Package org.freedesktop.gstreamer.audio

Class GstAudio

java.lang.Object

org.freedesktop.gstreamer.audio.GstAudio

@Generated("io.github.jwharm.JavaGI")
public final class GstAudio
extends Object

Constants and functions that are declared in the global GstAudio namespace.

Field Summary

Fields

Modifier and Type	Field	Description
static final String	AUDIO_CHANNELS_RANGE	Maximum range of allowed channels, for use in template caps strings.
static final String	AUDIO_CONVERTER_OPT_DITHER_METHOD	GstAudioDitherMethod, The dither method to use when changing bit depth.
static final String	AUDIO_CONVERTER_OPT_DITHER_THRESHOLD	Threshold for the output bit depth at/below which to apply dithering.
static final String	AUDIO_CONVERTER_OPT_MIX_MATRIX	GST_TYPE_LIST, The channel mapping matrix.
static final String	AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD	GstAudioNoiseShapingMethod, The noise shaping method to use to mask noise from quantization errors.
static final String	AUDIO_CONVERTER_OPT_QUANTIZATION	G_TYPE_UINT, The quantization amount.
static final String	AUDIO_CONVERTER_OPT_RESAMPLER_METHOD	GstAudioResamplerMethod, The resampler method to use when changing sample rates.
static final int	AUDIO_DECODER_MAX_ERRORS	Default maximum number of errors tolerated before signaling error.
static final String	AUDIO_DECODER_SINK_NAME	The name of the templates for the sink pad.
static final String	AUDIO_DECODER_SRC_NAME	The name of the templates for the source pad.
static final int	AUDIO_DEF_CHANNELS	Standard number of channels used in consumer audio.
static final String	AUDIO_DEF_FORMAT	Standard format used in consumer audio.
static final int	AUDIO_DEF_RATE	Standard sampling rate used in consumer audio.
static final String	AUDIO_ENCODER_SINK_NAME	the name of the templates for the sink pad
static final String	AUDIO_ENCODER_SRC_NAME	the name of the templates for the source pad
static final String	AUDIO_FORMATS_ALL	List of all audio formats, for use in template caps strings.
static final String	AUDIO_RATE_RANGE	Maximum range of allowed sample rates, for use in template caps strings.
static final String	AUDIO_RESAMPLER_OPT_CUBIC_B	G_TYPE_DOUBLE, B parameter of the cubic filter.
static final String	AUDIO_RESAMPLER_OPT_CUBIC_C	G_TYPE_DOUBLE, C parameter of the cubic filter.
static final String	AUDIO_RESAMPLER_OPT_CUTOFF	G_TYPE_DOUBLE, Cutoff parameter for the filter.
static final String	AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION	GST_TYPE_AUDIO_RESAMPLER_INTERPOLATION: how the filter coefficients should be interpolated.
static final String	AUDIO_RESAMPLER_OPT_FILTER_MODE	GST_TYPE_AUDIO_RESAMPLER_FILTER_MODE: how the filter tables should be constructed.
static final String	AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD	G_TYPE_UINT: the amount of memory to use for full filter tables before switching to interpolated filter tables.
static final String	AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE	G_TYPE_UINT, oversampling to use when interpolating filters 8 is the default.
static final String	AUDIO_RESAMPLER_OPT_MAX_PHASE_ERROR	G_TYPE_DOUBLE: The maximum allowed phase error when switching sample rates.
static final String	AUDIO_RESAMPLER_OPT_N_TAPS	G_TYPE_INT: the number of taps to use for the filter.
static final String	AUDIO_RESAMPLER_OPT_STOP_ATTENUATION	G_TYPE_DOUBLE, stopband attenuation in decibels.
static final String	AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH	G_TYPE_DOUBLE, transition bandwidth.
static final int	AUDIO_RESAMPLER_QUALITY_DEFAULT
static final int	AUDIO_RESAMPLER_QUALITY_MAX
static final int	AUDIO_RESAMPLER_QUALITY_MIN
static final String	DSD_FORMATS_ALL	List of all DSD formats, for use in template caps strings.
static final String	DSD_MEDIA_TYPE	The GStreamer media type for DSD.
static final int	DSD_SILENCE_PATTERN_BYTE	Silence pattern for DSD data.
static final String	META_TAG_AUDIO_CHANNELS_STR	This metadata stays relevant as long as channels are unchanged.
static final String	META_TAG_AUDIO_RATE_STR	This metadata stays relevant as long as sample rate is unchanged.
static final String	META_TAG_AUDIO_STR	This metadata is relevant for audio streams.
static final String	META_TAG_DSD_PLANE_OFFSETS_STR	This metadata stays relevant as long as the DSD plane offsets are unchanged.

Constructor Summary

Constructors

Constructor	Description
GstAudio()

Method Summary

Modifier and Type	Method	Description
static long	audioChannelGetFallbackMask(int channels)	Get the fallback channel-mask for the given number of channels.
static boolean	audioChannelPositionsFromMask(long channelMask, AudioChannelPosition[] position)	Convert the channels present in channelMask to a position array (which should have at least channels entries ensured by caller).
static boolean	audioChannelPositionsToMask(AudioChannelPosition[] position, boolean forceOrder, Out<Long> channelMask)	Convert the position array of channels channels to a bitmask.
static String	audioChannelPositionsToString(AudioChannelPosition[] position)	Converts position to a human-readable string representation for debugging purposes.
static boolean	audioChannelPositionsToValidOrder(AudioChannelPosition[] position)	Reorders the channel positions in position from any order to the GStreamer channel order.
static boolean	audioCheckValidChannelPositions(AudioChannelPosition[] position, boolean forceOrder)	Checks if position contains valid channel positions for channels channels.
static Type	audioClippingMetaApiGetType()
static Type	audioDownmixMetaApiGetType()
static AudioFormat[]	audioFormatsRaw()	Return all the raw audio formats supported by GStreamer.
static boolean	audioGetChannelReorderMap(AudioChannelPosition[] from, AudioChannelPosition[] to, int[] reorderMap)	Returns a reorder map for from to to that can be used in custom channel reordering code, e.g.
static int	audioIec61937FrameSize(AudioRingBufferSpec spec)	Calculated the size of the buffer expected by gst_audio_iec61937_payload() for payloading type from spec.
static boolean	audioIec61937Payload(byte[] src, byte[] dst, AudioRingBufferSpec spec, int endianness)	Payloads src in the form specified by IEC 61937 for the type from spec and stores the result in dst.
static Type	audioLevelMetaApiGetType()	Return the GType associated with GstAudioLevelMeta.
static Caps	audioMakeRawCaps(@Nullable AudioFormat[] formats, AudioLayout layout)	Return a generic raw audio caps for formats defined in formats.
static Type	audioMetaApiGetType()
static boolean	audioReorderChannels(byte[] data, AudioFormat format, AudioChannelPosition[] from, AudioChannelPosition[] to)	Reorders data from the channel positions from to the channel positions to.
static AudioClippingMeta	bufferAddAudioClippingMeta(Buffer buffer, Format format, long start, long end)	Attaches GstAudioClippingMeta metadata to buffer with the given parameters.
static AudioDownmixMeta	bufferAddAudioDownmixMeta(Buffer buffer, AudioChannelPosition[] fromPosition, AudioChannelPosition[] toPosition, MemorySegment[] matrix)	Attaches GstAudioDownmixMeta metadata to buffer with the given parameters.
static AudioLevelMeta	bufferAddAudioLevelMeta(Buffer buffer, byte level, boolean voiceActivity)	Attaches audio level information to buffer.
static AudioMeta	bufferAddAudioMeta(Buffer buffer, AudioInfo info, long samples, @Nullable MemorySegment offsets)	Allocates and attaches a GstAudioMeta on buffer, which must be writable for that purpose.
static DsdPlaneOffsetMeta	bufferAddDsdPlaneOffsetMeta(Buffer buffer, int numChannels, long numBytesPerChannel, @Nullable MemorySegment offsets)	Allocates and attaches a GstDsdPlaneOffsetMeta on buffer, which must be writable for that purpose.
static AudioDownmixMeta	bufferGetAudioDownmixMetaForChannels(Buffer buffer, AudioChannelPosition[] toPosition)	Find the GstAudioDownmixMeta on buffer for the given destination channel positions.
static AudioLevelMeta	bufferGetAudioLevelMeta(Buffer buffer)	Find the GstAudioLevelMeta on buffer.
static void	dsdConvert(MemorySegment inputData, MemorySegment outputData, DsdFormat inputFormat, DsdFormat outputFormat, AudioLayout inputLayout, AudioLayout outputLayout, MemorySegment inputPlaneOffsets, MemorySegment outputPlaneOffsets, long numDsdBytes, int numChannels, boolean reverseByteBits)	Converts DSD data from one layout and grouping format to another.
static Type	dsdPlaneOffsetMetaApiGetType()
static void	javagi$ensureInitialized()

Methods inherited from class java.lang.Object

clone, equals, finalize, getClass, hashCode, notify, notifyAll, toString, wait, wait, wait

Field Details

AUDIO_CHANNELS_RANGE

public static final String AUDIO_CHANNELS_RANGE

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

See Also:

• Constant Field Values

AUDIO_CONVERTER_OPT_DITHER_METHOD

public static final String AUDIO_CONVERTER_OPT_DITHER_METHOD

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

See Also:

• Constant Field Values

AUDIO_CONVERTER_OPT_DITHER_THRESHOLD

public static final String AUDIO_CONVERTER_OPT_DITHER_THRESHOLD

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

Default is 20 bit.

See Also:

• Constant Field Values

AUDIO_CONVERTER_OPT_MIX_MATRIX

public static final String AUDIO_CONVERTER_OPT_MIX_MATRIX

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);
 

See Also:

• Constant Field Values

AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD

public static final String 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.

See Also:

• Constant Field Values

AUDIO_CONVERTER_OPT_QUANTIZATION

public static final String AUDIO_CONVERTER_OPT_QUANTIZATION

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

See Also:

• Constant Field Values

AUDIO_CONVERTER_OPT_RESAMPLER_METHOD

public static final String AUDIO_CONVERTER_OPT_RESAMPLER_METHOD

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

See Also:

• Constant Field Values

AUDIO_DECODER_MAX_ERRORS

public static final int AUDIO_DECODER_MAX_ERRORS

Default maximum number of errors tolerated before signaling error.

See Also:

• Constant Field Values

AUDIO_DECODER_SINK_NAME

public static final String AUDIO_DECODER_SINK_NAME

The name of the templates for the sink pad.

See Also:

• Constant Field Values

AUDIO_DECODER_SRC_NAME

public static final String AUDIO_DECODER_SRC_NAME

The name of the templates for the source pad.

See Also:

• Constant Field Values

AUDIO_DEF_CHANNELS

public static final int AUDIO_DEF_CHANNELS

Standard number of channels used in consumer audio.

See Also:

• Constant Field Values

AUDIO_DEF_FORMAT

public static final String AUDIO_DEF_FORMAT

Standard format used in consumer audio.

See Also:

• Constant Field Values

AUDIO_DEF_RATE

public static final int AUDIO_DEF_RATE

Standard sampling rate used in consumer audio.

See Also:

• Constant Field Values

AUDIO_ENCODER_SINK_NAME

public static final String AUDIO_ENCODER_SINK_NAME

the name of the templates for the sink pad

See Also:

• Constant Field Values

AUDIO_ENCODER_SRC_NAME

public static final String AUDIO_ENCODER_SRC_NAME

the name of the templates for the source pad

See Also:

• Constant Field Values

AUDIO_FORMATS_ALL

public static final String AUDIO_FORMATS_ALL

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

See Also:

• Constant Field Values

AUDIO_RATE_RANGE

public static final String AUDIO_RATE_RANGE

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

See Also:

• Constant Field Values

AUDIO_RESAMPLER_OPT_CUBIC_B

public static final String 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.

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

See Also:

• Constant Field Values

AUDIO_RESAMPLER_OPT_CUBIC_C

public static final String 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.

See GST_AUDIO_RESAMPLER_OPT_CUBIC_B for some more common values

See Also:

• Constant Field Values

AUDIO_RESAMPLER_OPT_CUTOFF

public static final String AUDIO_RESAMPLER_OPT_CUTOFF

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

See Also:

• Constant Field Values

AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION

public static final String AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION

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

See Also:

• Constant Field Values

AUDIO_RESAMPLER_OPT_FILTER_MODE

public static final String AUDIO_RESAMPLER_OPT_FILTER_MODE

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

See Also:

• Constant Field Values

AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD

public static final String AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD

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

See Also:

• Constant Field Values

AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE

public static final String AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE

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

See Also:

• Constant Field Values

AUDIO_RESAMPLER_OPT_MAX_PHASE_ERROR

public static final String AUDIO_RESAMPLER_OPT_MAX_PHASE_ERROR

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

See Also:

• Constant Field Values

AUDIO_RESAMPLER_OPT_N_TAPS

public static final String AUDIO_RESAMPLER_OPT_N_TAPS

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

See Also:

• Constant Field Values

AUDIO_RESAMPLER_OPT_STOP_ATTENUATION

public static final String 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.

See Also:

• Constant Field Values

AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH

public static final String 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.

See Also:

• Constant Field Values

AUDIO_RESAMPLER_QUALITY_DEFAULT

public static final int AUDIO_RESAMPLER_QUALITY_DEFAULT

See Also:

• Constant Field Values

AUDIO_RESAMPLER_QUALITY_MAX

public static final int AUDIO_RESAMPLER_QUALITY_MAX

See Also:

• Constant Field Values

AUDIO_RESAMPLER_QUALITY_MIN

public static final int AUDIO_RESAMPLER_QUALITY_MIN

See Also:

• Constant Field Values

DSD_FORMATS_ALL

public static final String DSD_FORMATS_ALL

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.

See Also:

• Constant Field Values

DSD_MEDIA_TYPE

public static final String DSD_MEDIA_TYPE

The GStreamer media type for DSD.

See Also:

• Constant Field Values

DSD_SILENCE_PATTERN_BYTE

public static final int DSD_SILENCE_PATTERN_BYTE

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.

See Also:

• Constant Field Values

META_TAG_AUDIO_CHANNELS_STR

public static final String META_TAG_AUDIO_CHANNELS_STR

This metadata stays relevant as long as channels are unchanged.

See Also:

• Constant Field Values

META_TAG_AUDIO_RATE_STR

public static final String META_TAG_AUDIO_RATE_STR

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

See Also:

• Constant Field Values

META_TAG_AUDIO_STR

public static final String META_TAG_AUDIO_STR

This metadata is relevant for audio streams.

See Also:

• Constant Field Values

META_TAG_DSD_PLANE_OFFSETS_STR

public static final String META_TAG_DSD_PLANE_OFFSETS_STR

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

See Also:

• Constant Field Values

Constructor Details

GstAudio

public GstAudio()

Method Details

javagi$ensureInitialized

public static void javagi$ensureInitialized()

audioChannelGetFallbackMask

public static long audioChannelGetFallbackMask(int channels)

Get the fallback channel-mask for the given number of channels.

This function returns a reasonable fallback channel-mask and should be called as a last resort when the specific channel map is unknown.

Parameters:

channels - the number of channels

Returns:

a fallback channel-mask for channels or 0 when there is no mask and mono.

audioChannelPositionsFromMask

public static boolean audioChannelPositionsFromMask(long channelMask,
 AudioChannelPosition[] position)

Convert the channels present in channelMask to a position array (which should have at least channels entries ensured by caller). If channelMask is set to 0, it is considered as 'not present' for purpose of conversion. A partially valid channelMask with less bits set than the number of channels is considered valid.

Parameters:

channelMask - The input channel_mask

position - The GstAudioChannelPosition<!-- -->s

Returns:

true if channel and channel mask are valid and could be converted

audioChannelPositionsToMask

public static boolean audioChannelPositionsToMask(AudioChannelPosition[] position,
 boolean forceOrder,
 Out<Long> channelMask)

Convert the position array of channels channels to a bitmask.

If forceOrder is true it additionally checks if the channels are in the order required by GStreamer.

Parameters:

position - The GstAudioChannelPositions

forceOrder - Only consider the GStreamer channel order.

channelMask - the output channel mask

Returns:

true if the channel positions are valid and could be converted.

audioChannelPositionsToString

public static String audioChannelPositionsToString(AudioChannelPosition[] position)

Converts position to a human-readable string representation for debugging purposes.

Parameters:

position - The GstAudioChannelPositions to convert.

Returns:

a newly allocated string representing position

audioChannelPositionsToValidOrder

public static boolean audioChannelPositionsToValidOrder(AudioChannelPosition[] position)

Reorders the channel positions in position from any order to the GStreamer channel order.

Parameters:

position - The channel positions to reorder to.

Returns:

true if the channel positions are valid and reordering was successful.

audioCheckValidChannelPositions

public static boolean audioCheckValidChannelPositions(AudioChannelPosition[] position,
 boolean forceOrder)

Checks if position contains valid channel positions for channels channels. If forceOrder is true it additionally checks if the channels are in the order required by GStreamer.

Parameters:

position - The GstAudioChannelPositions to check.

forceOrder - Only consider the GStreamer channel order.

Returns:

true if the channel positions are valid.

audioClippingMetaApiGetType

public static Type audioClippingMetaApiGetType()

audioDownmixMetaApiGetType

public static Type audioDownmixMetaApiGetType()

audioFormatsRaw

public static AudioFormat[] audioFormatsRaw()

Return all the raw audio formats supported by GStreamer.

Returns:

an array of GstAudioFormat

audioGetChannelReorderMap

public static boolean audioGetChannelReorderMap(AudioChannelPosition[] from,
 AudioChannelPosition[] to,
 int[] reorderMap)

Returns a reorder map for from to to that can be used in custom channel reordering code, e.g. to convert from or to the GStreamer channel order. from and to must contain the same number of positions and the same positions, only in a different order.

The resulting reorderMap can be used for reordering by assigning channel i of the input to channel reorder_map[i] of the output.

Parameters:

from - The channel positions to reorder from.

to - The channel positions to reorder to.

reorderMap - Pointer to the reorder map.

Returns:

true if the channel positions are valid and reordering is possible.

audioIec61937FrameSize

public static int audioIec61937FrameSize(AudioRingBufferSpec spec)

Calculated the size of the buffer expected by gst_audio_iec61937_payload() for payloading type from spec.

Parameters:

spec - the ringbufer spec

Returns:

the size or 0 if the given type is not supported or cannot be payloaded.

audioIec61937Payload

public static boolean audioIec61937Payload(byte[] src,
 byte[] dst,
 AudioRingBufferSpec spec,
 int endianness)

Payloads src in the form specified by IEC 61937 for the type from spec and stores the result in dst. src must contain exactly one frame of data and the frame is not checked for errors.

Parameters:

src - a buffer containing the data to payload

dst - the destination buffer to store the payloaded contents in. Should not overlap with src

spec - the ringbufer spec for src

endianness - the expected byte order of the payloaded data

Returns:

transfer-full: true if the payloading was successful, false otherwise.

audioLevelMetaApiGetType

public static Type audioLevelMetaApiGetType()

Return the GType associated with GstAudioLevelMeta.

Returns:

a GType

audioMakeRawCaps

public static Caps audioMakeRawCaps(@Nullable
 @Nullable AudioFormat[] formats,
 AudioLayout layout)

Return a generic raw audio caps for formats defined in formats. If formats is null returns a caps for all the supported raw audio formats, see gst_audio_formats_raw().

Parameters:

formats - an array of raw GstAudioFormat, or null

layout - the layout of audio samples

Returns:

an audio GstCaps

audioMetaApiGetType

public static Type audioMetaApiGetType()

audioReorderChannels

public static boolean audioReorderChannels(byte[] data,
 AudioFormat format,
 AudioChannelPosition[] from,
 AudioChannelPosition[] to)

Reorders data from the channel positions from to the channel positions to. from and to must contain the same number of positions and the same positions, only in a different order.

Note: this function assumes the audio data is in interleaved layout

Parameters:

data - The pointer to the memory.

format - The GstAudioFormat of the buffer.

from - The channel positions in the buffer.

to - The channel positions to convert to.

Returns:

true if the reordering was possible.

bufferAddAudioClippingMeta

public static AudioClippingMeta bufferAddAudioClippingMeta(Buffer buffer,
 Format format,
 long start,
 long end)

Attaches GstAudioClippingMeta metadata to buffer with the given parameters.

Parameters:

buffer - a GstBuffer

format - GstFormat of start and stop, GST_FORMAT_DEFAULT is samples

start - Amount of audio to clip from start of buffer

end - Amount of to clip from end of buffer

Returns:

the GstAudioClippingMeta on buffer.

bufferAddAudioDownmixMeta

public static AudioDownmixMeta bufferAddAudioDownmixMeta(Buffer buffer,
 AudioChannelPosition[] fromPosition,
 AudioChannelPosition[] toPosition,
 MemorySegment[] matrix)

Attaches GstAudioDownmixMeta metadata to buffer with the given parameters.

matrix is an two-dimensional array of toChannels times fromChannels coefficients, i.e. the i-th output channels is constructed by multiplicating the input channels with the coefficients in matrix[i] and taking the sum of the results.

Parameters:

buffer - a GstBuffer

fromPosition - the channel positions of the source

toPosition - the channel positions of the destination

matrix - The matrix coefficients.

Returns:

the GstAudioDownmixMeta on buffer.

bufferAddAudioLevelMeta

public static AudioLevelMeta bufferAddAudioLevelMeta(Buffer buffer,
 byte level,
 boolean voiceActivity)

Attaches audio level information to buffer. (RFC 6464)

Parameters:

buffer - a GstBuffer

level - the -dBov from 0-127 (127 is silence).

voiceActivity - whether the buffer contains voice activity.

Returns:

the GstAudioLevelMeta on buffer.

bufferAddAudioMeta

public static AudioMeta bufferAddAudioMeta(Buffer buffer,
 AudioInfo info,
 long samples,
 @Nullable
 @Nullable MemorySegment offsets)

Allocates and attaches a GstAudioMeta on buffer, which must be writable for that purpose. The fields of the GstAudioMeta are directly populated from the arguments of this function.

When info->layout is AudioLayout.NON_INTERLEAVED and offsets is null, the offsets are calculated with a formula that assumes the planes are tightly packed and in sequence: offsetssamples * sample_stride It is not allowed for channels to overlap in memory, i.e. for each i in [0, channels), the range [@offsets[i, offsets[i] + samples * sample_stride) must not overlap with any other such range. This function will assert if the parameters specified cause this restriction to be violated.

It is, obviously, also not allowed to specify parameters that would cause out-of-bounds memory access on buffer. This is also checked, which means that you must add enough memory on the buffer before adding this meta.

Parameters:

buffer - a GstBuffer

info - the audio properties of the buffer

samples - the number of valid samples in the buffer

offsets - the offsets (in bytes) where each channel plane starts in the buffer or null to calculate it (see below); must be null also when info->layout is AudioLayout.INTERLEAVED

Returns:

the GstAudioMeta that was attached on the buffer

bufferAddDsdPlaneOffsetMeta

public static DsdPlaneOffsetMeta bufferAddDsdPlaneOffsetMeta(Buffer buffer,
 int numChannels,
 long numBytesPerChannel,
 @Nullable
 @Nullable MemorySegment offsets)

Allocates and attaches a GstDsdPlaneOffsetMeta on buffer, which must be writable for that purpose. The fields of the GstDsdPlaneOffsetMeta are directly populated from the arguments of this function.

If offsets is NULL, then the meta's offsets field is left uninitialized. This is useful if for example offset values are to be calculated in the meta's offsets field in-place. Similarly, numBytesPerChannel can be set to 0, but only if offsets is NULL. This is useful if the number of bytes per channel is known only later.

It is not allowed for channels to overlap in memory, i.e. for each i in [0, channels), the range offsets[i + numBytesPerChannel) must not overlap with any other such range. This function will assert if the parameters specified cause this restriction to be violated.

It is, obviously, also not allowed to specify parameters that would cause out-of-bounds memory access on buffer. This is also checked, which means that you must add enough memory on the buffer before adding this meta.

This meta is only needed for non-interleaved (= planar) DSD data.

Parameters:

buffer - a GstBuffer

numChannels - Number of channels in the DSD data

numBytesPerChannel - Number of bytes per channel

offsets - the offsets (in bytes) where each channel plane starts in the buffer

Returns:

the GstDsdPlaneOffsetMeta that was attached on the buffer

bufferGetAudioDownmixMetaForChannels

public static AudioDownmixMeta bufferGetAudioDownmixMetaForChannels(Buffer buffer,
 AudioChannelPosition[] toPosition)

Find the GstAudioDownmixMeta on buffer for the given destination channel positions.

Parameters:

buffer - a GstBuffer

toPosition - the channel positions of the destination

Returns:

the GstAudioDownmixMeta on buffer.

bufferGetAudioLevelMeta

public static AudioLevelMeta bufferGetAudioLevelMeta(Buffer buffer)

Find the GstAudioLevelMeta on buffer.

Parameters:

buffer - a GstBuffer

Returns:

the GstAudioLevelMeta or null when there is no such metadata on buffer.

dsdConvert

public static void dsdConvert(MemorySegment inputData,
 MemorySegment outputData,
 DsdFormat inputFormat,
 DsdFormat outputFormat,
 AudioLayout inputLayout,
 AudioLayout outputLayout,
 MemorySegment inputPlaneOffsets,
 MemorySegment outputPlaneOffsets,
 long numDsdBytes,
 int numChannels,
 boolean reverseByteBits)

Converts DSD data from one layout and grouping format to another. numBytes must be an integer multiple of the width of both input and output format. For example, if the input format is GST_DSD_FORMAT_U32LE, and the output format is GST_DSD_FORMAT_U16BE, then numBytes must be an integer multiple of both 4 (U32LE width) and 2 (U16BE width).

reverseByteBits is necessary if the bit order within the DSD bytes needs to be reversed. This is rarely necessary, and is not to be confused with the endianness of formats (which determines the ordering of *bytes*).

inputPlaneOffsets must not be NULL if inputLayout is set to GST_AUDIO_LAYOUT_NON_INTERLEAVED. The same applies to outputPlaneOffsets. These plane offsets define the starting offset of the planes (there is exactly one plane per channel) within inputData and outputData respectively. If GST_AUDIO_LAYOUT_INTERLEAVED is used, the plane offsets are ignored.

Parameters:

inputData - the DSD format conversion's input source

outputData - the DSD format conversion's output destination

inputFormat - DSD format of the input data to convert from

outputFormat - DSD format of the output data to convert to

inputLayout - Input data layout

outputLayout - Output data layout

inputPlaneOffsets - Plane offsets for non-interleaved input data

outputPlaneOffsets - Plane offsets for non-interleaved output data

numDsdBytes - How many bytes with DSD data to convert

numChannels - Number of channels (must be at least 1)

reverseByteBits - If TRUE, reverse the bits in each DSD byte

dsdPlaneOffsetMetaApiGetType

public static Type dsdPlaneOffsetMetaApiGetType()