gstreamer-audio-1.0

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Description:

GStreamer Audio Library

• Home: https://gstreamer.freedesktop.org/
• C-Documentation: https://gstreamer.freedesktop.org/documentation/audio/index.html
• Devhelp-Package download

Content:

Namespaces:

• Gst
• Audio

Interfaces:

• StreamVolume - This interface is implemented by elements that provide a stream volume.

Classes:

• Aggregator - Subclasses must use (a subclass of) AggregatorPad for both their source and sink pads, add_static_pad_template_with_gtype is a convenient helper.
• AggregatorConvertPad - An implementation of GstPad that can be used with Aggregator .
• AggregatorPad - The default implementation of GstPad used with Aggregator
• BaseSink - This is the base class for audio sinks.
• BaseSrc - This is the base class for audio sources.
• CdSrc - Provides a base class for CD digital audio (CDDA) sources, which handles things like seeking, querying, discid calculation, tags, and buffer timestamping.
• ChannelMixer
• Clock - Clock makes it easy for elements to implement a Clock, they simply need to provide a function that returns the current clock time.
• Converter - This object is used to convert audio samples from one format to another.
• Decoder - This base class is for audio decoders turning encoded data into raw audio samples.
• Encoder - This base class is for audio encoders turning raw audio samples into encoded audio data.
• Filter - Filter is a Transform <!-- -->-derived base class for simple audio filters, ie.
• Info - Information describing audio properties.
• Quantize
• Resampler - Resampler is a structure which holds the information required to perform various kinds of resampling filtering.
• RingBuffer - This object is the base class for audio ringbuffers used by the base audio source and sink classes.
• Sink - This is the most simple base class for audio sinks that only requires subclasses to implement a set of simple functions:
• Src - This is the most simple base class for audio sources that only requires subclasses to implement a set of simple functions:
• StreamAlign - StreamAlign provides a helper object that helps tracking audio stream alignment and discontinuities, and detects discontinuities if possible.

Structs:

• Buffer - A structure containing the result of an audio buffer map operation, which is executed with map .
• CdSrcTrack - CD track abstraction to communicate TOC entries to the base class.
• ClippingMeta - Extra buffer metadata describing how much audio has to be clipped from the start or end of a buffer.
• DownmixMeta - Extra buffer metadata describing audio downmixing matrix.
• FormatInfo - Information for an audio format.
• LevelMeta - Meta containing Audio Level Indication: https://tools.
• Meta - DownmixMeta defines an audio downmix matrix to be send along with audio buffers.
• RingBufferSpec - The structure containing the format specification of the ringbuffer.
• SinkClassExtension

Enums:

• BaseSinkDiscontReason - Different possible reasons for discontinuities.
• BaseSinkSlaveMethod - Different possible clock slaving algorithms used when the internal audio clock is not selected as the pipeline master clock.
• BaseSrcSlaveMethod - Different possible clock slaving algorithms when the internal audio clock was not selected as the pipeline clock.
• CdSrcMode - Mode in which the CD audio source operates.
• ChannelMixerFlags - Flags passed to gst_audio_channel_mixer_new
• ChannelPosition - Audio channel positions.
• ConverterFlags - Extra flags passed to Converter and samples.
• DitherMethod - Set of available dithering methods.
• Flags - Extra audio flags
• Format - Enum value describing the most common audio formats.
• FormatFlags - The different audio flags that a format info can have.
• Layout - Layout of the audio samples for the different channels.
• NoiseShapingMethod - Set of available noise shaping methods
• PackFlags - The different flags that can be used when packing and unpacking.
• QuantizeFlags - Extra flags that can be passed to gst_audio_quantize_new
• ResamplerFilterInterpolation - The different filter interpolation methods.
• ResamplerFilterMode - Select for the filter tables should be set up.
• ResamplerFlags - Different resampler flags.
• ResamplerMethod - Different subsampling and upsampling methods
• RingBufferFormatType - The format of the samples in the ringbuffer.
• RingBufferState - The state of the ringbuffer.
• StreamVolumeFormat - Different representations of a stream volume.

Constants:

• public const string CHANNELS_RANGE
  Maximum range of allowed channels, for use in template caps strings.
• public const string CONVERTER_OPT_DITHER_METHOD
  DitherMethod, The dither method to use when changing bit depth.
• public const string CONVERTER_OPT_DITHER_THRESHOLD
  Threshold for the output bit depth at/below which to apply dithering.
• public const string CONVERTER_OPT_MIX_MATRIX
  GST_TYPE_LIST, The channel mapping matrix.
• public const string CONVERTER_OPT_NOISE_SHAPING_METHOD
  NoiseShapingMethod, The noise shaping method to use to mask noise from quantization errors.
• public const string CONVERTER_OPT_QUANTIZATION
  UINT, The quantization amount.
• public const string CONVERTER_OPT_RESAMPLER_METHOD
  ResamplerMethod, The resampler method to use when changing sample rates.
• public const int DECODER_MAX_ERRORS
  Default maximum number of errors tolerated before signaling error.
• public const string DECODER_SINK_NAME
  The name of the templates for the sink pad.
• public const string DECODER_SRC_NAME
  The name of the templates for the source pad.
• public const int DEF_CHANNELS
  Standard number of channels used in consumer audio.
• public const string DEF_FORMAT
  Standard format used in consumer audio.
• public const int DEF_RATE
  Standard sampling rate used in consumer audio.
• public const string ENCODER_SINK_NAME
  the name of the templates for the sink pad
• public const string ENCODER_SRC_NAME
  the name of the templates for the source pad
• public const string FORMATS_ALL
  List of all audio formats, for use in template caps strings.
• public const string META_TAG_AUDIO_CHANNELS_STR
  This metadata stays relevant as long as channels are unchanged.
• public const string META_TAG_AUDIO_RATE_STR
  This metadata stays relevant as long as sample rate is unchanged.
• public const string META_TAG_AUDIO_STR
  This metadata is relevant for audio streams.
• public const string RATE_RANGE
  Maximum range of allowed sample rates, for use in template caps strings.
• public const string RESAMPLER_OPT_CUBIC_B
  G_TYPE_DOUBLE, B parameter of the cubic filter.
• public const string RESAMPLER_OPT_CUBIC_C
  G_TYPE_DOUBLE, C parameter of the cubic filter.
• public const string RESAMPLER_OPT_CUTOFF
  G_TYPE_DOUBLE, Cutoff parameter for the filter.
• public const string RESAMPLER_OPT_FILTER_INTERPOLATION
  GST_TYPE_AUDIO_RESAMPLER_INTERPOLATION: how the filter coefficients should be interpolated.
• public const string RESAMPLER_OPT_FILTER_MODE
  GST_TYPE_AUDIO_RESAMPLER_FILTER_MODE: how the filter tables should be constructed.
• public const string RESAMPLER_OPT_FILTER_MODE_THRESHOLD
  G_TYPE_UINT: the amount of memory to use for full filter tables before switching to interpolated filter tables.
• public const string RESAMPLER_OPT_FILTER_OVERSAMPLE
  G_TYPE_UINT, oversampling to use when interpolating filters 8 is the default.
• public const string RESAMPLER_OPT_MAX_PHASE_ERROR
  G_TYPE_DOUBLE: The maximum allowed phase error when switching sample rates.
• public const string RESAMPLER_OPT_N_TAPS
  G_TYPE_INT: the number of taps to use for the filter.
• public const string RESAMPLER_OPT_STOP_ATTENUATION
  G_TYPE_DOUBLE, stopband attenuation in decibels.
• public const string RESAMPLER_OPT_TRANSITION_BANDWIDTH
  G_TYPE_DOUBLE, transition bandwidth.
• public const int RESAMPLER_QUALITY_DEFAULT
• public const int RESAMPLER_QUALITY_MAX
• public const int RESAMPLER_QUALITY_MIN

Delegates:

• public delegate void BaseSinkCustomSlavingCallback (BaseSink sink, ClockTime etime, ClockTime itime, out ClockTimeDiff requested_skew, BaseSinkDiscontReason discont_reason)
  This function is set with set_custom_slaving_callback and is called during playback.
• public delegate ClockTime ClockGetTimeFunc (Clock clock)
  This function will be called whenever the current clock time needs to be calculated.
• public delegate void FormatPack (FormatInfo info, PackFlags flags, uint8[] src, uint8[] data, int length)
  Packs length samples from src to the data array in format info.
• public delegate void FormatUnpack (FormatInfo info, PackFlags flags, uint8[] dest, uint8[] data, int length)
  Unpacks length samples from the given data of format info.
• public delegate void RingBufferCallback (RingBuffer rbuf, uint8[] data)
  This function is set with gst_audio_ring_buffer_set_callback and is called to fill the memory at data with data.length bytes of samples.

Methods:

• public Buffer? audio_buffer_clip (owned Buffer buffer, Segment segment, int rate, int bpf)
• public bool audio_buffer_map (out Buffer buffer, Info info, Buffer gstbuffer, MapFlags flags)
• public bool audio_buffer_reorder_channels (Buffer buffer, Format format, ChannelPosition[] from, ChannelPosition[] to)
• public Buffer audio_buffer_truncate (owned Buffer buffer, int bpf, size_t trim, size_t samples)
• public uint64 audio_channel_get_fallback_mask (int channels)
  Get the fallback channel-mask for the given number of channels.
• public bool audio_channel_positions_from_mask (uint64 channel_mask, ChannelPosition[] position)
  Convert the position.length present in channel_mask to a position array (which should have at least position.length entries ensured by caller).
• public bool audio_channel_positions_to_mask (ChannelPosition[] position, bool force_order, out uint64 channel_mask)
  Convert the position array of position.length channels to a bitmask.
• public string audio_channel_positions_to_string (ChannelPosition[] position)
  Converts position to a human-readable string representation for debugging purposes.
• public bool audio_channel_positions_to_valid_order (ChannelPosition[] position)
  Reorders the channel positions in position from any order to the GStreamer channel order.
• public bool audio_check_valid_channel_positions (ChannelPosition[] position, bool force_order)
  Checks if position contains valid channel positions for position.length channels.
• public Type audio_clipping_meta_api_get_type ()
• public unowned MetaInfo? audio_clipping_meta_get_info ()
• public Type audio_downmix_meta_api_get_type ()
• public unowned MetaInfo? audio_downmix_meta_get_info ()
• public Format audio_format_build_integer (bool sign, int endianness, int width, int depth)
• public void audio_format_fill_silence (FormatInfo info, uint8[] dest)
• public Format audio_format_from_string (string format)
• public unowned FormatInfo? audio_format_get_info (Format format)
• public Type audio_format_info_get_type ()
• public unowned string audio_format_to_string (Format format)
• public unowned Format[] audio_formats_raw ()
  Return all the raw audio formats supported by GStreamer.
• public bool audio_get_channel_reorder_map (ChannelPosition[] from, ChannelPosition[] to, int[] reorder_map)
  Returns a reorder map for from to to that can be used in custom channel reordering code, e.
• public uint audio_iec61937_frame_size (RingBufferSpec spec)
  Calculated the size of the buffer expected by audio_iec61937_payload for payloading type from spec.
• public bool audio_iec61937_payload (uint8[] src, uint8[] dst, RingBufferSpec spec, int endianness)
  Payloads src in the form specified by IEC 61937 for the type from spec and stores the result in dst.
• public bool audio_info_from_caps (out unowned Info info, Caps caps)
  Parse caps and update info.
• public void audio_info_init (out unowned Info info)
  Initialize info with default values.
• public Type audio_level_meta_api_get_type ()
  Return the Type associated with LevelMeta.
• public unowned MetaInfo? audio_level_meta_get_info ()
• public Caps audio_make_raw_caps (Format[]? formats, Layout layout)
  Return a generic raw audio caps for formats defined in formats .
• public Type audio_meta_api_get_type ()
• public unowned MetaInfo? audio_meta_get_info ()
• public bool audio_reorder_channels (uint8[] data, Format format, ChannelPosition[] from, ChannelPosition[] to)
  Reorders data from the channel positions from to the channel positions to.
• public void audio_resampler_options_set_quality (ResamplerMethod method, uint quality, int in_rate, int out_rate, Structure options)
  Set the parameters for resampling from in_rate to out_rate using method for quality in options.
• public unowned ClippingMeta? buffer_add_audio_clipping_meta (Buffer buffer, Format format, uint64 start, uint64 end)
  Attaches ClippingMeta metadata to buffer with the given parameters.
• public unowned DownmixMeta? buffer_add_audio_downmix_meta (Buffer buffer, ChannelPosition[] from_position, ChannelPosition[] to_position, float*[] matrix)
  Attaches DownmixMeta metadata to buffer with the given parameters.
• public unowned LevelMeta? buffer_add_audio_level_meta (Buffer buffer, uint8 level, bool voice_activity)
  Attaches audio level information to buffer.
• public unowned Meta? buffer_add_audio_meta (Buffer buffer, Info info, size_t samples, size_t? offsets)
  Allocates and attaches a Meta on buffer, which must be writable for that purpose.
• public unowned DownmixMeta? buffer_get_audio_downmix_meta_for_channels (Buffer buffer, ChannelPosition[] to_position)
  Find the DownmixMeta on buffer for the given destination channel positions.
• public unowned LevelMeta? buffer_get_audio_level_meta (Buffer buffer)
  Find the LevelMeta on buffer.
• public double stream_volume_convert_volume (StreamVolumeFormat from, StreamVolumeFormat to, double val)