gstaudio.types

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.

The external clock is the sink's element clock, the internal one is the internal audio clock. The internal audio clock's calibration is applied to the timestamps before they are passed to the callback. The difference between etime and itime is the skew; how much internal and external clock lie apart from each other. A skew of 0 means both clocks are perfectly in sync. itime > etime means the external clock is going slower, while itime < etime means it is going faster than the internal clock. etime and itime are always valid timestamps, except for when a discontinuity happens.

requested_skew is an output value the callback can write to. It informs the sink of whether or not it should move the playout pointer, and if so, by how much. This pointer is only NULL if a discontinuity occurs; otherwise, it is safe to write to *requested_skew. The default skew is 0.

The sink may experience discontinuities. If one happens, discont is TRUE, itime, etime are set to GST_CLOCK_TIME_NONE, and requested_skew is NULL. This makes it possible to reset custom clock slaving algorithms when a discontinuity happens.

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.

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.

Maximum range of allowed channels, for use in template caps strings.

#GstAudioDitherMethod, The dither method to use when changing bit depth. Default is #GST_AUDIO_DITHER_NONE.

Threshold for the output bit depth at/below which to apply dithering.

Default is 20 bit.

#GST_TYPE_LIST, The channel mapping matrix.

The matrix coefficients must be between -1 and 1: the number of rows is equal to the number of output channels and the number of columns is equal to the number of input channels.

Example matrix generation code

To generate the matrix using code:

GValue v = G_VALUE_INIT;
GValue v2 = G_VALUE_INIT;
GValue v3 = G_VALUE_INIT;

g_value_init (&v2, GST_TYPE_ARRAY);
g_value_init (&v3, G_TYPE_DOUBLE);
g_value_set_double (&v3, 1);
gst_value_array_append_value (&v2, &v3);
g_value_unset (&v3);
[ Repeat for as many double as your input channels - unset and reinit v3 ]
g_value_init (&v, GST_TYPE_ARRAY);
gst_value_array_append_value (&v, &v2);
g_value_unset (&v2);
[ Repeat for as many v2's as your output channels - unset and reinit v2]
g_object_set_property (G_OBJECT (audiomixmatrix), "matrix", &v);
g_value_unset (&v);

#GstAudioNoiseShapingMethod, The noise shaping method to use to mask noise from quantization errors. Default is #GST_AUDIO_NOISE_SHAPING_NONE.

#G_TYPE_UINT, The quantization amount. Components will be quantized to multiples of this value. Default is 1

#GstAudioResamplerMethod, The resampler method to use when changing sample rates. Default is #GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL.

Default maximum number of errors tolerated before signaling error.

The name of the templates for the sink pad.

The name of the templates for the source pad.

Standard number of channels used in consumer audio.

Standard format used in consumer audio.

Standard sampling rate used in consumer audio.

the name of the templates for the sink pad

the name of the templates for the source pad

List of all audio formats, for use in template caps strings.

Formats are sorted by decreasing "quality", using these criteria by priority:

• depth
• width
• Float > Signed > Unsigned
• native endianness preferred

Maximum range of allowed sample rates, for use in template caps strings.

G_TYPE_DOUBLE, B parameter of the cubic filter. Values between 0.0 and 2.0 are accepted. 1.0 is the default.

Below are some values of popular filters: B C Hermite 0.0 0.0 Spline 1.0 0.0 Catmull-Rom 0.0 1/2

G_TYPE_DOUBLE, C parameter of the cubic filter. Values between 0.0 and 2.0 are accepted. 0.0 is the default.

See #GST_AUDIO_RESAMPLER_OPT_CUBIC_B for some more common values

G_TYPE_DOUBLE, Cutoff parameter for the filter. 0.940 is the default.

GST_TYPE_AUDIO_RESAMPLER_INTERPOLATION: how the filter coefficients should be interpolated. GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC is default.

GST_TYPE_AUDIO_RESAMPLER_FILTER_MODE: how the filter tables should be constructed. GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO is the default.

G_TYPE_UINT: the amount of memory to use for full filter tables before switching to interpolated filter tables. 1048576 is the default.

G_TYPE_UINT, oversampling to use when interpolating filters 8 is the default.

G_TYPE_DOUBLE: The maximum allowed phase error when switching sample rates. 0.1 is the default.

G_TYPE_INT: the number of taps to use for the filter. 0 is the default and selects the taps automatically.

G_TYPE_DOUBLE, stopband attenuation in decibels. The attenuation after the stopband for the kaiser window. 85 dB is the default.

G_TYPE_DOUBLE, transition bandwidth. The width of the transition band for the kaiser window. 0.087 is the default.

List of all DSD formats, for use in template caps strings.

Big endian formats are preferred, since little-endian ones flip around the DSD bytes, and most DSD hardware uses big endian formats.

The GStreamer media type for DSD.

Silence pattern for DSD data.

In DSD, a nullbyte does not correspond to silence. To fill memory regions with "DSD silence", these regions must be filled with byte 0x69 instead (this is the DSD silence pattern). This constant provides that pattern in a more readable fashion.

This metadata stays relevant as long as channels are unchanged.

This metadata stays relevant as long as sample rate is unchanged.

This metadata is relevant for audio streams.

This metadata stays relevant as long as the DSD plane offsets are unchanged.