Class GstAudio

java.lang.Object
org.freedesktop.gstreamer.audio.GstAudio
@Generated("org.javagi.JavaGI") public final class GstAudio extends Object
Constants and functions that are declared in the global GstAudio namespace.

  • 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:

    • 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:

    • 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.

      Since:
      1.22
      See Also:

    • 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:

    • 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:

    • 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:

    • 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:

    • AUDIO_DECODER_MAX_ERRORS

      public static final int AUDIO_DECODER_MAX_ERRORS
      Default maximum number of errors tolerated before signaling error.
      See Also:

    • AUDIO_DECODER_SINK_NAME

      public static final String AUDIO_DECODER_SINK_NAME
      The name of the templates for the sink pad.
      See Also:

    • AUDIO_DECODER_SRC_NAME

      public static final String AUDIO_DECODER_SRC_NAME
      The name of the templates for the source pad.
      See Also:

    • AUDIO_DEF_CHANNELS

      public static final int AUDIO_DEF_CHANNELS
      Standard number of channels used in consumer audio.
      See Also:

    • AUDIO_DEF_FORMAT

      public static final String AUDIO_DEF_FORMAT
      Standard format used in consumer audio.
      See Also:

    • AUDIO_DEF_RATE

      public static final int AUDIO_DEF_RATE
      Standard sampling rate used in consumer audio.
      See Also:

    • AUDIO_ENCODER_SINK_NAME

      public static final String AUDIO_ENCODER_SINK_NAME
      the name of the templates for the sink pad
      See Also:

    • AUDIO_ENCODER_SRC_NAME

      public static final String AUDIO_ENCODER_SRC_NAME
      the name of the templates for the source pad
      See Also:

    • 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:

    • AUDIO_FORMAT_LAST

      public static final int AUDIO_FORMAT_LAST
      Number of audio formats in GstAudioFormat.
      Since:
      1.26
      See Also:

    • AUDIO_RATE_RANGE

      public static final String AUDIO_RATE_RANGE
      Maximum range of allowed sample rates, for use in template caps strings.
      See Also:

    • 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:

    • 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:

    • 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:

    • 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:

    • 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:

    • 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:

    • 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:

    • 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:

    • 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:

    • 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:

    • 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:

    • AUDIO_RESAMPLER_QUALITY_DEFAULT

      public static final int AUDIO_RESAMPLER_QUALITY_DEFAULT
      See Also:

    • AUDIO_RESAMPLER_QUALITY_MAX

      public static final int AUDIO_RESAMPLER_QUALITY_MAX
      See Also:

    • AUDIO_RESAMPLER_QUALITY_MIN

      public static final int AUDIO_RESAMPLER_QUALITY_MIN
      See Also:

    • 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.

      Since:
      1.24
      See Also:

    • DSD_MEDIA_TYPE

      public static final String DSD_MEDIA_TYPE
      The GStreamer media type for DSD.
      Since:
      1.24
      See Also:

    • 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.

      Since:
      1.24
      See Also:

    • META_TAG_AUDIO_CHANNELS_STR

      public static final String META_TAG_AUDIO_CHANNELS_STR
      This metadata stays relevant as long as channels are unchanged.
      Since:
      1.2
      See Also:

    • 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.
      Since:
      1.8
      See Also:

    • META_TAG_AUDIO_STR

      public static final String META_TAG_AUDIO_STR
      This metadata is relevant for audio streams.
      Since:
      1.2
      See Also:

    • 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.
      Since:
      1.24
      See Also:

  • 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.
      Since:
      1.8

    • audioChannelPositionsFromMask

      public static boolean audioChannelPositionsFromMask(long channelMask, @Nullable AudioChannelPosition @Nullable [] 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 GstAudioChannelPositions
      Returns:
      true if channel and channel mask are valid and could be converted

    • audioChannelPositionsToMask

      public static boolean audioChannelPositionsToMask(@Nullable AudioChannelPosition @Nullable [] 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(@Nullable AudioChannelPosition @Nullable [] 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
      Since:
      1.10

    • audioChannelPositionsToValidOrder

      public static boolean audioChannelPositionsToValidOrder(@Nullable AudioChannelPosition @Nullable [] 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(@Nullable AudioChannelPosition @Nullable [] 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
      Since:
      1.18

    • audioGetChannelReorderMap

      public static boolean audioGetChannelReorderMap(@Nullable AudioChannelPosition @Nullable [] from, @Nullable AudioChannelPosition @Nullable [] to, @Nullable int @Nullable [] 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

      invalid reference 
      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(@Nullable byte @Nullable [] src, @Nullable byte @Nullable [] 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
      Since:
      1.20

    • audioMakeRawCaps

      public static Caps audioMakeRawCaps(@Nullable AudioFormat @Nullable [] 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
      Since:
      1.18

    • audioMetaApiGetType

      public static Type audioMetaApiGetType()

    • audioReorderChannels

      public static boolean audioReorderChannels(@Nullable byte @Nullable [] data, AudioFormat format, @Nullable AudioChannelPosition @Nullable [] from, @Nullable AudioChannelPosition @Nullable [] 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.

      This function internally calls gst_audio_get_channel_reorder_map() and gst_audio_reorder_channels_with_reorder_map(). It is more efficient to call gst_audio_get_channel_reorder_map() once to retrieve the reorder map and then call gst_audio_reorder_channels_with_reorder_map() with the same reorder map until the channel positions change.

      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.

    • audioReorderChannelsWithReorderMap

      public static void audioReorderChannelsWithReorderMap(@Nullable byte @Nullable [] data, int bps, @Nullable int @Nullable [] reorderMap)

      Reorders data with the given reorderMap.

      The reorder map can be retrieved for example with gst_audio_get_channel_reorder_map().

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

      Parameters:
      data - The pointer to the memory.
      bps - The number of bytes per sample.
      reorderMap - The channel reorder map.
      Since:
      1.26

    • 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.
      Since:
      1.8

    • bufferAddAudioDownmixMeta

      public static AudioDownmixMeta bufferAddAudioDownmixMeta(Buffer buffer, @Nullable AudioChannelPosition @Nullable [] fromPosition, @Nullable AudioChannelPosition @Nullable [] toPosition, float[] 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 @Nullable 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.
      Since:
      1.20

    • bufferAddAudioMeta

      public 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. 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: offsets`samples * 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

      invalid reference 
      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
      Since:
      1.16

    • bufferAddDsdPlaneOffsetMeta

      public 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. 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 offsets[i, 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
      Since:
      1.24

    • bufferGetAudioDownmixMetaForChannels

      public static AudioDownmixMeta bufferGetAudioDownmixMetaForChannels(Buffer buffer, @Nullable AudioChannelPosition @Nullable [] 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 @Nullable 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.
      Since:
      1.20

    • 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
      Since:
      1.24

    • dsdPlaneOffsetMetaApiGetType

      public static Type dsdPlaneOffsetMetaApiGetType()