;
public int segbase;
public int segdone;
public size_t size;
public Gst.Audio.RingBufferSpec spec;
public int state;
public int waiting;
[CCode (has_construct_function = false)]
protected RingBuffer ();
public virtual bool acquire (Gst.Audio.RingBufferSpec spec);
public virtual bool activate (bool active);
public void advance (uint advance);
public void clear (int segment);
public virtual void clear_all ();
public virtual bool close_device ();
public virtual uint commit (uint64 sample, uint8 data, int in_samples, int out_samples, int accum);
public bool convert (Gst.Format src_fmt, int64 src_val, Gst.Format dest_fmt, int64 dest_val);
public static void debug_spec_buff (Gst.Audio.RingBufferSpec spec);
public static void debug_spec_caps (Gst.Audio.RingBufferSpec spec);
public virtual uint delay ();
public bool device_is_open ();
public bool is_acquired ();
public bool is_active ();
public bool is_flushing ();
public void may_start (bool allowed);
public virtual bool open_device ();
public static bool parse_caps (Gst.Audio.RingBufferSpec spec, Gst.Caps caps);
public virtual bool pause ();
public bool prepare_read (int segment, uint8 readptr, int len);
public uint read (uint64 sample, uint8 data, uint len, Gst.ClockTime timestamp);
public virtual bool release ();
[NoWrapper]
public virtual bool resume ();
public uint64 samples_done ();
[Version (since = "1.12")]
public void set_callback_full (owned Gst.Audio.RingBufferCallback? cb);
public void set_channel_positions (Gst.Audio.ChannelPosition position);
public void set_flushing (bool flushing);
public void set_sample (uint64 sample);
public void set_timestamp (int readseg, Gst.ClockTime timestamp);
public virtual bool start ();
public virtual bool stop ();
}
[CCode (cheader_filename = "gst/audio/audio.h", type_id = "gst_audio_sink_get_type ()")]
[GIR (name = "AudioSink")]
public class Sink : Gst.Audio.BaseSink {
[CCode (has_construct_function = false)]
protected Sink ();
[NoWrapper]
public virtual bool close ();
[NoWrapper]
public virtual uint delay ();
[NoWrapper]
public virtual bool open ();
[NoWrapper]
public virtual bool prepare (Gst.Audio.RingBufferSpec spec);
[NoWrapper]
public virtual void reset ();
[NoWrapper]
public virtual bool unprepare ();
[NoWrapper]
public virtual int write ([CCode (array_length_cname = "length", array_length_pos = 1.1, array_length_type = "guint", type = "gpointer")] uint8[] data);
}
[CCode (cheader_filename = "gst/audio/audio.h", type_id = "gst_audio_src_get_type ()")]
[GIR (name = "AudioSrc")]
public class Src : Gst.Audio.BaseSrc {
[CCode (has_construct_function = false)]
protected Src ();
[NoWrapper]
public virtual bool close ();
[NoWrapper]
public virtual uint delay ();
[NoWrapper]
public virtual bool open ();
[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);
[NoWrapper]
public virtual void reset ();
[NoWrapper]
public virtual bool unprepare ();
}
[CCode (cheader_filename = "gst/audio/audio.h", copy_function = "g_boxed_copy", free_function = "g_boxed_free", type_id = "gst_audio_stream_align_get_type ()")]
[Compact]
[GIR (name = "AudioStreamAlign")]
[Version (since = "1.14")]
public class StreamAlign {
[CCode (has_construct_function = false)]
public StreamAlign (int rate, Gst.ClockTime alignment_threshold, Gst.ClockTime discont_wait);
public Gst.Audio.StreamAlign copy ();
public void free ();
public Gst.ClockTime get_alignment_threshold ();
public Gst.ClockTime get_discont_wait ();
public int get_rate ();
public uint64 get_samples_since_discont ();
public Gst.ClockTime get_timestamp_at_discont ();
public void mark_discont ();
public bool process (bool discont, Gst.ClockTime timestamp, uint n_samples, out Gst.ClockTime out_timestamp, out Gst.ClockTime out_duration, out uint64 out_sample_position);
public void set_alignment_threshold (Gst.ClockTime alignment_threshold);
public void set_discont_wait (Gst.ClockTime discont_wait);
public void set_rate (int rate);
}
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GstStreamVolume", lower_case_cprefix = "gst_stream_volume_", type_cname = "GstStreamVolumeInterface", type_id = "gst_stream_volume_get_type ()")]
[GIR (name = "StreamVolume")]
public interface StreamVolume : GLib.Object {
public static double convert_volume (Gst.Audio.StreamVolumeFormat from, Gst.Audio.StreamVolumeFormat to, double val);
public bool get_mute ();
public double get_volume (Gst.Audio.StreamVolumeFormat format);
public void set_mute (bool mute);
public void set_volume (Gst.Audio.StreamVolumeFormat format, double val);
[ConcreteAccessor]
public abstract bool mute { get; set; }
[NoAccessorMethod]
public abstract double volume { get; set; }
}
[CCode (cheader_filename = "gst/audio/audio.h", has_type_id = false)]
[GIR (name = "AudioCdSrcTrack")]
public struct CdSrcTrack {
public bool is_audio;
public uint num;
public uint start;
public uint end;
public weak Gst.TagList tags;
}
[CCode (cheader_filename = "gst/audio/audio.h", has_type_id = false)]
[GIR (name = "AudioClippingMeta")]
[Version (since = "1.8")]
public struct ClippingMeta {
public Gst.Meta meta;
public Gst.Format format;
public uint64 start;
public uint64 end;
}
[CCode (cheader_filename = "gst/audio/audio.h", has_type_id = false)]
[GIR (name = "AudioDownmixMeta")]
public struct DownmixMeta {
public Gst.Meta meta;
public Gst.Audio.ChannelPosition from_position;
public Gst.Audio.ChannelPosition to_position;
public int from_channels;
public int to_channels;
public float matrix;
}
[CCode (cheader_filename = "gst/audio/audio.h", has_type_id = false)]
[GIR (name = "AudioFormatInfo")]
public struct FormatInfo {
public Gst.Audio.Format format;
public weak string name;
public weak string description;
public Gst.Audio.FormatFlags flags;
public int endianness;
public int width;
public int depth;
[CCode (array_length = false)]
public weak uint8 silence[8];
public Gst.Audio.Format unpack_format;
public weak Gst.Audio.FormatUnpack unpack_func;
public weak Gst.Audio.FormatPack pack_func;
}
[CCode (cheader_filename = "gst/audio/audio.h", has_type_id = false)]
[GIR (name = "AudioRingBufferSpec")]
public struct RingBufferSpec {
public weak Gst.Caps caps;
public Gst.Audio.RingBufferFormatType type;
public weak Gst.Audio.Info info;
public uint64 latency_time;
public uint64 buffer_time;
public int segsize;
public int segtotal;
public int seglatency;
}
[CCode (cheader_filename = "gst/audio/audio.h", cprefix = "GST_AUDIO_BASE_SINK_DISCONT_REASON_", type_id = "gst_audio_base_sink_discont_reason_get_type ()")]
[GIR (name = "AudioBaseSinkDiscontReason")]
[Version (since = "1.6")]
public enum BaseSinkDiscontReason {
NO_DISCONT,
NEW_CAPS,
FLUSH,
SYNC_LATENCY,
ALIGNMENT,
DEVICE_FAILURE
}
[CCode (cheader_filename = "gst/audio/audio.h", cprefix = "GST_AUDIO_BASE_SINK_SLAVE_", type_id = "gst_audio_base_sink_slave_method_get_type ()")]
[GIR (name = "AudioBaseSinkSlaveMethod")]
public enum BaseSinkSlaveMethod {
RESAMPLE,
SKEW,
NONE,
CUSTOM
}
[CCode (cheader_filename = "gst/audio/audio.h", cprefix = "GST_AUDIO_BASE_SRC_SLAVE_", type_id = "gst_audio_base_src_slave_method_get_type ()")]
[GIR (name = "AudioBaseSrcSlaveMethod")]
public enum BaseSrcSlaveMethod {
RESAMPLE,
RE_TIMESTAMP,
SKEW,
NONE
}
[CCode (cheader_filename = "gst/audio/audio.h", cprefix = "GST_AUDIO_CD_SRC_MODE_", type_id = "gst_audio_cd_src_mode_get_type ()")]
[GIR (name = "AudioCdSrcMode")]
public enum CdSrcMode {
NORMAL,
CONTINUOUS
}
[CCode (cheader_filename = "gst/audio/audio.h", cprefix = "GST_AUDIO_CHANNEL_MIXER_FLAGS_", type_id = "gst_audio_channel_mixer_flags_get_type ()")]
[Flags]
[GIR (name = "AudioChannelMixerFlags")]
public enum ChannelMixerFlags {
NONE,
NON_INTERLEAVED_IN,
NON_INTERLEAVED_OUT,
UNPOSITIONED_IN,
UNPOSITIONED_OUT
}
[CCode (cheader_filename = "gst/audio/audio.h", cprefix = "GST_AUDIO_CHANNEL_POSITION_", type_id = "gst_audio_channel_position_get_type ()")]
[GIR (name = "AudioChannelPosition")]
public enum ChannelPosition {
NONE,
MONO,
INVALID,
FRONT_LEFT,
FRONT_RIGHT,
FRONT_CENTER,
LFE1,
REAR_LEFT,
REAR_RIGHT,
FRONT_LEFT_OF_CENTER,
FRONT_RIGHT_OF_CENTER,
REAR_CENTER,
LFE2,
SIDE_LEFT,
SIDE_RIGHT,
TOP_FRONT_LEFT,
TOP_FRONT_RIGHT,
TOP_FRONT_CENTER,
TOP_CENTER,
TOP_REAR_LEFT,
TOP_REAR_RIGHT,
TOP_SIDE_LEFT,
TOP_SIDE_RIGHT,
TOP_REAR_CENTER,
BOTTOM_FRONT_CENTER,
BOTTOM_FRONT_LEFT,
BOTTOM_FRONT_RIGHT,
WIDE_LEFT,
WIDE_RIGHT,
SURROUND_LEFT,
SURROUND_RIGHT
}
[CCode (cheader_filename = "gst/audio/audio.h", cprefix = "GST_AUDIO_CONVERTER_FLAG_", type_id = "gst_audio_converter_flags_get_type ()")]
[Flags]
[GIR (name = "AudioConverterFlags")]
public enum ConverterFlags {
NONE,
IN_WRITABLE,
VARIABLE_RATE
}
[CCode (cheader_filename = "gst/audio/audio.h", cprefix = "GST_AUDIO_DITHER_", type_id = "gst_audio_dither_method_get_type ()")]
[GIR (name = "AudioDitherMethod")]
public enum DitherMethod {
NONE,
RPDF,
TPDF,
TPDF_HF
}
[CCode (cheader_filename = "gst/audio/audio.h", cprefix = "GST_AUDIO_FLAG_", type_id = "gst_audio_flags_get_type ()")]
[Flags]
[GIR (name = "AudioFlags")]
public enum Flags {
NONE,
UNPOSITIONED
}
[CCode (cheader_filename = "gst/audio/audio.h", cprefix = "GST_AUDIO_FORMAT_", type_id = "gst_audio_format_get_type ()")]
[GIR (name = "AudioFormat")]
public enum Format {
UNKNOWN,
ENCODED,
S8,
U8,
S16LE,
S16BE,
U16LE,
U16BE,
S24_32LE,
S24_32BE,
U24_32LE,
U24_32BE,
S32LE,
S32BE,
U32LE,
U32BE,
S24LE,
S24BE,
U24LE,
U24BE,
S20LE,
S20BE,
U20LE,
U20BE,
S18LE,
S18BE,
U18LE,
U18BE,
F32LE,
F32BE,
F64LE,
F64BE,
S16,
U16,
S24_32,
U24_32,
S32,
U32,
S24,
U24,
S20,
U20,
S18,
U18,
F32,
F64
}
[CCode (cheader_filename = "gst/audio/audio.h", cprefix = "GST_AUDIO_FORMAT_FLAG_", type_id = "gst_audio_format_flags_get_type ()")]
[Flags]
[GIR (name = "AudioFormatFlags")]
public enum FormatFlags {
INTEGER,
FLOAT,
SIGNED,
COMPLEX,
UNPACK
}
[CCode (cheader_filename = "gst/audio/audio.h", cprefix = "GST_AUDIO_LAYOUT_", type_id = "gst_audio_layout_get_type ()")]
[GIR (name = "AudioLayout")]
public enum Layout {
INTERLEAVED,
NON_INTERLEAVED
}
[CCode (cheader_filename = "gst/audio/audio.h", cprefix = "GST_AUDIO_NOISE_SHAPING_", type_id = "gst_audio_noise_shaping_method_get_type ()")]
[GIR (name = "AudioNoiseShapingMethod")]
public enum NoiseShapingMethod {
NONE,
ERROR_FEEDBACK,
SIMPLE,
MEDIUM,
HIGH
}
[CCode (cheader_filename = "gst/audio/audio.h", cprefix = "GST_AUDIO_PACK_FLAG_", type_id = "gst_audio_pack_flags_get_type ()")]
[Flags]
[GIR (name = "AudioPackFlags")]
public enum PackFlags {
NONE,
TRUNCATE_RANGE
}
[CCode (cheader_filename = "gst/audio/audio.h", cprefix = "GST_AUDIO_QUANTIZE_FLAG_", type_id = "gst_audio_quantize_flags_get_type ()")]
[Flags]
[GIR (name = "AudioQuantizeFlags")]
public enum QuantizeFlags {
NONE,
NON_INTERLEAVED
}
[CCode (cheader_filename = "gst/audio/audio.h", cprefix = "GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_", type_id = "gst_audio_resampler_filter_interpolation_get_type ()")]
[GIR (name = "AudioResamplerFilterInterpolation")]
public enum ResamplerFilterInterpolation {
NONE,
LINEAR,
CUBIC
}
[CCode (cheader_filename = "gst/audio/audio.h", cprefix = "GST_AUDIO_RESAMPLER_FILTER_MODE_", type_id = "gst_audio_resampler_filter_mode_get_type ()")]
[GIR (name = "AudioResamplerFilterMode")]
public enum ResamplerFilterMode {
INTERPOLATED,
FULL,
AUTO
}
[CCode (cheader_filename = "gst/audio/audio.h", cprefix = "GST_AUDIO_RESAMPLER_FLAG_", type_id = "gst_audio_resampler_flags_get_type ()")]
[Flags]
[GIR (name = "AudioResamplerFlags")]
public enum ResamplerFlags {
NONE,
NON_INTERLEAVED_IN,
NON_INTERLEAVED_OUT,
VARIABLE_RATE
}
[CCode (cheader_filename = "gst/audio/audio.h", cprefix = "GST_AUDIO_RESAMPLER_METHOD_", type_id = "gst_audio_resampler_method_get_type ()")]
[GIR (name = "AudioResamplerMethod")]
[Version (since = "1.6")]
public enum ResamplerMethod {
NEAREST,
LINEAR,
CUBIC,
BLACKMAN_NUTTALL,
KAISER
}
[CCode (cheader_filename = "gst/audio/audio.h", cprefix = "GST_AUDIO_RING_BUFFER_FORMAT_TYPE_", type_id = "gst_audio_ring_buffer_format_type_get_type ()")]
[GIR (name = "AudioRingBufferFormatType")]
public enum RingBufferFormatType {
RAW,
MU_LAW,
A_LAW,
IMA_ADPCM,
MPEG,
GSM,
IEC958,
AC3,
EAC3,
DTS,
MPEG2_AAC,
MPEG4_AAC,
MPEG2_AAC_RAW,
MPEG4_AAC_RAW,
FLAC
}
[CCode (cheader_filename = "gst/audio/audio.h", cprefix = "GST_AUDIO_RING_BUFFER_STATE_", type_id = "gst_audio_ring_buffer_state_get_type ()")]
[GIR (name = "AudioRingBufferState")]
public enum RingBufferState {
STOPPED,
PAUSED,
STARTED,
ERROR
}
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GstStreamVolumeFormat", cprefix = "GST_STREAM_VOLUME_FORMAT_", has_type_id = false)]
[GIR (name = "StreamVolumeFormat")]
public enum StreamVolumeFormat {
LINEAR,
CUBIC,
DB
}
[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);
[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)]
public delegate void FormatPack (Gst.Audio.FormatInfo info, Gst.Audio.PackFlags flags, [CCode (array_length = false)] uint8[] src, [CCode (array_length = false)] uint8[] data, int length);
[CCode (cheader_filename = "gst/audio/audio.h", has_target = false)]
public delegate void FormatUnpack (Gst.Audio.FormatInfo info, Gst.Audio.PackFlags flags, [CCode (array_length = false)] uint8[] dest, [CCode (array_length = false)] uint8[] data, int length);
[CCode (cheader_filename = "gst/audio/audio.h", instance_pos = 2.9)]
public delegate void RingBufferCallback (Gst.Audio.RingBuffer rbuf, [CCode (array_length_cname = "len", array_length_pos = 2.1, array_length_type = "guint")] uint8[] data);
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_AUDIO_CHANNELS_RANGE")]
public const string CHANNELS_RANGE;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_AUDIO_CONVERTER_OPT_DITHER_METHOD")]
public const string CONVERTER_OPT_DITHER_METHOD;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_AUDIO_CONVERTER_OPT_MIX_MATRIX")]
public const string CONVERTER_OPT_MIX_MATRIX;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD")]
public const string CONVERTER_OPT_NOISE_SHAPING_METHOD;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_AUDIO_CONVERTER_OPT_QUANTIZATION")]
public const string CONVERTER_OPT_QUANTIZATION;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_AUDIO_CONVERTER_OPT_RESAMPLER_METHOD")]
public const string CONVERTER_OPT_RESAMPLER_METHOD;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_AUDIO_DECODER_MAX_ERRORS")]
public const int DECODER_MAX_ERRORS;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_AUDIO_DECODER_SINK_NAME")]
public const string DECODER_SINK_NAME;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_AUDIO_DECODER_SRC_NAME")]
public const string DECODER_SRC_NAME;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_AUDIO_DEF_CHANNELS")]
public const int DEF_CHANNELS;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_AUDIO_DEF_FORMAT")]
public const string DEF_FORMAT;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_AUDIO_DEF_RATE")]
public const int DEF_RATE;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_AUDIO_ENCODER_SINK_NAME")]
public const string ENCODER_SINK_NAME;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_AUDIO_ENCODER_SRC_NAME")]
public const string ENCODER_SRC_NAME;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_AUDIO_FORMATS_ALL")]
public const string FORMATS_ALL;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_META_TAG_AUDIO_CHANNELS_STR")]
[Version (since = "1.2")]
public const string META_TAG_AUDIO_CHANNELS_STR;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_META_TAG_AUDIO_RATE_STR")]
[Version (since = "1.8")]
public const string META_TAG_AUDIO_RATE_STR;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_META_TAG_AUDIO_STR")]
[Version (since = "1.2")]
public const string META_TAG_AUDIO_STR;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_AUDIO_RATE_RANGE")]
public const string RATE_RANGE;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_AUDIO_RESAMPLER_OPT_CUBIC_B")]
public const string RESAMPLER_OPT_CUBIC_B;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_AUDIO_RESAMPLER_OPT_CUBIC_C")]
public const string RESAMPLER_OPT_CUBIC_C;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_AUDIO_RESAMPLER_OPT_CUTOFF")]
public const string RESAMPLER_OPT_CUTOFF;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION")]
public const string RESAMPLER_OPT_FILTER_INTERPOLATION;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_AUDIO_RESAMPLER_OPT_FILTER_MODE")]
public const string RESAMPLER_OPT_FILTER_MODE;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD")]
public const string RESAMPLER_OPT_FILTER_MODE_THRESHOLD;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE")]
public const string RESAMPLER_OPT_FILTER_OVERSAMPLE;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_AUDIO_RESAMPLER_OPT_MAX_PHASE_ERROR")]
public const string RESAMPLER_OPT_MAX_PHASE_ERROR;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_AUDIO_RESAMPLER_OPT_N_TAPS")]
public const string RESAMPLER_OPT_N_TAPS;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUATION")]
public const string RESAMPLER_OPT_STOP_ATTENUATION;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH")]
public const string RESAMPLER_OPT_TRANSITION_BANDWIDTH;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_AUDIO_RESAMPLER_QUALITY_DEFAULT")]
public const int RESAMPLER_QUALITY_DEFAULT;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_AUDIO_RESAMPLER_QUALITY_MAX")]
public const int RESAMPLER_QUALITY_MAX;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "GST_AUDIO_RESAMPLER_QUALITY_MIN")]
public const int RESAMPLER_QUALITY_MIN;
[CCode (cheader_filename = "gst/audio/audio.h", cname = "gst_audio_buffer_clip")]
public static Gst.Buffer audio_buffer_clip (owned Gst.Buffer buffer, Gst.Segment segment, int rate, int bpf);
[CCode (cheader_filename = "gst/audio/audio.h", cname = "gst_audio_buffer_reorder_channels")]
public static bool audio_buffer_reorder_channels (Gst.Buffer buffer, Gst.Audio.Format format, int channels, [CCode (array_length = false)] Gst.Audio.ChannelPosition[] from, [CCode (array_length = false)] Gst.Audio.ChannelPosition[] to);
[CCode (cheader_filename = "gst/audio/audio.h", cname = "gst_audio_channel_get_fallback_mask")]
[Version (since = "1.8")]
public static uint64 audio_channel_get_fallback_mask (int channels);
[CCode (cheader_filename = "gst/audio/audio.h", cname = "gst_audio_channel_positions_from_mask")]
public static bool audio_channel_positions_from_mask (uint64 channel_mask, [CCode (array_length_cname = "channels", array_length_pos = 0.5)] Gst.Audio.ChannelPosition[] position);
[CCode (cheader_filename = "gst/audio/audio.h", cname = "gst_audio_channel_positions_to_mask")]
public static bool audio_channel_positions_to_mask ([CCode (array_length_cname = "channels", array_length_pos = 1.5)] Gst.Audio.ChannelPosition[] position, bool force_order, out uint64 channel_mask);
[CCode (cheader_filename = "gst/audio/audio.h", cname = "gst_audio_channel_positions_to_string")]
public static string audio_channel_positions_to_string ([CCode (array_length_cname = "channels", array_length_pos = 1.1)] Gst.Audio.ChannelPosition[] position);
[CCode (cheader_filename = "gst/audio/audio.h", cname = "gst_audio_channel_positions_to_valid_order")]
public static bool audio_channel_positions_to_valid_order ([CCode (array_length_cname = "channels", array_length_pos = 1.1)] Gst.Audio.ChannelPosition[] position);
[CCode (cheader_filename = "gst/audio/audio.h", cname = "gst_audio_check_valid_channel_positions")]
public static bool audio_check_valid_channel_positions ([CCode (array_length_cname = "channels", array_length_pos = 1.5)] Gst.Audio.ChannelPosition[] position, bool force_order);
[CCode (cheader_filename = "gst/audio/audio.h", cname = "gst_audio_clipping_meta_api_get_type")]
public static GLib.Type audio_clipping_meta_api_get_type ();
[CCode (cheader_filename = "gst/audio/audio.h", cname = "gst_audio_clipping_meta_get_info")]
public static unowned Gst.MetaInfo? audio_clipping_meta_get_info ();
[CCode (cheader_filename = "gst/audio/audio.h", cname = "gst_audio_downmix_meta_api_get_type")]
public static GLib.Type audio_downmix_meta_api_get_type ();
[CCode (cheader_filename = "gst/audio/audio.h", cname = "gst_audio_downmix_meta_get_info")]
public static unowned Gst.MetaInfo? audio_downmix_meta_get_info ();
[CCode (cheader_filename = "gst/audio/audio.h", cname = "gst_audio_format_build_integer")]
public static Gst.Audio.Format audio_format_build_integer (bool sign, int endianness, int width, int depth);
[CCode (cheader_filename = "gst/audio/audio.h", cname = "gst_audio_format_fill_silence")]
public static void audio_format_fill_silence (Gst.Audio.FormatInfo info, [CCode (array_length_cname = "length", array_length_pos = 2.1, array_length_type = "gsize")] uint8[] dest);
[CCode (cheader_filename = "gst/audio/audio.h", cname = "gst_audio_format_from_string")]
public static Gst.Audio.Format audio_format_from_string (string format);
[CCode (cheader_filename = "gst/audio/audio.h", cname = "gst_audio_format_get_info")]
public static unowned Gst.Audio.FormatInfo? audio_format_get_info (Gst.Audio.Format format);
[CCode (cheader_filename = "gst/audio/audio.h", cname = "gst_audio_format_info_get_type")]
public static GLib.Type audio_format_info_get_type ();
[CCode (cheader_filename = "gst/audio/audio.h", cname = "gst_audio_format_to_string")]
public static unowned string audio_format_to_string (Gst.Audio.Format format);
[CCode (cheader_filename = "gst/audio/audio.h", cname = "gst_audio_get_channel_reorder_map")]
public static bool audio_get_channel_reorder_map (int channels, [CCode (array_length = false)] Gst.Audio.ChannelPosition[] from, [CCode (array_length = false)] Gst.Audio.ChannelPosition[] to, [CCode (array_length = false)] int[] reorder_map);
[CCode (cheader_filename = "gst/audio/audio.h", cname = "gst_audio_iec61937_frame_size")]
public static uint audio_iec61937_frame_size (Gst.Audio.RingBufferSpec spec);
[CCode (cheader_filename = "gst/audio/audio.h", cname = "gst_audio_iec61937_payload")]
public static bool audio_iec61937_payload ([CCode (array_length_cname = "src_n", array_length_pos = 1.5, array_length_type = "guint")] uint8[] src, [CCode (array_length_cname = "dst_n", array_length_pos = 2.5, array_length_type = "guint")] uint8[] dst, Gst.Audio.RingBufferSpec spec, int endianness);
[CCode (cheader_filename = "gst/audio/audio.h", cname = "gst_audio_reorder_channels")]
public static bool audio_reorder_channels ([CCode (array_length_cname = "size", array_length_pos = 1.5, array_length_type = "gsize")] uint8[] data, Gst.Audio.Format format, int channels, [CCode (array_length = false)] Gst.Audio.ChannelPosition[] from, [CCode (array_length = false)] Gst.Audio.ChannelPosition[] to);
[CCode (cheader_filename = "gst/audio/audio.h", cname = "gst_audio_resampler_options_set_quality")]
public static void audio_resampler_options_set_quality (Gst.Audio.ResamplerMethod method, uint quality, int in_rate, int out_rate, Gst.Structure options);
[CCode (cheader_filename = "gst/audio/audio.h", cname = "gst_buffer_add_audio_clipping_meta")]
[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);
[CCode (cheader_filename = "gst/audio/audio.h", cname = "gst_buffer_get_audio_downmix_meta_for_channels")]
public static unowned Gst.Audio.DownmixMeta? buffer_get_audio_downmix_meta_for_channels (Gst.Buffer buffer, [CCode (array_length_cname = "to_channels", array_length_pos = 2.1)] Gst.Audio.ChannelPosition[] to_position);
[CCode (cheader_filename = "gst/audio/audio.h", cname = "gst_stream_volume_convert_volume")]
public static double stream_volume_convert_volume (Gst.Audio.StreamVolumeFormat from, Gst.Audio.StreamVolumeFormat to, double val);
}
}
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