Different possible reasons for discontinuities. This enum is useful for the custom slave method.
Different possible clock slaving algorithms used when the internal audio clock is not selected as the pipeline master clock.
Different possible clock slaving algorithms when the internal audio clock was not selected as the pipeline clock.
Mode in which the CD audio source operates. Influences timestamping, EOS handling and seeking.
Flags passed to gstaudio.audio_channel_mixer.AudioChannelMixer.new_
Audio channel positions.
Extra flags passed to gstaudio.audio_converter.AudioConverter.new_ and gstaudio.audio_converter.AudioConverter.samples.
Set of available dithering methods.
Extra audio flags
Enum value describing the most common audio formats.
The different audio flags that a format info can have.
Layout of the audio samples for the different channels.
Set of available noise shaping methods
The different flags that can be used when packing and unpacking.
Extra flags that can be passed to gstaudio.audio_quantize.AudioQuantize.new_
The different filter interpolation methods.
Select for the filter tables should be set up.
Different resampler flags.
Different subsampling and upsampling methods
The format of the samples in the ringbuffer.
The state of the ringbuffer.
Enum value describing how DSD bits are grouped.
Different representations of a stream volume. gstaudio.stream_volume.StreamVolume.convertVolume allows to convert between the different representations.
Subclasses must use (a subclass of) #GstAudioAggregatorPad for both their source and sink pads, gst.element_class.ElementClass.addStaticPadTemplateWithGtype is a convenient helper.
An implementation of GstPad that can be used with #GstAudioAggregator.
The default implementation of GstPad used with #GstAudioAggregator
This is the base class for audio sinks. Subclasses need to implement the ::create_ringbuffer vmethod. This base class will then take care of writing samples to the ringbuffer, synchronisation, clipping and flushing.
#GstAudioBaseSink class. Override the vmethod to implement functionality.
This is the base class for audio sources. Subclasses need to implement the ::create_ringbuffer vmethod. This base class will then take care of reading samples from the ringbuffer, synchronisation and flushing.
#GstAudioBaseSrc class. Override the vmethod to implement functionality.
A structure containing the result of an audio buffer map operation, which is executed with gstaudio.audio_buffer.AudioBuffer.map. For non-interleaved (planar) buffers, the beginning of each channel in the buffer has its own pointer in the @planes array. For interleaved buffers, the @planes array only contains one item, which is the pointer to the beginning of the buffer, and @n_planes equals 1.
Provides a base class for CD digital audio (CDDA) sources, which handles things like seeking, querying, discid calculation, tags, and buffer timestamping.
Audio CD source base class.
CD track abstraction to communicate TOC entries to the base class.
Extra buffer metadata describing how much audio has to be clipped from the start or end of a buffer. This is used for compressed formats, where the first frame usually has some additional samples due to encoder and decoder delays, and the last frame usually has some additional samples to be able to fill the complete last frame.
#GstAudioClock makes it easy for elements to implement a #GstClock, they simply need to provide a function that returns the current clock time.
This object is used to convert audio samples from one format to another. The object can perform conversion of:
This base class is for audio decoders turning encoded data into raw audio samples.
Subclasses can override any of the available virtual methods or not, as needed. At minimum @handle_frame (and likely @set_format) needs to be overridden.
Extra buffer metadata describing audio downmixing matrix. This metadata is attached to audio buffers and contains a matrix to downmix the buffer number of channels to @channels.
This base class is for audio encoders turning raw audio samples into encoded audio data.
Subclasses can override any of the available virtual methods or not, as needed. At minimum @set_format and @handle_frame needs to be overridden.
#GstAudioFilter is a #GstBaseTransform<!-- -->-derived base class for simple audio filters, ie. those that output the same format that they get as input.
In addition to the @setup virtual function, you should also override the GstBaseTransform::transform and/or GstBaseTransform::transform_ip virtual function.
Information for an audio format.
Information describing audio properties. This information can be filled in from GstCaps with gstaudio.audio_info.AudioInfo.fromCaps.
Meta containing Audio Level Indication: https://tools.ietf.org/html/rfc6464
#GstAudioDownmixMeta defines an audio downmix matrix to be send along with audio buffers. These functions in this module help to create and attach the meta as well as extracting it.
#GstAudioResampler is a structure which holds the information required to perform various kinds of resampling filtering.
This object is the base class for audio ringbuffers used by the base audio source and sink classes.
The vmethods that subclasses can override to implement the ringbuffer.
The structure containing the format specification of the ringbuffer.
This is the most simple base class for audio sinks that only requires subclasses to implement a set of simple functions:
This is the most simple base class for audio sources that only requires subclasses to implement a set of simple functions:
#GstAudioSrc class. Override the vmethod to implement functionality.
#GstAudioStreamAlign provides a helper object that helps tracking audio stream alignment and discontinuities, and detects discontinuities if possible.
Information describing DSD audio properties.
Buffer metadata describing planar DSD contents in the buffer. This is not needed for interleaved DSD data, and is required for non-interleaved (= planar) data.
This interface is implemented by elements that provide a stream volume. Examples for such elements are #volume and #playbin.