GstAudioDecoder
This base class is for audio decoders turning encoded data into raw audio samples.
GstAudioDecoder and subclass should cooperate as follows.
Configuration
- Initially, GstAudioDecoder calls start when the decoder element is activated, which allows subclass to perform any global setup. Base class (context) parameters can already be set according to subclass capabilities (or possibly upon receive more information in subsequent set_format).
- GstAudioDecoder calls set_format to inform subclass of the format of input audio data that it is about to receive. While unlikely, it might be called more than once, if changing input parameters require reconfiguration.
- GstAudioDecoder calls stop at end of all processing.
As of configuration stage, and throughout processing, GstAudioDecoder provides various (context) parameters, e.g. describing the format of output audio data (valid when output caps have been set) or current parsing state. Conversely, subclass can and should configure context to inform base class of its expectation w.r.t. buffer handling.
Data processing
* Base class gathers input data, and optionally allows subclass
to parse this into subsequently manageable (as defined by subclass)
chunks. Such chunks are subsequently referred to as 'frames',
though they may or may not correspond to 1 (or more) audio format frame.
* Input frame is provided to subclass' @handle_frame.
* If codec processing results in decoded data, subclass should call
@gst_audio_decoder_finish_frame to have decoded data pushed
downstream.
* Just prior to actually pushing a buffer downstream,
it is passed to @pre_push. Subclass should either use this callback
to arrange for additional downstream pushing or otherwise ensure such
custom pushing occurs after at least a method call has finished since
setting src pad caps.
* During the parsing process GstAudioDecoderClass will handle both
srcpad and sinkpad events. Sink events will be passed to subclass
if @event callback has been provided.
Shutdown phase
- GstAudioDecoder class calls stop to inform the subclass that data parsing will be stopped.
Subclass is responsible for providing pad template caps for source and sink pads. The pads need to be named "sink" and "src". It also needs to set the fixed caps on srcpad, when the format is ensured. This is typically when base class calls subclass' set_format function, though it might be delayed until calling gst_audio_decoder_finish_frame.
In summary, above process should have subclass concentrating on codec data processing while leaving other matters to base class, such as most notably timestamp handling. While it may exert more control in this area (see e.g. pre_push), it is very much not recommended.
In particular, base class will try to arrange for perfect output timestamps as much as possible while tracking upstream timestamps. To this end, if deviation between the next ideal expected perfect timestamp and upstream exceeds tolerance, then resync to upstream occurs (which would happen always if the tolerance mechanism is disabled).
In non-live pipelines, baseclass can also (configurably) arrange for output buffer aggregation which may help to redue large(r) numbers of small(er) buffers being pushed and processed downstream. Note that this feature is only available if the buffer layout is interleaved. For planar buffers, the decoder implementation is fully responsible for the output buffer size.
On the other hand, it should be noted that baseclass only provides limited seeking support (upon explicit subclass request), as full-fledged support should rather be left to upstream demuxer, parser or alike. This simple approach caters for seeking and duration reporting using estimated input bitrates.
Things that subclass need to take care of:
-
Provide pad templates
-
Set source pad caps when appropriate
-
Set user-configurable properties to sane defaults for format and implementing codec at hand, and convey some subclass capabilities and expectations in context.
-
Accept data in handle_frame and provide encoded results to gst_audio_decoder_finish_frame. If it is prepared to perform PLC, it should also accept NULL data in handle_frame and provide for data for indicated duration.
GstAudioDecoder
GObject ╰──GInitiallyUnowned ╰──GstObject ╰──GstElement ╰──GstAudioDecoder
The opaque GstAudioDecoder data structure.
Members
element
(GstElement)
–
sinkpad
(GstPad *)
–
srcpad
(GstPad *)
–
stream_lock
(GRecMutex)
–
input_segment
(GstSegment)
–
output_segment
(GstSegment)
–
Class structure
GstAudioDecoderClass
Subclasses can override any of the available virtual methods or not, as needed. At minimum handle_frame (and likely set_format) needs to be overridden.
Fields
element_class
(GstElementClass)
–
The parent class structure
GstAudio.AudioDecoderClass
Subclasses can override any of the available virtual methods or not, as needed. At minimum handle_frame (and likely set_format) needs to be overridden.
Attributes
element_class
(Gst.ElementClass)
–
The parent class structure
GstAudio.AudioDecoderClass
Subclasses can override any of the available virtual methods or not, as needed. At minimum handle_frame (and likely set_format) needs to be overridden.
Attributes
element_class
(Gst.ElementClass)
–
The parent class structure
GstAudio.AudioDecoder
GObject.Object ╰──GObject.InitiallyUnowned ╰──Gst.Object ╰──Gst.Element ╰──GstAudio.AudioDecoder
The opaque GstAudio.AudioDecoder data structure.
Members
element
(Gst.Element)
–
sinkpad
(Gst.Pad)
–
srcpad
(Gst.Pad)
–
stream_lock
(GLib.RecMutex)
–
input_segment
(Gst.Segment)
–
output_segment
(Gst.Segment)
–
GstAudio.AudioDecoder
GObject.Object ╰──GObject.InitiallyUnowned ╰──Gst.Object ╰──Gst.Element ╰──GstAudio.AudioDecoder
The opaque GstAudio.AudioDecoder data structure.
Members
element
(Gst.Element)
–
sinkpad
(Gst.Pad)
–
srcpad
(Gst.Pad)
–
stream_lock
(GLib.RecMutex)
–
input_segment
(Gst.Segment)
–
output_segment
(Gst.Segment)
–
Methods
gst_audio_decoder_allocate_output_buffer
GstBuffer * gst_audio_decoder_allocate_output_buffer (GstAudioDecoder * dec, gsize size)
Helper function that allocates a buffer to hold an audio frame for dec's current output format.
allocated buffer
GstAudio.AudioDecoder.prototype.allocate_output_buffer
function GstAudio.AudioDecoder.prototype.allocate_output_buffer(size: Number): {
// javascript wrapper for 'gst_audio_decoder_allocate_output_buffer'
}
Helper function that allocates a buffer to hold an audio frame for dec's current output format.
Parameters:
size of the buffer
allocated buffer
GstAudio.AudioDecoder.allocate_output_buffer
def GstAudio.AudioDecoder.allocate_output_buffer (self, size):
#python wrapper for 'gst_audio_decoder_allocate_output_buffer'
Helper function that allocates a buffer to hold an audio frame for dec's current output format.
allocated buffer
gst_audio_decoder_finish_frame
GstFlowReturn gst_audio_decoder_finish_frame (GstAudioDecoder * dec, GstBuffer * buf, gint frames)
Collects decoded data and pushes it downstream.
buf may be NULL in which case the indicated number of frames are discarded and considered to have produced no output (e.g. lead-in or setup frames). Otherwise, source pad caps must be set when it is called with valid data in buf.
Note that a frame received in handle_frame() may be invalidated by a call to this function.
Parameters:
dec
–
buf
(
[transfer: full][nullable])
–
decoded data
frames
–
number of decoded frames represented by decoded data
a GstFlowReturn that should be escalated to caller (of caller)
GstAudio.AudioDecoder.prototype.finish_frame
function GstAudio.AudioDecoder.prototype.finish_frame(buf: Gst.Buffer, frames: Number): {
// javascript wrapper for 'gst_audio_decoder_finish_frame'
}
Collects decoded data and pushes it downstream.
buf may be NULL in which case the indicated number of frames are discarded and considered to have produced no output (e.g. lead-in or setup frames). Otherwise, source pad caps must be set when it is called with valid data in buf.
Note that a frame received in vfunc_handle_frame() may be invalidated by a call to this function.
Parameters:
decoded data
number of decoded frames represented by decoded data
a Gst.FlowReturn that should be escalated to caller (of caller)
GstAudio.AudioDecoder.finish_frame
def GstAudio.AudioDecoder.finish_frame (self, buf, frames):
#python wrapper for 'gst_audio_decoder_finish_frame'
Collects decoded data and pushes it downstream.
buf may be NULL in which case the indicated number of frames are discarded and considered to have produced no output (e.g. lead-in or setup frames). Otherwise, source pad caps must be set when it is called with valid data in buf.
Note that a frame received in do_handle_frame() may be invalidated by a call to this function.
Parameters:
decoded data
number of decoded frames represented by decoded data
a Gst.FlowReturn that should be escalated to caller (of caller)
gst_audio_decoder_finish_subframe
GstFlowReturn gst_audio_decoder_finish_subframe (GstAudioDecoder * dec, GstBuffer * buf)
Collects decoded data and pushes it downstream. This function may be called multiple times for a given input frame.
buf may be NULL in which case it is assumed that the current input frame is finished. This is equivalent to calling gst_audio_decoder_finish_subframe with a NULL buffer and frames=1 after having pushed out all decoded audio subframes using this function.
When called with valid data in buf the source pad caps must have been set already.
Note that a frame received in handle_frame() may be invalidated by a call to this function.
a GstFlowReturn that should be escalated to caller (of caller)
Since : 1.16
GstAudio.AudioDecoder.prototype.finish_subframe
function GstAudio.AudioDecoder.prototype.finish_subframe(buf: Gst.Buffer): {
// javascript wrapper for 'gst_audio_decoder_finish_subframe'
}
Collects decoded data and pushes it downstream. This function may be called multiple times for a given input frame.
buf may be NULL in which case it is assumed that the current input frame is finished. This is equivalent to calling GstAudio.AudioDecoder.prototype.finish_subframe with a NULL buffer and frames=1 after having pushed out all decoded audio subframes using this function.
When called with valid data in buf the source pad caps must have been set already.
Note that a frame received in vfunc_handle_frame() may be invalidated by a call to this function.
a Gst.FlowReturn that should be escalated to caller (of caller)
Since : 1.16
GstAudio.AudioDecoder.finish_subframe
def GstAudio.AudioDecoder.finish_subframe (self, buf):
#python wrapper for 'gst_audio_decoder_finish_subframe'
Collects decoded data and pushes it downstream. This function may be called multiple times for a given input frame.
buf may be NULL in which case it is assumed that the current input frame is finished. This is equivalent to calling GstAudio.AudioDecoder.finish_subframe with a NULL buffer and frames=1 after having pushed out all decoded audio subframes using this function.
When called with valid data in buf the source pad caps must have been set already.
Note that a frame received in do_handle_frame() may be invalidated by a call to this function.
a Gst.FlowReturn that should be escalated to caller (of caller)
Since : 1.16
gst_audio_decoder_get_allocator
gst_audio_decoder_get_allocator (GstAudioDecoder * dec, GstAllocator ** allocator, GstAllocationParams * params)
Lets GstAudioDecoder sub-classes to know the memory allocator used by the base class and its params.
Unref the allocator after use it.
Parameters:
dec
–
allocator
(
[out][optional][nullable][transfer: full])
–
the GstAllocator used
params
(
[out][optional][transfer: full])
–
the GstAllocationParams of allocator
GstAudio.AudioDecoder.prototype.get_allocator
function GstAudio.AudioDecoder.prototype.get_allocator(): {
// javascript wrapper for 'gst_audio_decoder_get_allocator'
}
Lets GstAudio.AudioDecoder sub-classes to know the memory allocator used by the base class and its params.
Unref the allocator after use it.
Parameters:
GstAudio.AudioDecoder.get_allocator
def GstAudio.AudioDecoder.get_allocator (self):
#python wrapper for 'gst_audio_decoder_get_allocator'
Lets GstAudio.AudioDecoder sub-classes to know the memory allocator used by the base class and its params.
Unref the allocator after use it.
Parameters:
gst_audio_decoder_get_audio_info
GstAudioInfo * gst_audio_decoder_get_audio_info (GstAudioDecoder * dec)
Parameters:
dec
–
a GstAudioInfo describing the input audio format
GstAudio.AudioDecoder.prototype.get_audio_info
function GstAudio.AudioDecoder.prototype.get_audio_info(): {
// javascript wrapper for 'gst_audio_decoder_get_audio_info'
}
Parameters:
a GstAudio.AudioInfo describing the input audio format
GstAudio.AudioDecoder.get_audio_info
def GstAudio.AudioDecoder.get_audio_info (self):
#python wrapper for 'gst_audio_decoder_get_audio_info'
Parameters:
a GstAudio.AudioInfo describing the input audio format
gst_audio_decoder_get_delay
gint gst_audio_decoder_get_delay (GstAudioDecoder * dec)
Parameters:
dec
–
currently configured decoder delay
GstAudio.AudioDecoder.prototype.get_delay
function GstAudio.AudioDecoder.prototype.get_delay(): {
// javascript wrapper for 'gst_audio_decoder_get_delay'
}
Parameters:
currently configured decoder delay
GstAudio.AudioDecoder.get_delay
def GstAudio.AudioDecoder.get_delay (self):
#python wrapper for 'gst_audio_decoder_get_delay'
Parameters:
currently configured decoder delay
gst_audio_decoder_get_drainable
gboolean gst_audio_decoder_get_drainable (GstAudioDecoder * dec)
Queries decoder drain handling.
Parameters:
dec
–
TRUE if drainable handling is enabled.
MT safe.
GstAudio.AudioDecoder.prototype.get_drainable
function GstAudio.AudioDecoder.prototype.get_drainable(): {
// javascript wrapper for 'gst_audio_decoder_get_drainable'
}
Queries decoder drain handling.
Parameters:
GstAudio.AudioDecoder.get_drainable
def GstAudio.AudioDecoder.get_drainable (self):
#python wrapper for 'gst_audio_decoder_get_drainable'
Queries decoder drain handling.
Parameters:
gst_audio_decoder_get_estimate_rate
gint gst_audio_decoder_get_estimate_rate (GstAudioDecoder * dec)
Parameters:
dec
–
currently configured byte to time conversion setting
GstAudio.AudioDecoder.prototype.get_estimate_rate
function GstAudio.AudioDecoder.prototype.get_estimate_rate(): {
// javascript wrapper for 'gst_audio_decoder_get_estimate_rate'
}
Parameters:
currently configured byte to time conversion setting
GstAudio.AudioDecoder.get_estimate_rate
def GstAudio.AudioDecoder.get_estimate_rate (self):
#python wrapper for 'gst_audio_decoder_get_estimate_rate'
Parameters:
currently configured byte to time conversion setting
gst_audio_decoder_get_latency
gst_audio_decoder_get_latency (GstAudioDecoder * dec, GstClockTime * min, GstClockTime * max)
Sets the variables pointed to by min and max to the currently configured latency.
Parameters:
dec
–
min
(
[out][optional])
–
a pointer to storage to hold minimum latency
max
(
[out][optional])
–
a pointer to storage to hold maximum latency
GstAudio.AudioDecoder.prototype.get_latency
function GstAudio.AudioDecoder.prototype.get_latency(): {
// javascript wrapper for 'gst_audio_decoder_get_latency'
}
Sets the variables pointed to by min and max to the currently configured latency.
Parameters:
GstAudio.AudioDecoder.get_latency
def GstAudio.AudioDecoder.get_latency (self):
#python wrapper for 'gst_audio_decoder_get_latency'
Sets the variables pointed to by min and max to the currently configured latency.
Parameters:
gst_audio_decoder_get_max_errors
gint gst_audio_decoder_get_max_errors (GstAudioDecoder * dec)
Parameters:
dec
–
currently configured decoder tolerated error count.
GstAudio.AudioDecoder.prototype.get_max_errors
function GstAudio.AudioDecoder.prototype.get_max_errors(): {
// javascript wrapper for 'gst_audio_decoder_get_max_errors'
}
Parameters:
currently configured decoder tolerated error count.
GstAudio.AudioDecoder.get_max_errors
def GstAudio.AudioDecoder.get_max_errors (self):
#python wrapper for 'gst_audio_decoder_get_max_errors'
Parameters:
currently configured decoder tolerated error count.
gst_audio_decoder_get_min_latency
GstClockTime gst_audio_decoder_get_min_latency (GstAudioDecoder * dec)
Queries decoder's latency aggregation.
Parameters:
dec
–
aggregation latency.
MT safe.
GstAudio.AudioDecoder.prototype.get_min_latency
function GstAudio.AudioDecoder.prototype.get_min_latency(): {
// javascript wrapper for 'gst_audio_decoder_get_min_latency'
}
Queries decoder's latency aggregation.
Parameters:
GstAudio.AudioDecoder.get_min_latency
def GstAudio.AudioDecoder.get_min_latency (self):
#python wrapper for 'gst_audio_decoder_get_min_latency'
Queries decoder's latency aggregation.
Parameters:
gst_audio_decoder_get_needs_format
gboolean gst_audio_decoder_get_needs_format (GstAudioDecoder * dec)
Queries decoder required format handling.
Parameters:
dec
–
TRUE if required format handling is enabled.
MT safe.
GstAudio.AudioDecoder.prototype.get_needs_format
function GstAudio.AudioDecoder.prototype.get_needs_format(): {
// javascript wrapper for 'gst_audio_decoder_get_needs_format'
}
Queries decoder required format handling.
Parameters:
GstAudio.AudioDecoder.get_needs_format
def GstAudio.AudioDecoder.get_needs_format (self):
#python wrapper for 'gst_audio_decoder_get_needs_format'
Queries decoder required format handling.
Parameters:
gst_audio_decoder_get_parse_state
gst_audio_decoder_get_parse_state (GstAudioDecoder * dec, gboolean * sync, gboolean * eos)
Return current parsing (sync and eos) state.
Parameters:
dec
–
sync
(
[out][optional])
–
a pointer to a variable to hold the current sync state
eos
(
[out][optional])
–
a pointer to a variable to hold the current eos state
GstAudio.AudioDecoder.prototype.get_parse_state
function GstAudio.AudioDecoder.prototype.get_parse_state(): {
// javascript wrapper for 'gst_audio_decoder_get_parse_state'
}
Return current parsing (sync and eos) state.
Parameters:
GstAudio.AudioDecoder.get_parse_state
def GstAudio.AudioDecoder.get_parse_state (self):
#python wrapper for 'gst_audio_decoder_get_parse_state'
Return current parsing (sync and eos) state.
Parameters:
gst_audio_decoder_get_plc
gboolean gst_audio_decoder_get_plc (GstAudioDecoder * dec)
Queries decoder packet loss concealment handling.
Parameters:
dec
–
TRUE if packet loss concealment is enabled.
MT safe.
GstAudio.AudioDecoder.prototype.get_plc
function GstAudio.AudioDecoder.prototype.get_plc(): {
// javascript wrapper for 'gst_audio_decoder_get_plc'
}
Queries decoder packet loss concealment handling.
Parameters:
GstAudio.AudioDecoder.get_plc
def GstAudio.AudioDecoder.get_plc (self):
#python wrapper for 'gst_audio_decoder_get_plc'
Queries decoder packet loss concealment handling.
Parameters:
gst_audio_decoder_get_plc_aware
gint gst_audio_decoder_get_plc_aware (GstAudioDecoder * dec)
Parameters:
dec
–
currently configured plc handling
GstAudio.AudioDecoder.prototype.get_plc_aware
function GstAudio.AudioDecoder.prototype.get_plc_aware(): {
// javascript wrapper for 'gst_audio_decoder_get_plc_aware'
}
Parameters:
currently configured plc handling
GstAudio.AudioDecoder.get_plc_aware
def GstAudio.AudioDecoder.get_plc_aware (self):
#python wrapper for 'gst_audio_decoder_get_plc_aware'
Parameters:
currently configured plc handling
gst_audio_decoder_get_tolerance
GstClockTime gst_audio_decoder_get_tolerance (GstAudioDecoder * dec)
Queries current audio jitter tolerance threshold.
Parameters:
dec
–
decoder audio jitter tolerance threshold.
MT safe.
GstAudio.AudioDecoder.prototype.get_tolerance
function GstAudio.AudioDecoder.prototype.get_tolerance(): {
// javascript wrapper for 'gst_audio_decoder_get_tolerance'
}
Queries current audio jitter tolerance threshold.
Parameters:
GstAudio.AudioDecoder.get_tolerance
def GstAudio.AudioDecoder.get_tolerance (self):
#python wrapper for 'gst_audio_decoder_get_tolerance'
Queries current audio jitter tolerance threshold.
Parameters:
gst_audio_decoder_merge_tags
gst_audio_decoder_merge_tags (GstAudioDecoder * dec, const GstTagList * tags, GstTagMergeMode mode)
Sets the audio decoder tags and how they should be merged with any upstream stream tags. This will override any tags previously-set with gst_audio_decoder_merge_tags.
Note that this is provided for convenience, and the subclass is not required to use this and can still do tag handling on its own.
Parameters:
dec
–
tags
(
[nullable])
–
a GstTagList to merge, or NULL
mode
–
the GstTagMergeMode to use, usually GST_TAG_MERGE_REPLACE
GstAudio.AudioDecoder.prototype.merge_tags
function GstAudio.AudioDecoder.prototype.merge_tags(tags: Gst.TagList, mode: Gst.TagMergeMode): {
// javascript wrapper for 'gst_audio_decoder_merge_tags'
}
Sets the audio decoder tags and how they should be merged with any upstream stream tags. This will override any tags previously-set with GstAudio.AudioDecoder.prototype.merge_tags.
Note that this is provided for convenience, and the subclass is not required to use this and can still do tag handling on its own.
Parameters:
a Gst.TagList to merge, or NULL
the Gst.TagMergeMode to use, usually Gst.TagMergeMode.REPLACE
GstAudio.AudioDecoder.merge_tags
def GstAudio.AudioDecoder.merge_tags (self, tags, mode):
#python wrapper for 'gst_audio_decoder_merge_tags'
Sets the audio decoder tags and how they should be merged with any upstream stream tags. This will override any tags previously-set with GstAudio.AudioDecoder.merge_tags.
Note that this is provided for convenience, and the subclass is not required to use this and can still do tag handling on its own.
Parameters:
a Gst.TagList to merge, or NULL
the Gst.TagMergeMode to use, usually Gst.TagMergeMode.REPLACE
gst_audio_decoder_negotiate
gboolean gst_audio_decoder_negotiate (GstAudioDecoder * dec)
Negotiate with downstream elements to currently configured GstAudioInfo. Unmark GST_PAD_FLAG_NEED_RECONFIGURE in any case. But mark it again if negotiate fails.
Parameters:
dec
–
GstAudio.AudioDecoder.prototype.negotiate
function GstAudio.AudioDecoder.prototype.negotiate(): {
// javascript wrapper for 'gst_audio_decoder_negotiate'
}
Negotiate with downstream elements to currently configured GstAudio.AudioInfo. Unmark GST_PAD_FLAG_NEED_RECONFIGURE in any case. But mark it again if negotiate fails.
Parameters:
GstAudio.AudioDecoder.negotiate
def GstAudio.AudioDecoder.negotiate (self):
#python wrapper for 'gst_audio_decoder_negotiate'
Negotiate with downstream elements to currently configured GstAudio.AudioInfo. Unmark GST_PAD_FLAG_NEED_RECONFIGURE in any case. But mark it again if negotiate fails.
Parameters:
gst_audio_decoder_proxy_getcaps
GstCaps * gst_audio_decoder_proxy_getcaps (GstAudioDecoder * decoder, GstCaps * caps, GstCaps * filter)
Returns caps that express caps (or sink template caps if caps == NULL) restricted to rate/channels/... combinations supported by downstream elements.
Parameters:
decoder
–
caps
(
[nullable])
–
initial caps
filter
(
[nullable])
–
filter caps
a GstCaps owned by caller
Since : 1.6
GstAudio.AudioDecoder.prototype.proxy_getcaps
function GstAudio.AudioDecoder.prototype.proxy_getcaps(caps: Gst.Caps, filter: Gst.Caps): {
// javascript wrapper for 'gst_audio_decoder_proxy_getcaps'
}
Returns caps that express caps (or sink template caps if caps == NULL) restricted to rate/channels/... combinations supported by downstream elements.
Parameters:
initial caps
filter caps
Since : 1.6
GstAudio.AudioDecoder.proxy_getcaps
def GstAudio.AudioDecoder.proxy_getcaps (self, caps, filter):
#python wrapper for 'gst_audio_decoder_proxy_getcaps'
Returns caps that express caps (or sink template caps if caps == NULL) restricted to rate/channels/... combinations supported by downstream elements.
Parameters:
initial caps
filter caps
Since : 1.6
gst_audio_decoder_set_allocation_caps
gst_audio_decoder_set_allocation_caps (GstAudioDecoder * dec, GstCaps * allocation_caps)
Sets a caps in allocation query which are different from the set pad's caps. Use this function before calling gst_audio_decoder_negotiate. Setting to NULL the allocation query will use the caps from the pad.
Since : 1.10
GstAudio.AudioDecoder.prototype.set_allocation_caps
function GstAudio.AudioDecoder.prototype.set_allocation_caps(allocation_caps: Gst.Caps): {
// javascript wrapper for 'gst_audio_decoder_set_allocation_caps'
}
Sets a caps in allocation query which are different from the set pad's caps. Use this function before calling GstAudio.AudioDecoder.prototype.negotiate. Setting to null the allocation query will use the caps from the pad.
Parameters:
Since : 1.10
GstAudio.AudioDecoder.set_allocation_caps
def GstAudio.AudioDecoder.set_allocation_caps (self, allocation_caps):
#python wrapper for 'gst_audio_decoder_set_allocation_caps'
Sets a caps in allocation query which are different from the set pad's caps. Use this function before calling GstAudio.AudioDecoder.negotiate. Setting to None the allocation query will use the caps from the pad.
Parameters:
Since : 1.10
gst_audio_decoder_set_drainable
gst_audio_decoder_set_drainable (GstAudioDecoder * dec, gboolean enabled)
Configures decoder drain handling. If drainable, subclass might be handed a NULL buffer to have it return any leftover decoded data. Otherwise, it is not considered so capable and will only ever be passed real data.
MT safe.
GstAudio.AudioDecoder.prototype.set_drainable
function GstAudio.AudioDecoder.prototype.set_drainable(enabled: Number): {
// javascript wrapper for 'gst_audio_decoder_set_drainable'
}
Configures decoder drain handling. If drainable, subclass might be handed a NULL buffer to have it return any leftover decoded data. Otherwise, it is not considered so capable and will only ever be passed real data.
MT safe.
GstAudio.AudioDecoder.set_drainable
def GstAudio.AudioDecoder.set_drainable (self, enabled):
#python wrapper for 'gst_audio_decoder_set_drainable'
Configures decoder drain handling. If drainable, subclass might be handed a NULL buffer to have it return any leftover decoded data. Otherwise, it is not considered so capable and will only ever be passed real data.
MT safe.
gst_audio_decoder_set_estimate_rate
gst_audio_decoder_set_estimate_rate (GstAudioDecoder * dec, gboolean enabled)
Allows baseclass to perform byte to time estimated conversion.
GstAudio.AudioDecoder.prototype.set_estimate_rate
function GstAudio.AudioDecoder.prototype.set_estimate_rate(enabled: Number): {
// javascript wrapper for 'gst_audio_decoder_set_estimate_rate'
}
Allows baseclass to perform byte to time estimated conversion.
Parameters:
whether to enable byte to time conversion
GstAudio.AudioDecoder.set_estimate_rate
def GstAudio.AudioDecoder.set_estimate_rate (self, enabled):
#python wrapper for 'gst_audio_decoder_set_estimate_rate'
Allows baseclass to perform byte to time estimated conversion.
Parameters:
whether to enable byte to time conversion
gst_audio_decoder_set_latency
gst_audio_decoder_set_latency (GstAudioDecoder * dec, GstClockTime min, GstClockTime max)
Sets decoder latency. If the provided values changed from previously provided ones, this will also post a LATENCY message on the bus so the pipeline can reconfigure its global latency.
GstAudio.AudioDecoder.prototype.set_latency
function GstAudio.AudioDecoder.prototype.set_latency(min: Number, max: Number): {
// javascript wrapper for 'gst_audio_decoder_set_latency'
}
Sets decoder latency. If the provided values changed from previously provided ones, this will also post a LATENCY message on the bus so the pipeline can reconfigure its global latency.
Parameters:
minimum latency
maximum latency
GstAudio.AudioDecoder.set_latency
def GstAudio.AudioDecoder.set_latency (self, min, max):
#python wrapper for 'gst_audio_decoder_set_latency'
Sets decoder latency. If the provided values changed from previously provided ones, this will also post a LATENCY message on the bus so the pipeline can reconfigure its global latency.
Parameters:
minimum latency
maximum latency
gst_audio_decoder_set_max_errors
gst_audio_decoder_set_max_errors (GstAudioDecoder * dec, gint num)
Sets numbers of tolerated decoder errors, where a tolerated one is then only warned about, but more than tolerated will lead to fatal error. You can set -1 for never returning fatal errors. Default is set to GST_AUDIO_DECODER_MAX_ERRORS.
GstAudio.AudioDecoder.prototype.set_max_errors
function GstAudio.AudioDecoder.prototype.set_max_errors(num: Number): {
// javascript wrapper for 'gst_audio_decoder_set_max_errors'
}
Sets numbers of tolerated decoder errors, where a tolerated one is then only warned about, but more than tolerated will lead to fatal error. You can set -1 for never returning fatal errors. Default is set to GST_AUDIO_DECODER_MAX_ERRORS.
Parameters:
max tolerated errors
GstAudio.AudioDecoder.set_max_errors
def GstAudio.AudioDecoder.set_max_errors (self, num):
#python wrapper for 'gst_audio_decoder_set_max_errors'
Sets numbers of tolerated decoder errors, where a tolerated one is then only warned about, but more than tolerated will lead to fatal error. You can set -1 for never returning fatal errors. Default is set to GST_AUDIO_DECODER_MAX_ERRORS.
gst_audio_decoder_set_min_latency
gst_audio_decoder_set_min_latency (GstAudioDecoder * dec, GstClockTime num)
Sets decoder minimum aggregation latency.
MT safe.
GstAudio.AudioDecoder.prototype.set_min_latency
function GstAudio.AudioDecoder.prototype.set_min_latency(num: Number): {
// javascript wrapper for 'gst_audio_decoder_set_min_latency'
}
Sets decoder minimum aggregation latency.
MT safe.
Parameters:
new minimum latency
GstAudio.AudioDecoder.set_min_latency
def GstAudio.AudioDecoder.set_min_latency (self, num):
#python wrapper for 'gst_audio_decoder_set_min_latency'
Sets decoder minimum aggregation latency.
MT safe.
gst_audio_decoder_set_needs_format
gst_audio_decoder_set_needs_format (GstAudioDecoder * dec, gboolean enabled)
Configures decoder format needs. If enabled, subclass needs to be negotiated with format caps before it can process any data. It will then never be handed any data before it has been configured. Otherwise, it might be handed data without having been configured and is then expected being able to do so either by default or based on the input data.
MT safe.
GstAudio.AudioDecoder.prototype.set_needs_format
function GstAudio.AudioDecoder.prototype.set_needs_format(enabled: Number): {
// javascript wrapper for 'gst_audio_decoder_set_needs_format'
}
Configures decoder format needs. If enabled, subclass needs to be negotiated with format caps before it can process any data. It will then never be handed any data before it has been configured. Otherwise, it might be handed data without having been configured and is then expected being able to do so either by default or based on the input data.
MT safe.
GstAudio.AudioDecoder.set_needs_format
def GstAudio.AudioDecoder.set_needs_format (self, enabled):
#python wrapper for 'gst_audio_decoder_set_needs_format'
Configures decoder format needs. If enabled, subclass needs to be negotiated with format caps before it can process any data. It will then never be handed any data before it has been configured. Otherwise, it might be handed data without having been configured and is then expected being able to do so either by default or based on the input data.
MT safe.
gst_audio_decoder_set_output_caps
gboolean gst_audio_decoder_set_output_caps (GstAudioDecoder * dec, GstCaps * caps)
Configure output caps on the srcpad of dec. Similar to gst_audio_decoder_set_output_format, but allows subclasses to specify output caps that can't be expressed via GstAudioInfo e.g. caps that have caps features.
TRUE on success.
Since : 1.16
GstAudio.AudioDecoder.prototype.set_output_caps
function GstAudio.AudioDecoder.prototype.set_output_caps(caps: Gst.Caps): {
// javascript wrapper for 'gst_audio_decoder_set_output_caps'
}
Configure output caps on the srcpad of dec. Similar to GstAudio.AudioDecoder.prototype.set_output_format, but allows subclasses to specify output caps that can't be expressed via GstAudio.AudioInfo e.g. caps that have caps features.
Parameters:
Since : 1.16
GstAudio.AudioDecoder.set_output_caps
def GstAudio.AudioDecoder.set_output_caps (self, caps):
#python wrapper for 'gst_audio_decoder_set_output_caps'
Configure output caps on the srcpad of dec. Similar to GstAudio.AudioDecoder.set_output_format, but allows subclasses to specify output caps that can't be expressed via GstAudio.AudioInfo e.g. caps that have caps features.
Parameters:
Since : 1.16
gst_audio_decoder_set_output_format
gboolean gst_audio_decoder_set_output_format (GstAudioDecoder * dec, const GstAudioInfo * info)
Configure output info on the srcpad of dec.
TRUE on success.
GstAudio.AudioDecoder.prototype.set_output_format
function GstAudio.AudioDecoder.prototype.set_output_format(info: GstAudio.AudioInfo): {
// javascript wrapper for 'gst_audio_decoder_set_output_format'
}
Configure output info on the srcpad of dec.
Parameters:
GstAudio.AudioDecoder.set_output_format
def GstAudio.AudioDecoder.set_output_format (self, info):
#python wrapper for 'gst_audio_decoder_set_output_format'
Configure output info on the srcpad of dec.
Parameters:
gst_audio_decoder_set_plc
gst_audio_decoder_set_plc (GstAudioDecoder * dec, gboolean enabled)
Enable or disable decoder packet loss concealment, provided subclass and codec are capable and allow handling plc.
MT safe.
GstAudio.AudioDecoder.prototype.set_plc
function GstAudio.AudioDecoder.prototype.set_plc(enabled: Number): {
// javascript wrapper for 'gst_audio_decoder_set_plc'
}
Enable or disable decoder packet loss concealment, provided subclass and codec are capable and allow handling plc.
MT safe.
GstAudio.AudioDecoder.set_plc
def GstAudio.AudioDecoder.set_plc (self, enabled):
#python wrapper for 'gst_audio_decoder_set_plc'
Enable or disable decoder packet loss concealment, provided subclass and codec are capable and allow handling plc.
MT safe.
gst_audio_decoder_set_plc_aware
gst_audio_decoder_set_plc_aware (GstAudioDecoder * dec, gboolean plc)
Indicates whether or not subclass handles packet loss concealment (plc).
GstAudio.AudioDecoder.prototype.set_plc_aware
function GstAudio.AudioDecoder.prototype.set_plc_aware(plc: Number): {
// javascript wrapper for 'gst_audio_decoder_set_plc_aware'
}
Indicates whether or not subclass handles packet loss concealment (plc).
GstAudio.AudioDecoder.set_plc_aware
def GstAudio.AudioDecoder.set_plc_aware (self, plc):
#python wrapper for 'gst_audio_decoder_set_plc_aware'
Indicates whether or not subclass handles packet loss concealment (plc).
gst_audio_decoder_set_tolerance
gst_audio_decoder_set_tolerance (GstAudioDecoder * dec, GstClockTime tolerance)
Configures decoder audio jitter tolerance threshold.
MT safe.
GstAudio.AudioDecoder.prototype.set_tolerance
function GstAudio.AudioDecoder.prototype.set_tolerance(tolerance: Number): {
// javascript wrapper for 'gst_audio_decoder_set_tolerance'
}
Configures decoder audio jitter tolerance threshold.
MT safe.
Parameters:
new tolerance
GstAudio.AudioDecoder.set_tolerance
def GstAudio.AudioDecoder.set_tolerance (self, tolerance):
#python wrapper for 'gst_audio_decoder_set_tolerance'
Configures decoder audio jitter tolerance threshold.
MT safe.
gst_audio_decoder_set_use_default_pad_acceptcaps
gst_audio_decoder_set_use_default_pad_acceptcaps (GstAudioDecoder * decoder, gboolean use)
Lets GstAudioDecoder sub-classes decide if they want the sink pad to use the default pad query handler to reply to accept-caps queries.
By setting this to true it is possible to further customize the default handler with GST_PAD_SET_ACCEPT_INTERSECT and GST_PAD_SET_ACCEPT_TEMPLATE
Parameters:
decoder
–
use
–
if the default pad accept-caps query handling should be used
Since : 1.6
GstAudio.AudioDecoder.prototype.set_use_default_pad_acceptcaps
function GstAudio.AudioDecoder.prototype.set_use_default_pad_acceptcaps(use: Number): {
// javascript wrapper for 'gst_audio_decoder_set_use_default_pad_acceptcaps'
}
Lets GstAudio.AudioDecoder sub-classes decide if they want the sink pad to use the default pad query handler to reply to accept-caps queries.
By setting this to true it is possible to further customize the default handler with GST_PAD_SET_ACCEPT_INTERSECT (not introspectable) and GST_PAD_SET_ACCEPT_TEMPLATE (not introspectable)
Parameters:
if the default pad accept-caps query handling should be used
Since : 1.6
GstAudio.AudioDecoder.set_use_default_pad_acceptcaps
def GstAudio.AudioDecoder.set_use_default_pad_acceptcaps (self, use):
#python wrapper for 'gst_audio_decoder_set_use_default_pad_acceptcaps'
Lets GstAudio.AudioDecoder sub-classes decide if they want the sink pad to use the default pad query handler to reply to accept-caps queries.
By setting this to true it is possible to further customize the default handler with GST_PAD_SET_ACCEPT_INTERSECT (not introspectable) and GST_PAD_SET_ACCEPT_TEMPLATE (not introspectable)
Parameters:
if the default pad accept-caps query handling should be used
Since : 1.6
Properties
max-errors
“max-errors” gint
Maximum number of tolerated consecutive decode errors. See gst_audio_decoder_set_max_errors for more details.
Flags : Read / Write
Since : 1.18
max-errors
“max-errors” Number
Maximum number of tolerated consecutive decode errors. See GstAudio.AudioDecoder.prototype.set_max_errors for more details.
Flags : Read / Write
Since : 1.18
max_errors
“self.props.max_errors” int
Maximum number of tolerated consecutive decode errors. See GstAudio.AudioDecoder.set_max_errors for more details.
Flags : Read / Write
Since : 1.18
Virtual Methods
close
gboolean close (GstAudioDecoder * dec)
Optional. Called when the element changes to GST_STATE_NULL. Allows closing external resources.
Parameters:
dec
–
vfunc_close
function vfunc_close(dec: GstAudio.AudioDecoder): {
// javascript implementation of the 'close' virtual method
}
Optional. Called when the element changes to GST_STATE_NULL. Allows closing external resources.
Parameters:
do_close
def do_close (dec):
#python implementation of the 'close' virtual method
Optional. Called when the element changes to GST_STATE_NULL. Allows closing external resources.
Parameters:
decide_allocation
gboolean decide_allocation (GstAudioDecoder * dec, GstQuery * query)
Optional. Setup the allocation parameters for allocating output buffers. The passed in query contains the result of the downstream allocation query. Subclasses should chain up to the parent implementation to invoke the default handler.
Parameters:
dec
–
query
–
vfunc_decide_allocation
function vfunc_decide_allocation(dec: GstAudio.AudioDecoder, query: Gst.Query): {
// javascript implementation of the 'decide_allocation' virtual method
}
Optional. Setup the allocation parameters for allocating output buffers. The passed in query contains the result of the downstream allocation query. Subclasses should chain up to the parent implementation to invoke the default handler.
Parameters:
do_decide_allocation
def do_decide_allocation (dec, query):
#python implementation of the 'decide_allocation' virtual method
Optional. Setup the allocation parameters for allocating output buffers. The passed in query contains the result of the downstream allocation query. Subclasses should chain up to the parent implementation to invoke the default handler.
Parameters:
flush
flush (GstAudioDecoder * dec, gboolean hard)
Optional. Instructs subclass to clear any codec caches and discard any pending samples and not yet returned decoded data. hard indicates whether a FLUSH is being processed, or otherwise a DISCONT (or conceptually similar).
Parameters:
dec
–
hard
–
vfunc_flush
function vfunc_flush(dec: GstAudio.AudioDecoder, hard: Number): {
// javascript implementation of the 'flush' virtual method
}
Optional. Instructs subclass to clear any codec caches and discard any pending samples and not yet returned decoded data. hard indicates whether a FLUSH is being processed, or otherwise a DISCONT (or conceptually similar).
Parameters:
do_flush
def do_flush (dec, hard):
#python implementation of the 'flush' virtual method
Optional. Instructs subclass to clear any codec caches and discard any pending samples and not yet returned decoded data. hard indicates whether a FLUSH is being processed, or otherwise a DISCONT (or conceptually similar).
Parameters:
getcaps
GstCaps * getcaps (GstAudioDecoder * dec, GstCaps * filter)
Optional. Allows for a custom sink getcaps implementation. If not implemented, default returns gst_audio_decoder_proxy_getcaps applied to sink template caps.
Parameters:
dec
–
filter
–
vfunc_getcaps
function vfunc_getcaps(dec: GstAudio.AudioDecoder, filter: Gst.Caps): {
// javascript implementation of the 'getcaps' virtual method
}
Optional. Allows for a custom sink getcaps implementation. If not implemented, default returns gst_audio_decoder_proxy_getcaps applied to sink template caps.
Parameters:
do_getcaps
def do_getcaps (dec, filter):
#python implementation of the 'getcaps' virtual method
Optional. Allows for a custom sink getcaps implementation. If not implemented, default returns gst_audio_decoder_proxy_getcaps applied to sink template caps.
Parameters:
handle_frame
GstFlowReturn handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
Provides input data (or NULL to clear any remaining data) to subclass. Input data ref management is performed by base class, subclass should not care or intervene, and input data is only valid until next call to base class, most notably a call to gst_audio_decoder_finish_frame.
Parameters:
dec
–
buffer
–
vfunc_handle_frame
function vfunc_handle_frame(dec: GstAudio.AudioDecoder, buffer: Gst.Buffer): {
// javascript implementation of the 'handle_frame' virtual method
}
Provides input data (or NULL to clear any remaining data) to subclass. Input data ref management is performed by base class, subclass should not care or intervene, and input data is only valid until next call to base class, most notably a call to GstAudio.AudioDecoder.prototype.finish_frame.
Parameters:
do_handle_frame
def do_handle_frame (dec, buffer):
#python implementation of the 'handle_frame' virtual method
Provides input data (or NULL to clear any remaining data) to subclass. Input data ref management is performed by base class, subclass should not care or intervene, and input data is only valid until next call to base class, most notably a call to GstAudio.AudioDecoder.finish_frame.
Parameters:
negotiate
gboolean negotiate (GstAudioDecoder * dec)
Optional. Negotiate with downstream and configure buffer pools, etc. Subclasses should chain up to the parent implementation to invoke the default handler.
Parameters:
dec
–
vfunc_negotiate
function vfunc_negotiate(dec: GstAudio.AudioDecoder): {
// javascript implementation of the 'negotiate' virtual method
}
Optional. Negotiate with downstream and configure buffer pools, etc. Subclasses should chain up to the parent implementation to invoke the default handler.
Parameters:
do_negotiate
def do_negotiate (dec):
#python implementation of the 'negotiate' virtual method
Optional. Negotiate with downstream and configure buffer pools, etc. Subclasses should chain up to the parent implementation to invoke the default handler.
Parameters:
open
gboolean open (GstAudioDecoder * dec)
Optional. Called when the element changes to GST_STATE_READY. Allows opening external resources.
Parameters:
dec
–
vfunc_open
function vfunc_open(dec: GstAudio.AudioDecoder): {
// javascript implementation of the 'open' virtual method
}
Optional. Called when the element changes to GST_STATE_READY. Allows opening external resources.
Parameters:
do_open
def do_open (dec):
#python implementation of the 'open' virtual method
Optional. Called when the element changes to GST_STATE_READY. Allows opening external resources.
Parameters:
parse
GstFlowReturn parse (GstAudioDecoder * dec, GstAdapter * adapter, gint * offset, gint * length)
Optional. Allows chopping incoming data into manageable units (frames) for subsequent decoding. This division is at subclass discretion and may or may not correspond to 1 (or more) frames as defined by audio format.
Parameters:
dec
–
adapter
–
offset
–
length
–
vfunc_parse
function vfunc_parse(dec: GstAudio.AudioDecoder, adapter: GstBase.Adapter): {
// javascript implementation of the 'parse' virtual method
}
Optional. Allows chopping incoming data into manageable units (frames) for subsequent decoding. This division is at subclass discretion and may or may not correspond to 1 (or more) frames as defined by audio format.
Parameters:
Returns a tuple made of:
do_parse
def do_parse (dec, adapter):
#python implementation of the 'parse' virtual method
Optional. Allows chopping incoming data into manageable units (frames) for subsequent decoding. This division is at subclass discretion and may or may not correspond to 1 (or more) frames as defined by audio format.
Parameters:
Returns a tuple made of:
pre_push
GstFlowReturn pre_push (GstAudioDecoder * dec, GstBuffer ** buffer)
Optional. Called just prior to pushing (encoded data) buffer downstream. Subclass has full discretionary access to buffer, and a not OK flow return will abort downstream pushing.
Parameters:
dec
–
buffer
–
vfunc_pre_push
function vfunc_pre_push(dec: GstAudio.AudioDecoder, buffer: Gst.Buffer): {
// javascript implementation of the 'pre_push' virtual method
}
Optional. Called just prior to pushing (encoded data) buffer downstream. Subclass has full discretionary access to buffer, and a not OK flow return will abort downstream pushing.
Parameters:
do_pre_push
def do_pre_push (dec, buffer):
#python implementation of the 'pre_push' virtual method
Optional. Called just prior to pushing (encoded data) buffer downstream. Subclass has full discretionary access to buffer, and a not OK flow return will abort downstream pushing.
Parameters:
propose_allocation
gboolean propose_allocation (GstAudioDecoder * dec, GstQuery * query)
Optional. Propose buffer allocation parameters for upstream elements. Subclasses should chain up to the parent implementation to invoke the default handler.
Parameters:
dec
–
query
–
vfunc_propose_allocation
function vfunc_propose_allocation(dec: GstAudio.AudioDecoder, query: Gst.Query): {
// javascript implementation of the 'propose_allocation' virtual method
}
Optional. Propose buffer allocation parameters for upstream elements. Subclasses should chain up to the parent implementation to invoke the default handler.
Parameters:
do_propose_allocation
def do_propose_allocation (dec, query):
#python implementation of the 'propose_allocation' virtual method
Optional. Propose buffer allocation parameters for upstream elements. Subclasses should chain up to the parent implementation to invoke the default handler.
Parameters:
set_format
gboolean set_format (GstAudioDecoder * dec, GstCaps * caps)
Notifies subclass of incoming data format (caps).
Parameters:
dec
–
caps
–
vfunc_set_format
function vfunc_set_format(dec: GstAudio.AudioDecoder, caps: Gst.Caps): {
// javascript implementation of the 'set_format' virtual method
}
Notifies subclass of incoming data format (caps).
Parameters:
do_set_format
def do_set_format (dec, caps):
#python implementation of the 'set_format' virtual method
Notifies subclass of incoming data format (caps).
Parameters:
sink_event
gboolean sink_event (GstAudioDecoder * dec, GstEvent * event)
Optional. Event handler on the sink pad. Subclasses should chain up to the parent implementation to invoke the default handler.
Parameters:
dec
–
event
–
vfunc_sink_event
function vfunc_sink_event(dec: GstAudio.AudioDecoder, event: Gst.Event): {
// javascript implementation of the 'sink_event' virtual method
}
Optional. Event handler on the sink pad. Subclasses should chain up to the parent implementation to invoke the default handler.
Parameters:
do_sink_event
def do_sink_event (dec, event):
#python implementation of the 'sink_event' virtual method
Optional. Event handler on the sink pad. Subclasses should chain up to the parent implementation to invoke the default handler.
Parameters:
sink_query
gboolean sink_query (GstAudioDecoder * dec, GstQuery * query)
Optional. Query handler on the sink pad. This function should return TRUE if the query could be performed. Subclasses should chain up to the parent implementation to invoke the default handler. Since: 1.6
Parameters:
dec
–
query
–
vfunc_sink_query
function vfunc_sink_query(dec: GstAudio.AudioDecoder, query: Gst.Query): {
// javascript implementation of the 'sink_query' virtual method
}
Optional. Query handler on the sink pad. This function should return TRUE if the query could be performed. Subclasses should chain up to the parent implementation to invoke the default handler. Since: 1.6
Parameters:
do_sink_query
def do_sink_query (dec, query):
#python implementation of the 'sink_query' virtual method
Optional. Query handler on the sink pad. This function should return TRUE if the query could be performed. Subclasses should chain up to the parent implementation to invoke the default handler. Since: 1.6
Parameters:
src_event
gboolean src_event (GstAudioDecoder * dec, GstEvent * event)
Optional. Event handler on the src pad. Subclasses should chain up to the parent implementation to invoke the default handler.
Parameters:
dec
–
event
–
vfunc_src_event
function vfunc_src_event(dec: GstAudio.AudioDecoder, event: Gst.Event): {
// javascript implementation of the 'src_event' virtual method
}
Optional. Event handler on the src pad. Subclasses should chain up to the parent implementation to invoke the default handler.
Parameters:
do_src_event
def do_src_event (dec, event):
#python implementation of the 'src_event' virtual method
Optional. Event handler on the src pad. Subclasses should chain up to the parent implementation to invoke the default handler.
Parameters:
src_query
gboolean src_query (GstAudioDecoder * dec, GstQuery * query)
Optional. Query handler on the source pad. This function should return TRUE if the query could be performed. Subclasses should chain up to the parent implementation to invoke the default handler. Since: 1.6
Parameters:
dec
–
query
–
vfunc_src_query
function vfunc_src_query(dec: GstAudio.AudioDecoder, query: Gst.Query): {
// javascript implementation of the 'src_query' virtual method
}
Optional. Query handler on the source pad. This function should return TRUE if the query could be performed. Subclasses should chain up to the parent implementation to invoke the default handler. Since: 1.6
Parameters:
do_src_query
def do_src_query (dec, query):
#python implementation of the 'src_query' virtual method
Optional. Query handler on the source pad. This function should return TRUE if the query could be performed. Subclasses should chain up to the parent implementation to invoke the default handler. Since: 1.6
Parameters:
start
gboolean start (GstAudioDecoder * dec)
Optional. Called when the element starts processing. Allows opening external resources.
Parameters:
dec
–
vfunc_start
function vfunc_start(dec: GstAudio.AudioDecoder): {
// javascript implementation of the 'start' virtual method
}
Optional. Called when the element starts processing. Allows opening external resources.
Parameters:
do_start
def do_start (dec):
#python implementation of the 'start' virtual method
Optional. Called when the element starts processing. Allows opening external resources.
Parameters:
stop
gboolean stop (GstAudioDecoder * dec)
Optional. Called when the element stops processing. Allows closing external resources.
Parameters:
dec
–
vfunc_stop
function vfunc_stop(dec: GstAudio.AudioDecoder): {
// javascript implementation of the 'stop' virtual method
}
Optional. Called when the element stops processing. Allows closing external resources.
Parameters:
do_stop
def do_stop (dec):
#python implementation of the 'stop' virtual method
Optional. Called when the element stops processing. Allows closing external resources.
Parameters:
transform_meta
gboolean transform_meta (GstAudioDecoder * enc, GstBuffer * outbuf, GstMeta * meta, GstBuffer * inbuf)
Optional. Transform the metadata on the input buffer to the output buffer. By default this method copies all meta without tags and meta with only the "audio" tag. subclasses can implement this method and return TRUE if the metadata is to be copied. Since: 1.6
Parameters:
enc
–
outbuf
–
meta
–
inbuf
–
vfunc_transform_meta
function vfunc_transform_meta(enc: GstAudio.AudioDecoder, outbuf: Gst.Buffer, meta: Gst.Meta, inbuf: Gst.Buffer): {
// javascript implementation of the 'transform_meta' virtual method
}
Optional. Transform the metadata on the input buffer to the output buffer. By default this method copies all meta without tags and meta with only the "audio" tag. subclasses can implement this method and return true if the metadata is to be copied. Since: 1.6
Parameters:
do_transform_meta
def do_transform_meta (enc, outbuf, meta, inbuf):
#python implementation of the 'transform_meta' virtual method
Optional. Transform the metadata on the input buffer to the output buffer. By default this method copies all meta without tags and meta with only the "audio" tag. subclasses can implement this method and return True if the metadata is to be copied. Since: 1.6
Parameters:
Function Macros
GST_AUDIO_DECODER_CAST
#define GST_AUDIO_DECODER_CAST(obj) \ ((GstAudioDecoder *)(obj))
GST_AUDIO_DECODER_ERROR
#define GST_AUDIO_DECODER_ERROR(el, weight, domain, code, text, debug, ret) \ G_STMT_START { \ gchar *__txt = _gst_element_error_printf text; \ gchar *__dbg = _gst_element_error_printf debug; \ GstAudioDecoder *__dec = GST_AUDIO_DECODER (el); \ ret = _gst_audio_decoder_error (__dec, weight, GST_ ## domain ## _ERROR, \ GST_ ## domain ## _ERROR_ ## code, __txt, __dbg, __FILE__, \ GST_FUNCTION, __LINE__); \ } G_STMT_END
Utility function that audio decoder elements can use in case they encountered a data processing error that may be fatal for the current "data unit" but need not prevent subsequent decoding. Such errors are counted and if there are too many, as configured in the context's max_errors, the pipeline will post an error message and the application will be requested to stop further media processing. Otherwise, it is considered a "glitch" and only a warning is logged. In either case, ret is set to the proper value to return to upstream/caller (indicating either GST_FLOW_ERROR or GST_FLOW_OK).
Parameters:
el
–
the base audio decoder element that generates the error
weight
–
element defined weight of the error, added to error count
domain
–
like CORE, LIBRARY, RESOURCE or STREAM (see Core Library-GstGError)
code
–
error code defined for that domain (see Core Library-GstGError)
text
–
the message to display (format string and args enclosed in parentheses)
debug
–
debugging information for the message (format string and args enclosed in parentheses)
ret
–
variable to receive return value
GST_AUDIO_DECODER_INPUT_SEGMENT
#define GST_AUDIO_DECODER_INPUT_SEGMENT(obj) (GST_AUDIO_DECODER_CAST (obj)->input_segment)
Gives the input segment of the element.
Parameters:
obj
–
audio decoder instance
GST_AUDIO_DECODER_OUTPUT_SEGMENT
#define GST_AUDIO_DECODER_OUTPUT_SEGMENT(obj) (GST_AUDIO_DECODER_CAST (obj)->output_segment)
Gives the output segment of the element.
Parameters:
obj
–
audio decoder instance
GST_AUDIO_DECODER_SINK_PAD
#define GST_AUDIO_DECODER_SINK_PAD(obj) (((GstAudioDecoder *) (obj))->sinkpad)
Gives the pointer to the sink GstPad object of the element.
Parameters:
obj
–
base audio codec instance
GST_AUDIO_DECODER_SRC_PAD
#define GST_AUDIO_DECODER_SRC_PAD(obj) (((GstAudioDecoder *) (obj))->srcpad)
Gives the pointer to the source GstPad object of the element.
Parameters:
obj
–
base audio codec instance
GST_AUDIO_DECODER_STREAM_LOCK
#define GST_AUDIO_DECODER_STREAM_LOCK(dec) g_rec_mutex_lock (&GST_AUDIO_DECODER (dec)->stream_lock)
GST_AUDIO_DECODER_STREAM_UNLOCK
#define GST_AUDIO_DECODER_STREAM_UNLOCK(dec) g_rec_mutex_unlock (&GST_AUDIO_DECODER (dec)->stream_lock)
Constants
GST_AUDIO_DECODER_MAX_ERRORS
#define GST_AUDIO_DECODER_MAX_ERRORS -1
Default maximum number of errors tolerated before signaling error.
GstAudio.AUDIO_DECODER_MAX_ERRORS
Default maximum number of errors tolerated before signaling error.
GstAudio.AUDIO_DECODER_MAX_ERRORS
Default maximum number of errors tolerated before signaling error.
GST_AUDIO_DECODER_SINK_NAME
#define GST_AUDIO_DECODER_SINK_NAME "sink"
The name of the templates for the sink pad.
GstAudio.AUDIO_DECODER_SINK_NAME
The name of the templates for the sink pad.
GstAudio.AUDIO_DECODER_SINK_NAME
The name of the templates for the sink pad.
GST_AUDIO_DECODER_SRC_NAME
#define GST_AUDIO_DECODER_SRC_NAME "src"
The name of the templates for the source pad.
GstAudio.AUDIO_DECODER_SRC_NAME
The name of the templates for the source pad.
GstAudio.AUDIO_DECODER_SRC_NAME
The name of the templates for the source pad.
GST_TYPE_AUDIO_DECODER
#define GST_TYPE_AUDIO_DECODER \ (gst_audio_decoder_get_type())
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