GstAudio meta
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.
GstAudioClippingMeta
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.
This is used to ensure that decoded data in the end has the same amount of samples, and multiply decoded streams can be gaplessly concatenated.
Note: If clipping of the start is done by adjusting the segment, this meta has to be dropped from buffers as otherwise clipping could happen twice.
Members
format
(GstFormat)
–
GstFormat of start and stop, GST_FORMAT_DEFAULT is samples
start
(guint64)
–
Amount of audio to clip from start of buffer
end
(guint64)
–
Amount of to clip from end of buffer
Since : 1.8
GstAudio.AudioClippingMeta
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.
This is used to ensure that decoded data in the end has the same amount of samples, and multiply decoded streams can be gaplessly concatenated.
Note: If clipping of the start is done by adjusting the segment, this meta has to be dropped from buffers as otherwise clipping could happen twice.
Members
format
(Gst.Format)
–
GstFormat of start and stop, GST_FORMAT_DEFAULT is samples
start
(Number)
–
Amount of audio to clip from start of buffer
end
(Number)
–
Amount of to clip from end of buffer
Since : 1.8
GstAudio.AudioClippingMeta
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.
This is used to ensure that decoded data in the end has the same amount of samples, and multiply decoded streams can be gaplessly concatenated.
Note: If clipping of the start is done by adjusting the segment, this meta has to be dropped from buffers as otherwise clipping could happen twice.
Members
format
(Gst.Format)
–
GstFormat of start and stop, GST_FORMAT_DEFAULT is samples
start
(int)
–
Amount of audio to clip from start of buffer
end
(int)
–
Amount of to clip from end of buffer
Since : 1.8
Functions
gst_audio_clipping_meta_get_info
const GstMetaInfo * gst_audio_clipping_meta_get_info ()
GstAudio.prototype.audio_clipping_meta_get_info
function GstAudio.prototype.audio_clipping_meta_get_info(): {
// javascript wrapper for 'gst_audio_clipping_meta_get_info'
}
GstAudio.audio_clipping_meta_get_info
def GstAudio.audio_clipping_meta_get_info ():
#python wrapper for 'gst_audio_clipping_meta_get_info'
GstAudioDownmixMeta
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.
matrix is an two-dimensional array of to_channels times from_channels 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.
Members
from_position
(GstAudioChannelPosition *)
–
the channel positions of the source
to_position
(GstAudioChannelPosition *)
–
the channel positions of the destination
from_channels
(gint)
–
the number of channels of the source
to_channels
(gint)
–
the number of channels of the destination
matrix
(gfloat **)
–
the matrix coefficients.
GstAudio.AudioDownmixMeta
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.
matrix is an two-dimensional array of to_channels times from_channels 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.
Members
from_position
(GstAudio.AudioChannelPosition)
–
the channel positions of the source
to_position
(GstAudio.AudioChannelPosition)
–
the channel positions of the destination
from_channels
(Number)
–
the number of channels of the source
to_channels
(Number)
–
the number of channels of the destination
matrix
(Number)
–
the matrix coefficients.
GstAudio.AudioDownmixMeta
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.
matrix is an two-dimensional array of to_channels times from_channels 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.
Members
from_position
(GstAudio.AudioChannelPosition)
–
the channel positions of the source
to_position
(GstAudio.AudioChannelPosition)
–
the channel positions of the destination
from_channels
(int)
–
the number of channels of the source
to_channels
(int)
–
the number of channels of the destination
matrix
(float)
–
the matrix coefficients.
Functions
gst_audio_downmix_meta_get_info
const GstMetaInfo * gst_audio_downmix_meta_get_info ()
GstAudio.prototype.audio_downmix_meta_get_info
function GstAudio.prototype.audio_downmix_meta_get_info(): {
// javascript wrapper for 'gst_audio_downmix_meta_get_info'
}
GstAudio.audio_downmix_meta_get_info
def GstAudio.audio_downmix_meta_get_info ():
#python wrapper for 'gst_audio_downmix_meta_get_info'
GstAudioLevelMeta
Meta containing Audio Level Indication: https://tools.ietf.org/html/rfc6464
Members
level
(guint8)
–
the -dBov from 0-127 (127 is silence).
voice_activity
(gboolean)
–
whether the buffer contains voice activity
Since : 1.20
Functions
gst_audio_level_meta_get_info
const GstMetaInfo * gst_audio_level_meta_get_info ()
Return the GstMetaInfo associated with GstAudioLevelMeta.
Since : 1.20
GstAudio.prototype.audio_level_meta_get_info
function GstAudio.prototype.audio_level_meta_get_info(): {
// javascript wrapper for 'gst_audio_level_meta_get_info'
}
Return the Gst.MetaInfo associated with GstAudio.AudioLevelMeta.
Since : 1.20
GstAudio.audio_level_meta_get_info
def GstAudio.audio_level_meta_get_info ():
#python wrapper for 'gst_audio_level_meta_get_info'
Return the Gst.MetaInfo associated with GstAudio.AudioLevelMeta.
Since : 1.20
GstAudioMeta
Buffer metadata describing how data is laid out inside the buffer. This is useful for non-interleaved (planar) buffers, where it is necessary to have a place to store where each plane starts and how long each plane is.
It is a requirement for non-interleaved buffers to have this metadata attached and to be mapped with gst_audio_buffer_map in order to ensure correct handling of clipping and channel reordering.
The different channels in offsets are always in the GStreamer channel order. Zero-copy channel reordering can be implemented by swapping the values in offsets.
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.
It is, however, allowed to have parts of the buffer memory unused, by using offsets and samples in such a way that leave gaps on it. This is used to implement zero-copy clipping in non-interleaved buffers.
Obviously, due to the above, it is not safe to infer the number of valid samples from the size of the buffer. You should always use the samples variable of this metadata.
Note that for interleaved audio it is not a requirement to have this metadata attached and at the moment of writing, there is actually no use case to do so. It is, however, allowed to attach it, for some potential future use case.
Since 1.24 it can be serialized using gst_meta_serialize and gst_meta_deserialize.
Members
info
(GstAudioInfo)
–
the audio properties of the buffer
samples
(gsize)
–
the number of valid samples in the buffer
Since : 1.16
GstAudio.AudioMeta
Buffer metadata describing how data is laid out inside the buffer. This is useful for non-interleaved (planar) buffers, where it is necessary to have a place to store where each plane starts and how long each plane is.
It is a requirement for non-interleaved buffers to have this metadata attached and to be mapped with GstAudio.prototype.audio_buffer_map in order to ensure correct handling of clipping and channel reordering.
The different channels in offsets are always in the GStreamer channel order. Zero-copy channel reordering can be implemented by swapping the values in offsets.
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.
It is, however, allowed to have parts of the buffer memory unused, by using offsets and samples in such a way that leave gaps on it. This is used to implement zero-copy clipping in non-interleaved buffers.
Obviously, due to the above, it is not safe to infer the number of valid samples from the size of the buffer. You should always use the samples variable of this metadata.
Note that for interleaved audio it is not a requirement to have this metadata attached and at the moment of writing, there is actually no use case to do so. It is, however, allowed to attach it, for some potential future use case.
Since 1.24 it can be serialized using Gst.Meta.prototype.serialize and Gst.prototype.meta_deserialize.
Members
info
(GstAudio.AudioInfo)
–
the audio properties of the buffer
samples
(Number)
–
the number of valid samples in the buffer
Since : 1.16
GstAudio.AudioMeta
Buffer metadata describing how data is laid out inside the buffer. This is useful for non-interleaved (planar) buffers, where it is necessary to have a place to store where each plane starts and how long each plane is.
It is a requirement for non-interleaved buffers to have this metadata attached and to be mapped with GstAudio.audio_buffer_map in order to ensure correct handling of clipping and channel reordering.
The different channels in offsets are always in the GStreamer channel order. Zero-copy channel reordering can be implemented by swapping the values in offsets.
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.
It is, however, allowed to have parts of the buffer memory unused, by using offsets and samples in such a way that leave gaps on it. This is used to implement zero-copy clipping in non-interleaved buffers.
Obviously, due to the above, it is not safe to infer the number of valid samples from the size of the buffer. You should always use the samples variable of this metadata.
Note that for interleaved audio it is not a requirement to have this metadata attached and at the moment of writing, there is actually no use case to do so. It is, however, allowed to attach it, for some potential future use case.
Since 1.24 it can be serialized using Gst.Meta.serialize and Gst.meta_deserialize.
Members
info
(GstAudio.AudioInfo)
–
the audio properties of the buffer
samples
(int)
–
the number of valid samples in the buffer
Since : 1.16
Functions
gst_audio_meta_get_info
const GstMetaInfo * gst_audio_meta_get_info ()
GstAudio.prototype.audio_meta_get_info
function GstAudio.prototype.audio_meta_get_info(): {
// javascript wrapper for 'gst_audio_meta_get_info'
}
GstAudio.audio_meta_get_info
def GstAudio.audio_meta_get_info ():
#python wrapper for 'gst_audio_meta_get_info'
Functions
gst_audio_clipping_meta_api_get_type
GType gst_audio_clipping_meta_api_get_type ()
GstAudio.prototype.audio_clipping_meta_api_get_type
function GstAudio.prototype.audio_clipping_meta_api_get_type(): {
// javascript wrapper for 'gst_audio_clipping_meta_api_get_type'
}
GstAudio.audio_clipping_meta_api_get_type
def GstAudio.audio_clipping_meta_api_get_type ():
#python wrapper for 'gst_audio_clipping_meta_api_get_type'
gst_audio_downmix_meta_api_get_type
GType gst_audio_downmix_meta_api_get_type ()
GstAudio.prototype.audio_downmix_meta_api_get_type
function GstAudio.prototype.audio_downmix_meta_api_get_type(): {
// javascript wrapper for 'gst_audio_downmix_meta_api_get_type'
}
GstAudio.audio_downmix_meta_api_get_type
def GstAudio.audio_downmix_meta_api_get_type ():
#python wrapper for 'gst_audio_downmix_meta_api_get_type'
gst_audio_level_meta_api_get_type
GType gst_audio_level_meta_api_get_type ()
Return the GType associated with GstAudioLevelMeta.
a GType
Since : 1.20
GstAudio.prototype.audio_level_meta_api_get_type
function GstAudio.prototype.audio_level_meta_api_get_type(): {
// javascript wrapper for 'gst_audio_level_meta_api_get_type'
}
Return the GObject.Type associated with GstAudio.AudioLevelMeta.
Since : 1.20
GstAudio.audio_level_meta_api_get_type
def GstAudio.audio_level_meta_api_get_type ():
#python wrapper for 'gst_audio_level_meta_api_get_type'
Return the GObject.Type associated with GstAudio.AudioLevelMeta.
Since : 1.20
GstAudio.prototype.audio_meta_api_get_type
function GstAudio.prototype.audio_meta_api_get_type(): {
// javascript wrapper for 'gst_audio_meta_api_get_type'
}
GstAudio.audio_meta_api_get_type
def GstAudio.audio_meta_api_get_type ():
#python wrapper for 'gst_audio_meta_api_get_type'
gst_buffer_add_audio_clipping_meta
GstAudioClippingMeta * gst_buffer_add_audio_clipping_meta (GstBuffer * buffer, GstFormat format, guint64 start, guint64 end)
Attaches GstAudioClippingMeta metadata to buffer with the given parameters.
Parameters:
buffer
–
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
the GstAudioClippingMeta on buffer.
Since : 1.8
GstAudio.prototype.buffer_add_audio_clipping_meta
function GstAudio.prototype.buffer_add_audio_clipping_meta(buffer: Gst.Buffer, format: Gst.Format, start: Number, end: Number): {
// javascript wrapper for 'gst_buffer_add_audio_clipping_meta'
}
Attaches GstAudio.AudioClippingMeta metadata to buffer with the given parameters.
Parameters:
GstFormat of start and stop, GST_FORMAT_DEFAULT is samples
Amount of audio to clip from start of buffer
Amount of to clip from end of buffer
the GstAudio.AudioClippingMeta on buffer.
Since : 1.8
GstAudio.buffer_add_audio_clipping_meta
def GstAudio.buffer_add_audio_clipping_meta (buffer, format, start, end):
#python wrapper for 'gst_buffer_add_audio_clipping_meta'
Attaches GstAudio.AudioClippingMeta metadata to buffer with the given parameters.
Parameters:
GstFormat of start and stop, GST_FORMAT_DEFAULT is samples
Amount of audio to clip from start of buffer
Amount of to clip from end of buffer
the GstAudio.AudioClippingMeta on buffer.
Since : 1.8
gst_buffer_add_audio_downmix_meta
GstAudioDownmixMeta * gst_buffer_add_audio_downmix_meta (GstBuffer * buffer, const GstAudioChannelPosition * from_position, gint from_channels, const GstAudioChannelPosition * to_position, gint to_channels, const gfloat ** matrix)
Attaches GstAudioDownmixMeta metadata to buffer with the given parameters.
matrix is an two-dimensional array of to_channels times from_channels 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
–
from_position
(
[arraylength=from_channels])
–
the channel positions of the source
from_channels
–
The number of channels of the source
to_position
(
[arraylength=to_channels])
–
the channel positions of the destination
to_channels
–
The number of channels of the destination
matrix
–
The matrix coefficients.
the GstAudioDownmixMeta on buffer.
GstAudio.prototype.buffer_add_audio_downmix_meta
function GstAudio.prototype.buffer_add_audio_downmix_meta(buffer: Gst.Buffer, from_position: [ GstAudio.AudioChannelPosition ], from_channels: Number, to_position: [ GstAudio.AudioChannelPosition ], to_channels: Number, matrix: Number): {
// javascript wrapper for 'gst_buffer_add_audio_downmix_meta'
}
Attaches GstAudio.AudioDownmixMeta metadata to buffer with the given parameters.
matrix is an two-dimensional array of to_channels times from_channels 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:
the channel positions of the source
The number of channels of the source
the channel positions of the destination
The number of channels of the destination
The matrix coefficients.
the GstAudio.AudioDownmixMeta on buffer.
GstAudio.buffer_add_audio_downmix_meta
def GstAudio.buffer_add_audio_downmix_meta (buffer, from_position, from_channels, to_position, to_channels, matrix):
#python wrapper for 'gst_buffer_add_audio_downmix_meta'
Attaches GstAudio.AudioDownmixMeta metadata to buffer with the given parameters.
matrix is an two-dimensional array of to_channels times from_channels 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:
the channel positions of the source
The number of channels of the source
the channel positions of the destination
The number of channels of the destination
The matrix coefficients.
the GstAudio.AudioDownmixMeta on buffer.
gst_buffer_add_audio_level_meta
GstAudioLevelMeta * gst_buffer_add_audio_level_meta (GstBuffer * buffer, guint8 level, gboolean voice_activity)
Attaches audio level information to buffer. (RFC 6464)
Parameters:
buffer
–
level
–
the -dBov from 0-127 (127 is silence).
voice_activity
–
whether the buffer contains voice activity.
the GstAudioLevelMeta on buffer.
Since : 1.20
GstAudio.prototype.buffer_add_audio_level_meta
function GstAudio.prototype.buffer_add_audio_level_meta(buffer: Gst.Buffer, level: Number, voice_activity: Number): {
// javascript wrapper for 'gst_buffer_add_audio_level_meta'
}
Attaches audio level information to buffer. (RFC 6464)
Parameters:
the -dBov from 0-127 (127 is silence).
whether the buffer contains voice activity.
the GstAudio.AudioLevelMeta on buffer.
Since : 1.20
GstAudio.buffer_add_audio_level_meta
def GstAudio.buffer_add_audio_level_meta (buffer, level, voice_activity):
#python wrapper for 'gst_buffer_add_audio_level_meta'
Attaches audio level information to buffer. (RFC 6464)
Parameters:
the -dBov from 0-127 (127 is silence).
whether the buffer contains voice activity.
the GstAudio.AudioLevelMeta on buffer.
Since : 1.20
gst_buffer_add_audio_meta
GstAudioMeta * gst_buffer_add_audio_meta (GstBuffer * buffer, const GstAudioInfo * info, gsize samples, gsize * 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 GST_AUDIO_LAYOUT_NON_INTERLEAVED and offsets is NULL, the offsets are calculated with a formula that assumes the planes are tightly packed and in sequence: offsets[channel] = channel * 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[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
–
info
–
the audio properties of the buffer
samples
–
the number of valid samples in the buffer
offsets
(
[nullable])
–
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 GST_AUDIO_LAYOUT_INTERLEAVED
the GstAudioMeta that was attached on the buffer
Since : 1.16
GstAudio.prototype.buffer_add_audio_meta
function GstAudio.prototype.buffer_add_audio_meta(buffer: Gst.Buffer, info: GstAudio.AudioInfo, samples: Number, offsets: Number): {
// javascript wrapper for 'gst_buffer_add_audio_meta'
}
Allocates and attaches a GstAudio.AudioMeta on buffer, which must be writable for that purpose. The fields of the GstAudio.AudioMeta are directly populated from the arguments of this function.
When info->layout is GstAudio.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[channel] = channel * 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[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:
the audio properties of the buffer
the number of valid samples in the buffer
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 GstAudio.AudioLayout.INTERLEAVED
the GstAudio.AudioMeta that was attached on the buffer
Since : 1.16
GstAudio.buffer_add_audio_meta
def GstAudio.buffer_add_audio_meta (buffer, info, samples, offsets):
#python wrapper for 'gst_buffer_add_audio_meta'
Allocates and attaches a GstAudio.AudioMeta on buffer, which must be writable for that purpose. The fields of the GstAudio.AudioMeta are directly populated from the arguments of this function.
When info->layout is GstAudio.AudioLayout.NON_INTERLEAVED and offsets is None, the offsets are calculated with a formula that assumes the planes are tightly packed and in sequence: offsets[channel] = channel * 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[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:
the audio properties of the buffer
the number of valid samples in the buffer
the offsets (in bytes) where each channel plane starts in the buffer or None to calculate it (see below); must be None also when info->layout is GstAudio.AudioLayout.INTERLEAVED
the GstAudio.AudioMeta that was attached on the buffer
Since : 1.16
gst_buffer_get_audio_downmix_meta_for_channels
GstAudioDownmixMeta * gst_buffer_get_audio_downmix_meta_for_channels (GstBuffer * buffer, const GstAudioChannelPosition * to_position, gint to_channels)
Find the GstAudioDownmixMeta on buffer for the given destination channel positions.
Parameters:
buffer
–
to_position
(
[arraylength=to_channels])
–
the channel positions of the destination
to_channels
–
The number of channels of the destination
the GstAudioDownmixMeta on buffer.
GstAudio.prototype.buffer_get_audio_downmix_meta_for_channels
function GstAudio.prototype.buffer_get_audio_downmix_meta_for_channels(buffer: Gst.Buffer, to_position: [ GstAudio.AudioChannelPosition ], to_channels: Number): {
// javascript wrapper for 'gst_buffer_get_audio_downmix_meta_for_channels'
}
Find the GstAudio.AudioDownmixMeta on buffer for the given destination channel positions.
Parameters:
the channel positions of the destination
The number of channels of the destination
the GstAudio.AudioDownmixMeta on buffer.
GstAudio.buffer_get_audio_downmix_meta_for_channels
def GstAudio.buffer_get_audio_downmix_meta_for_channels (buffer, to_position, to_channels):
#python wrapper for 'gst_buffer_get_audio_downmix_meta_for_channels'
Find the GstAudio.AudioDownmixMeta on buffer for the given destination channel positions.
Parameters:
the channel positions of the destination
The number of channels of the destination
the GstAudio.AudioDownmixMeta on buffer.
gst_buffer_get_audio_level_meta
GstAudioLevelMeta * gst_buffer_get_audio_level_meta (GstBuffer * buffer)
Find the GstAudioLevelMeta on buffer.
Parameters:
buffer
–
the GstAudioLevelMeta or NULL when there is no such metadata on buffer.
Since : 1.20
GstAudio.prototype.buffer_get_audio_level_meta
function GstAudio.prototype.buffer_get_audio_level_meta(buffer: Gst.Buffer): {
// javascript wrapper for 'gst_buffer_get_audio_level_meta'
}
Find the GstAudio.AudioLevelMeta on buffer.
Parameters:
the GstAudio.AudioLevelMeta or null when there is no such metadata on buffer.
Since : 1.20
GstAudio.buffer_get_audio_level_meta
def GstAudio.buffer_get_audio_level_meta (buffer):
#python wrapper for 'gst_buffer_get_audio_level_meta'
Find the GstAudio.AudioLevelMeta on buffer.
Parameters:
the GstAudio.AudioLevelMeta or None when there is no such metadata on buffer.
Since : 1.20
Function Macros
gst_buffer_get_audio_clipping_meta
#define gst_buffer_get_audio_clipping_meta(b) ((GstAudioClippingMeta*)gst_buffer_get_meta((b), GST_AUDIO_CLIPPING_META_API_TYPE))
gst_buffer_get_audio_downmix_meta
#define gst_buffer_get_audio_downmix_meta(b) ((GstAudioDownmixMeta*)gst_buffer_get_meta((b), GST_AUDIO_DOWNMIX_META_API_TYPE))
gst_buffer_get_audio_meta
#define gst_buffer_get_audio_meta(b) \ ((GstAudioMeta*)gst_buffer_get_meta((b), GST_AUDIO_META_API_TYPE))
Constants
GST_AUDIO_CLIPPING_META_API_TYPE
#define GST_AUDIO_CLIPPING_META_API_TYPE (gst_audio_clipping_meta_api_get_type())
GST_AUDIO_CLIPPING_META_INFO
#define GST_AUDIO_CLIPPING_META_INFO (gst_audio_clipping_meta_get_info())
GST_AUDIO_DOWNMIX_META_API_TYPE
#define GST_AUDIO_DOWNMIX_META_API_TYPE (gst_audio_downmix_meta_api_get_type())
GST_AUDIO_DOWNMIX_META_INFO
#define GST_AUDIO_DOWNMIX_META_INFO (gst_audio_downmix_meta_get_info())
GST_AUDIO_LEVEL_META_API_TYPE
#define GST_AUDIO_LEVEL_META_API_TYPE (gst_audio_level_meta_api_get_type())
The GType associated with GstAudioLevelMeta.
Since : 1.20
GST_AUDIO_LEVEL_META_INFO
#define GST_AUDIO_LEVEL_META_INFO (gst_audio_level_meta_get_info())
The GstMetaInfo associated with GstAudioLevelMeta.
Since : 1.20
GST_AUDIO_META_API_TYPE
#define GST_AUDIO_META_API_TYPE (gst_audio_meta_api_get_type())
GST_AUDIO_META_INFO
#define GST_AUDIO_META_INFO (gst_audio_meta_get_info())
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