Enums¶
Details¶
- class GstAudio.AudioBaseSinkDiscontReason(value)¶
Bases:
GObject.GEnum
Different possible reasons for discontinuities. This enum is useful for the custom slave method.
New in version 1.6.
- NO_DISCONT = 0¶
No discontinuity occurred
- NEW_CAPS = 1¶
New caps are set, causing renegotiotion
- FLUSH = 2¶
Samples have been flushed
- SYNC_LATENCY = 3¶
Sink was synchronized to the estimated latency (occurs during initialization)
- ALIGNMENT = 4¶
Aligning buffers failed because the timestamps are too discontinuous
- DEVICE_FAILURE = 5¶
Audio output device experienced and recovered from an error but introduced latency in the process (see also
GstAudio.AudioBaseSink.report_device_failure
())
- class GstAudio.AudioBaseSinkSlaveMethod(value)¶
Bases:
GObject.GEnum
Different possible clock slaving algorithms used when the internal audio clock is not selected as the pipeline master clock.
- RESAMPLE = 0¶
Resample to match the master clock
- SKEW = 1¶
Adjust playout pointer when master clock drifts too much.
- NONE = 2¶
No adjustment is done.
- CUSTOM = 3¶
Use custom clock slaving algorithm
New in version 1.6.
- class GstAudio.AudioBaseSrcSlaveMethod(value)¶
Bases:
GObject.GEnum
Different possible clock slaving algorithms when the internal audio clock was not selected as the pipeline clock.
- RESAMPLE = 0¶
Resample to match the master clock.
- RE_TIMESTAMP = 1¶
Retimestamp output buffers with master clock time.
- SKEW = 2¶
Adjust capture pointer when master clock drifts too much.
- NONE = 3¶
No adjustment is done.
- class GstAudio.AudioCdSrcMode(value)¶
Bases:
GObject.GEnum
Mode in which the CD audio source operates. Influences timestamping, EOS handling and seeking.
- NORMAL = 0¶
each single track is a stream
- CONTINUOUS = 1¶
the entire disc is a single stream
- class GstAudio.AudioChannelPosition(value)¶
Bases:
GObject.GEnum
Audio channel positions.
These are the channels defined in SMPTE 2036-2-2008 Table 1 for 22.2 audio systems with the Surround and Wide channels from DTS Coherent Acoustics (v.1.3.1) and 10.2 and 7.1 layouts. In the caps the actual channel layout is expressed with a channel count and a channel mask, which describes the existing channels. The positions in the bit mask correspond to the enum values. For negotiation it is allowed to have more bits set in the channel mask than the number of channels to specify the allowed channel positions but this is not allowed in negotiated caps. It is not allowed in any situation other than the one mentioned below to have less bits set in the channel mask than the number of channels.
GstAudio.AudioChannelPosition.MONO
can only be used with a single mono channel that has no direction information and would be mixed into all directional channels. This is expressed in caps by having a single channel and no channel mask.GstAudio.AudioChannelPosition.NONE
can only be used if all channels have this position. This is expressed in caps by having a channel mask with no bits set.As another special case it is allowed to have two channels without a channel mask. This implicitly means that this is a stereo stream with a front left and front right channel.
- INVALID = -1¶
invalid position
- MONO = -2¶
Mono without direction; can only be used with 1 channel
- NONE = -3¶
used for position-less channels, e.g. from a sound card that records 1024 channels; mutually exclusive with any other channel position
- FRONT_LEFT = 0¶
Front left
- FRONT_RIGHT = 1¶
Front right
- SIDE_LEFT = 10¶
Side left
- SIDE_RIGHT = 11¶
Side right
- TOP_FRONT_LEFT = 12¶
Top front left
- TOP_FRONT_RIGHT = 13¶
Top front right
- TOP_FRONT_CENTER = 14¶
Top front center
- TOP_CENTER = 15¶
Top center
- TOP_REAR_LEFT = 16¶
Top rear left
- TOP_REAR_RIGHT = 17¶
Top rear right
- TOP_SIDE_LEFT = 18¶
Top side right
- TOP_SIDE_RIGHT = 19¶
Top rear right
- FRONT_CENTER = 2¶
Front center
- TOP_REAR_CENTER = 20¶
Top rear center
- BOTTOM_FRONT_CENTER = 21¶
Bottom front center
- BOTTOM_FRONT_LEFT = 22¶
Bottom front left
- BOTTOM_FRONT_RIGHT = 23¶
Bottom front right
- WIDE_LEFT = 24¶
Wide left (between front left and side left)
- WIDE_RIGHT = 25¶
Wide right (between front right and side right)
- SURROUND_LEFT = 26¶
Surround left (between rear left and side left)
- SURROUND_RIGHT = 27¶
Surround right (between rear right and side right)
- LFE1 = 3¶
Low-frequency effects 1 (subwoofer)
- REAR_LEFT = 4¶
Rear left
- REAR_RIGHT = 5¶
Rear right
- FRONT_LEFT_OF_CENTER = 6¶
Front left of center
- FRONT_RIGHT_OF_CENTER = 7¶
Front right of center
- REAR_CENTER = 8¶
Rear center
- LFE2 = 9¶
Low-frequency effects 2 (subwoofer)
- class GstAudio.AudioDitherMethod(value)¶
Bases:
GObject.GEnum
Set of available dithering methods.
- NONE = 0¶
No dithering
- RPDF = 1¶
Rectangular dithering
- TPDF = 2¶
Triangular dithering (default)
- TPDF_HF = 3¶
High frequency triangular dithering
- class GstAudio.AudioFormat(value)¶
Bases:
GObject.GEnum
Enum value describing the most common audio formats.
- classmethod build_integer(sign, endianness, width, depth)[source]¶
- Parameters:
sign (
bool
) – signed or unsigned formatendianness (
int
) –GLib.LITTLE_ENDIAN
orGLib.BIG_ENDIAN
width (
int
) – amount of bits used per sampledepth (
int
) – amount of used bits in width
- Returns:
a
GstAudio.AudioFormat
orGstAudio.AudioFormat.UNKNOWN
when no audio format exists with the given parameters.- Return type:
Construct a
GstAudio.AudioFormat
with given parameters.
- classmethod fill_silence(info, dest)[source]¶
- Parameters:
info (
GstAudio.AudioFormatInfo
) – aGstAudio.AudioFormatInfo
dest (
bytes
) – a destination to fill
Fill length bytes in dest with silence samples for info.
Deprecated since version 1.20: Use
GstAudio.AudioFormatInfo.fill_silence
() instead.
- classmethod from_string(format)[source]¶
- Parameters:
format (
str
) – a format string- Returns:
the
GstAudio.AudioFormat
for format orGstAudio.AudioFormat.UNKNOWN
when the string is not a known format.- Return type:
Convert the format string to its
GstAudio.AudioFormat
.
- classmethod get_info(format)[source]¶
- Parameters:
format (
GstAudio.AudioFormat
) – aGstAudio.AudioFormat
- Returns:
The
GstAudio.AudioFormatInfo
for format.- Return type:
Get the
GstAudio.AudioFormatInfo
for format
- classmethod to_string(format)[source]¶
- Parameters:
format (
GstAudio.AudioFormat
) –- Return type:
- UNKNOWN = 0¶
unknown or unset audio format
- ENCODED = 1¶
encoded audio format
- U24_32 = 10¶
24 bits in 32 bits, unsigned, native endianness
- U24_32LE = 10¶
24 bits in 32 bits, unsigned, little endian
- U24_32BE = 11¶
24 bits in 32 bits, unsigned, big endian
- S32 = 12¶
32 bits in 32 bits, signed, native endianness
- S32LE = 12¶
32 bits in 32 bits, signed, little endian
- S32BE = 13¶
32 bits in 32 bits, signed, big endian
- U32 = 14¶
32 bits in 32 bits, unsigned, native endianness
- U32LE = 14¶
32 bits in 32 bits, unsigned, little endian
- U32BE = 15¶
32 bits in 32 bits, unsigned, big endian
- S24 = 16¶
24 bits in 24 bits, signed, native endianness
- S24LE = 16¶
24 bits in 24 bits, signed, little endian
- S24BE = 17¶
24 bits in 24 bits, signed, big endian
- U24 = 18¶
24 bits in 24 bits, unsigned, native endianness
- U24LE = 18¶
24 bits in 24 bits, unsigned, little endian
- U24BE = 19¶
24 bits in 24 bits, unsigned, big endian
- S8 = 2¶
8 bits in 8 bits, signed
- S20 = 20¶
20 bits in 24 bits, signed, native endianness
- S20LE = 20¶
20 bits in 24 bits, signed, little endian
- S20BE = 21¶
20 bits in 24 bits, signed, big endian
- U20 = 22¶
20 bits in 24 bits, unsigned, native endianness
- U20LE = 22¶
20 bits in 24 bits, unsigned, little endian
- U20BE = 23¶
20 bits in 24 bits, unsigned, big endian
- S18 = 24¶
18 bits in 24 bits, signed, native endianness
- S18LE = 24¶
18 bits in 24 bits, signed, little endian
- S18BE = 25¶
18 bits in 24 bits, signed, big endian
- U18 = 26¶
18 bits in 24 bits, unsigned, native endianness
- U18LE = 26¶
18 bits in 24 bits, unsigned, little endian
- U18BE = 27¶
18 bits in 24 bits, unsigned, big endian
- F32 = 28¶
32-bit floating point samples, native endianness
- F32LE = 28¶
32-bit floating point samples, little endian
- F32BE = 29¶
32-bit floating point samples, big endian
- U8 = 3¶
8 bits in 8 bits, unsigned
- F64 = 30¶
64-bit floating point samples, native endianness
- F64LE = 30¶
64-bit floating point samples, little endian
- F64BE = 31¶
64-bit floating point samples, big endian
- S16 = 4¶
16 bits in 16 bits, signed, native endianness
- S16LE = 4¶
16 bits in 16 bits, signed, little endian
- S16BE = 5¶
16 bits in 16 bits, signed, big endian
- U16 = 6¶
16 bits in 16 bits, unsigned, native endianness
- U16LE = 6¶
16 bits in 16 bits, unsigned, little endian
- U16BE = 7¶
16 bits in 16 bits, unsigned, big endian
- S24_32 = 8¶
24 bits in 32 bits, signed, native endianness
- S24_32LE = 8¶
24 bits in 32 bits, signed, little endian
- S24_32BE = 9¶
24 bits in 32 bits, signed, big endian
- class GstAudio.AudioLayout(value)¶
Bases:
GObject.GEnum
Layout of the audio samples for the different channels.
- INTERLEAVED = 0¶
interleaved audio
- NON_INTERLEAVED = 1¶
non-interleaved audio
- class GstAudio.AudioNoiseShapingMethod(value)¶
Bases:
GObject.GEnum
Set of available noise shaping methods
- NONE = 0¶
No noise shaping (default)
- ERROR_FEEDBACK = 1¶
Error feedback
- SIMPLE = 2¶
Simple 2-pole noise shaping
- MEDIUM = 3¶
Medium 5-pole noise shaping
- HIGH = 4¶
High 8-pole noise shaping
- class GstAudio.AudioResamplerFilterInterpolation(value)¶
Bases:
GObject.GEnum
The different filter interpolation methods.
New in version 1.10.
- NONE = 0¶
no interpolation
- LINEAR = 1¶
linear interpolation of the filter coefficients.
- CUBIC = 2¶
cubic interpolation of the filter coefficients.
- class GstAudio.AudioResamplerFilterMode(value)¶
Bases:
GObject.GEnum
Select for the filter tables should be set up.
New in version 1.10.
- INTERPOLATED = 0¶
Use interpolated filter tables. This uses less memory but more CPU and is slightly less accurate but it allows for more efficient variable rate resampling with
GstAudio.AudioResampler.update
().
- FULL = 1¶
Use full filter table. This uses more memory but less CPU.
- AUTO = 2¶
Automatically choose between interpolated and full filter tables.
- class GstAudio.AudioResamplerMethod(value)¶
Bases:
GObject.GEnum
Different subsampling and upsampling methods
New in version 1.10.
- NEAREST = 0¶
Duplicates the samples when upsampling and drops when downsampling
- LINEAR = 1¶
Uses linear interpolation to reconstruct missing samples and averaging to downsample
- CUBIC = 2¶
Uses cubic interpolation
- BLACKMAN_NUTTALL = 3¶
Uses Blackman-Nuttall windowed sinc interpolation
- KAISER = 4¶
Uses Kaiser windowed sinc interpolation
- class GstAudio.AudioRingBufferFormatType(value)¶
Bases:
GObject.GEnum
The format of the samples in the ringbuffer.
- RAW = 0¶
samples in linear or float
- MU_LAW = 1¶
samples in mulaw
- MPEG2_AAC = 10¶
samples in MPEG-2 AAC ADTS format
- MPEG4_AAC = 11¶
samples in MPEG-4 AAC ADTS format
- MPEG2_AAC_RAW = 12¶
samples in MPEG-2 AAC raw format
New in version 1.12.
- MPEG4_AAC_RAW = 13¶
samples in MPEG-4 AAC raw format
New in version 1.12.
- FLAC = 14¶
samples in FLAC format
New in version 1.12.
- DSD = 15¶
samples in DSD format
New in version 1.24.
- A_LAW = 2¶
samples in alaw
- IMA_ADPCM = 3¶
samples in ima adpcm
- MPEG = 4¶
samples in mpeg audio (but not AAC) format
- GSM = 5¶
samples in gsm format
- IEC958 = 6¶
samples in IEC958 frames (e.g. AC3)
- AC3 = 7¶
samples in AC3 format
- EAC3 = 8¶
samples in EAC3 format
- DTS = 9¶
samples in DTS format
- class GstAudio.AudioRingBufferState(value)¶
Bases:
GObject.GEnum
The state of the ringbuffer.
- STOPPED = 0¶
The ringbuffer is stopped
- PAUSED = 1¶
The ringbuffer is paused
- STARTED = 2¶
The ringbuffer is started
- ERROR = 3¶
The ringbuffer has encountered an error after it has been started, e.g. because the device was disconnected
New in version 1.2.
- class GstAudio.DsdFormat(value)¶
Bases:
GObject.GEnum
Enum value describing how DSD bits are grouped.
New in version 1.24.
- classmethod from_string(str)[source]¶
- Parameters:
str (
str
) – a DSD format string- Returns:
the
GstAudio.DsdFormat
for format orGstAudio.DsdFormat.DSD_FORMAT_UNKNOWN
when the string is not a known format.- Return type:
Convert the DSD format string str to its
GstAudio.DsdFormat
.New in version 1.24.
- classmethod get_width(format)[source]¶
- Parameters:
format (
GstAudio.DsdFormat
) – aGstAudio.DsdFormat
- Returns:
Number of bytes in this DSD grouping format.
- Return type:
New in version 1.24.
- classmethod to_string(format)[source]¶
- Parameters:
format (
GstAudio.DsdFormat
) – aGstAudio.DsdFormat
- Returns:
the name corresponding to format
- Return type:
Returns a string containing a descriptive name for the
GstAudio.DsdFormat
if there is one, orNone
otherwise.New in version 1.24.
- DSD_FORMAT_UNKNOWN = 0¶
unknown / invalid DSD format
- DSD_FORMAT_U8 = 1¶
8 DSD bits in 1 byte
- DSD_FORMAT_U16 = 2¶
16 DSD bits in 2 bytes, native endianness
- DSD_FORMAT_U16LE = 2¶
16 DSD bits in 2 bytes, little endian order
- DSD_FORMAT_U16BE = 3¶
16 DSD bits in 2 bytes, big endian order
- DSD_FORMAT_U32 = 4¶
32 DSD bits in 4 bytes, native endianness
- DSD_FORMAT_U32LE = 4¶
32 DSD bits in 4 bytes, little endian order
- DSD_FORMAT_U32BE = 5¶
32 DSD bits in 4 bytes, big endian order
- NUM_DSD_FORMATS = 6¶
number of valid DSD formats
- class GstAudio.StreamVolumeFormat(value)¶
Bases:
GLib.Enum
Different representations of a stream volume.
GstAudio.StreamVolume.convert_volume
() allows to convert between the different representations.Formulas to convert from a linear to a cubic or dB volume are cbrt(val) and 20 * log10 (val).
- LINEAR = 0¶
Linear scale factor, 1.0 = 100%
- CUBIC = 1¶
Cubic volume scale
- DB = 2¶
Logarithmic volume scale (dB, amplitude not power)