gstaudio.types

Undocumented in source.

Members

Aliases

AudioBaseSinkCustomSlavingCallback alias AudioBaseSinkCustomSlavingCallback = void delegate(gstaudio.audio_base_sink.AudioBaseSink sink, gst.types.ClockTime etime, gst.types.ClockTime itime, out gst.types.ClockTimeDiff requestedSkew, gstaudio.types.AudioBaseSinkDiscontReason discontReason): This function is set with gstaudio.audio_base_sink.AudioBaseSink.setCustomSlavingCallback and is called during playback. It receives the current time of external and internal clocks, which the callback can then use to apply any custom slaving/synchronization schemes.
AudioBaseSinkDiscontReason alias AudioBaseSinkDiscontReason = GstAudioBaseSinkDiscontReason
AudioBaseSinkSlaveMethod alias AudioBaseSinkSlaveMethod = GstAudioBaseSinkSlaveMethod
AudioBaseSrcSlaveMethod alias AudioBaseSrcSlaveMethod = GstAudioBaseSrcSlaveMethod
AudioCdSrcMode alias AudioCdSrcMode = GstAudioCdSrcMode
AudioChannelMixerFlags alias AudioChannelMixerFlags = GstAudioChannelMixerFlags
AudioChannelPosition alias AudioChannelPosition = GstAudioChannelPosition
AudioClockGetTimeFunc alias AudioClockGetTimeFunc = gst.types.ClockTime delegate(gst.clock.Clock clock): This function will be called whenever the current clock time needs to be calculated. If this function returns #GST_CLOCK_TIME_NONE, the last reported time will be returned by the clock.
AudioConverterFlags alias AudioConverterFlags = GstAudioConverterFlags
AudioDitherMethod alias AudioDitherMethod = GstAudioDitherMethod
AudioFlags alias AudioFlags = GstAudioFlags
AudioFormat alias AudioFormat = GstAudioFormat
AudioFormatFlags alias AudioFormatFlags = GstAudioFormatFlags
AudioLayout alias AudioLayout = GstAudioLayout
AudioNoiseShapingMethod alias AudioNoiseShapingMethod = GstAudioNoiseShapingMethod
AudioPackFlags alias AudioPackFlags = GstAudioPackFlags
AudioQuantizeFlags alias AudioQuantizeFlags = GstAudioQuantizeFlags
AudioResamplerFilterInterpolation alias AudioResamplerFilterInterpolation = GstAudioResamplerFilterInterpolation
AudioResamplerFilterMode alias AudioResamplerFilterMode = GstAudioResamplerFilterMode
AudioResamplerFlags alias AudioResamplerFlags = GstAudioResamplerFlags
AudioResamplerMethod alias AudioResamplerMethod = GstAudioResamplerMethod
AudioRingBufferCallback alias AudioRingBufferCallback = void delegate(gstaudio.audio_ring_buffer.AudioRingBuffer rbuf, ubyte[] data): This function is set with gstaudio.audio_ring_buffer.AudioRingBuffer.setCallback and is called to fill the memory at data with len bytes of samples.
AudioRingBufferFormatType alias AudioRingBufferFormatType = GstAudioRingBufferFormatType
AudioRingBufferState alias AudioRingBufferState = GstAudioRingBufferState
AudioSinkClassExtension alias AudioSinkClassExtension = GstAudioSinkClassExtension
DsdFormat alias DsdFormat = GstDsdFormat
StreamVolumeFormat alias StreamVolumeFormat = GstStreamVolumeFormat

Manifest constants

AUDIO_CHANNELS_RANGE enum AUDIO_CHANNELS_RANGE;: Maximum range of allowed channels, for use in template caps strings.
AUDIO_CONVERTER_OPT_DITHER_METHOD enum AUDIO_CONVERTER_OPT_DITHER_METHOD;: #GstAudioDitherMethod, The dither method to use when changing bit depth. Default is #GST_AUDIO_DITHER_NONE.
AUDIO_CONVERTER_OPT_DITHER_THRESHOLD enum AUDIO_CONVERTER_OPT_DITHER_THRESHOLD;: Threshold for the output bit depth at/below which to apply dithering.
AUDIO_CONVERTER_OPT_MIX_MATRIX enum AUDIO_CONVERTER_OPT_MIX_MATRIX;: #GST_TYPE_LIST, The channel mapping matrix.
AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD enum AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD;: #GstAudioNoiseShapingMethod, The noise shaping method to use to mask noise from quantization errors. Default is #GST_AUDIO_NOISE_SHAPING_NONE.
AUDIO_CONVERTER_OPT_QUANTIZATION enum AUDIO_CONVERTER_OPT_QUANTIZATION;: #G_TYPE_UINT, The quantization amount. Components will be quantized to multiples of this value. Default is 1
AUDIO_CONVERTER_OPT_RESAMPLER_METHOD enum AUDIO_CONVERTER_OPT_RESAMPLER_METHOD;: #GstAudioResamplerMethod, The resampler method to use when changing sample rates. Default is #GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL.
AUDIO_DECODER_MAX_ERRORS enum AUDIO_DECODER_MAX_ERRORS;: Default maximum number of errors tolerated before signaling error.
AUDIO_DECODER_SINK_NAME enum AUDIO_DECODER_SINK_NAME;: The name of the templates for the sink pad.
AUDIO_DECODER_SRC_NAME enum AUDIO_DECODER_SRC_NAME;: The name of the templates for the source pad.
AUDIO_DEF_CHANNELS enum AUDIO_DEF_CHANNELS;: Standard number of channels used in consumer audio.
AUDIO_DEF_FORMAT enum AUDIO_DEF_FORMAT;: Standard format used in consumer audio.
AUDIO_DEF_RATE enum AUDIO_DEF_RATE;: Standard sampling rate used in consumer audio.
AUDIO_ENCODER_SINK_NAME enum AUDIO_ENCODER_SINK_NAME;: the name of the templates for the sink pad
AUDIO_ENCODER_SRC_NAME enum AUDIO_ENCODER_SRC_NAME;: the name of the templates for the source pad
AUDIO_FORMATS_ALL enum AUDIO_FORMATS_ALL;: List of all audio formats, for use in template caps strings.
AUDIO_RATE_RANGE enum AUDIO_RATE_RANGE;: Maximum range of allowed sample rates, for use in template caps strings.
AUDIO_RESAMPLER_OPT_CUBIC_B enum AUDIO_RESAMPLER_OPT_CUBIC_B;: G_TYPE_DOUBLE, B parameter of the cubic filter. Values between 0.0 and 2.0 are accepted. 1.0 is the default.
AUDIO_RESAMPLER_OPT_CUBIC_C enum AUDIO_RESAMPLER_OPT_CUBIC_C;: G_TYPE_DOUBLE, C parameter of the cubic filter. Values between 0.0 and 2.0 are accepted. 0.0 is the default.
AUDIO_RESAMPLER_OPT_CUTOFF enum AUDIO_RESAMPLER_OPT_CUTOFF;: G_TYPE_DOUBLE, Cutoff parameter for the filter. 0.940 is the default.
AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION enum AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION;
AUDIO_RESAMPLER_OPT_FILTER_MODE enum AUDIO_RESAMPLER_OPT_FILTER_MODE;
AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD enum AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD;
AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE enum AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE;: G_TYPE_UINT, oversampling to use when interpolating filters 8 is the default.
AUDIO_RESAMPLER_OPT_MAX_PHASE_ERROR enum AUDIO_RESAMPLER_OPT_MAX_PHASE_ERROR;
AUDIO_RESAMPLER_OPT_N_TAPS enum AUDIO_RESAMPLER_OPT_N_TAPS;
AUDIO_RESAMPLER_OPT_STOP_ATTENUATION enum AUDIO_RESAMPLER_OPT_STOP_ATTENUATION;: G_TYPE_DOUBLE, stopband attenuation in decibels. The attenuation after the stopband for the kaiser window. 85 dB is the default.
AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH enum AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH;: G_TYPE_DOUBLE, transition bandwidth. The width of the transition band for the kaiser window. 0.087 is the default.
AUDIO_RESAMPLER_QUALITY_DEFAULT enum AUDIO_RESAMPLER_QUALITY_DEFAULT;
AUDIO_RESAMPLER_QUALITY_MAX enum AUDIO_RESAMPLER_QUALITY_MAX;
AUDIO_RESAMPLER_QUALITY_MIN enum AUDIO_RESAMPLER_QUALITY_MIN;
DSD_FORMATS_ALL enum DSD_FORMATS_ALL;: List of all DSD formats, for use in template caps strings.
DSD_MEDIA_TYPE enum DSD_MEDIA_TYPE;: The GStreamer media type for DSD.
DSD_SILENCE_PATTERN_BYTE enum DSD_SILENCE_PATTERN_BYTE;: Silence pattern for DSD data.
META_TAG_AUDIO_CHANNELS_STR enum META_TAG_AUDIO_CHANNELS_STR;: This metadata stays relevant as long as channels are unchanged.
META_TAG_AUDIO_RATE_STR enum META_TAG_AUDIO_RATE_STR;: This metadata stays relevant as long as sample rate is unchanged.
META_TAG_AUDIO_STR enum META_TAG_AUDIO_STR;: This metadata is relevant for audio streams.
META_TAG_DSD_PLANE_OFFSETS_STR enum META_TAG_DSD_PLANE_OFFSETS_STR;: This metadata stays relevant as long as the DSD plane offsets are unchanged.