GstAudio.AudioDecoder¶
- Subclasses:
None
Methods¶
- Inherited:
- Structs:
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Virtual Methods¶
- Inherited:
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Properties¶
- Inherited:
Name |
Type |
Flags |
Short Description |
|---|---|---|---|
r/w |
Max consecutive decoder errors before returning flow error |
||
r/w |
Aggregate output data to a minimum of latency time (ns) |
||
r/w |
Perform packet loss concealment (if supported) |
||
r/w |
Perfect ts while timestamp jitter/imperfection within tolerance (ns) |
Signals¶
- Inherited:
Fields¶
- Inherited:
Name |
Type |
Access |
Description |
|---|---|---|---|
element |
r |
||
input_segment |
r |
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output_segment |
r |
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sinkpad |
r |
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srcpad |
r |
||
stream_lock |
r |
Class Details¶
- class GstAudio.AudioDecoder(**kwargs)¶
- Bases:
- Abstract:
Yes
- Structure:
This base class is for audio decoders turning encoded data into raw audio samples.
GstAudio.AudioDecoderand subclass should cooperate as follows.- Configuration
Initially,
GstAudio.AudioDecodercalls 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).GstAudio.AudioDecodercalls 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.GstAudio.AudioDecodercalls stop at end of all processing.
As of configuration stage, and throughout processing,
GstAudio.AudioDecoderprovides 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
GstAudio.AudioDecoderClasswill handle both srcpad and sinkpad events. Sink events will be passed to subclass if event callback has been provided.
- Shutdown phase
GstAudio.AudioDecoderclass 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
GstAudio.AudioDecoder: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
Nonedata in handle_frame and provide for data for indicated duration.
- allocate_output_buffer(size)¶
- Parameters:
size (
int) – size of the buffer- Returns:
allocated buffer
- Return type:
Helper function that allocates a buffer to hold an audio frame for self's current output format.
- finish_frame(buf, frames)¶
- Parameters:
buf (
Gst.BufferorNone) – decoded dataframes (
int) – number of decoded frames represented by decoded data
- Returns:
a
Gst.FlowReturnthat should be escalated to caller (of caller)- Return type:
Collects decoded data and pushes it downstream.
buf may be
Nonein 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
GstAudio.AudioDecoder.do_handle_frame() may be invalidated by a call to this function.
- finish_subframe(buf)¶
- Parameters:
buf (
Gst.BufferorNone) – decoded data- Returns:
a
Gst.FlowReturnthat should be escalated to caller (of caller)- Return type:
Collects decoded data and pushes it downstream. This function may be called multiple times for a given input frame.
buf may be
Nonein which case it is assumed that the current input frame is finished. This is equivalent to callingGstAudio.AudioDecoder.finish_subframe() with aNonebuffer 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
GstAudio.AudioDecoder.do_handle_frame() may be invalidated by a call to this function.New in version 1.16.
- get_allocator()¶
- Returns:
- allocator:
the
Gst.Allocatorused- params:
the
Gst.AllocationParamsof allocator
- Return type:
(allocator:
Gst.AllocatororNone, params:Gst.AllocationParams)
Lets
GstAudio.AudioDecodersub-classes to know the memory allocator used by the base class and its params.Unref the allocator after use it.
- get_audio_info()¶
- Returns:
a
GstAudio.AudioInfodescribing the input audio format- Return type:
- get_drainable()¶
-
Queries decoder drain handling.
- get_latency()¶
- Returns:
- min:
a pointer to storage to hold minimum latency
- max:
a pointer to storage to hold maximum latency
- Return type:
Sets the variables pointed to by min and max to the currently configured latency.
- get_min_latency()¶
- Returns:
aggregation latency.
MT safe.
- Return type:
Queries decoder’s latency aggregation.
- get_needs_format()¶
-
Queries decoder required format handling.
- get_parse_state()¶
- Returns:
- sync:
a pointer to a variable to hold the current sync state
- eos:
a pointer to a variable to hold the current eos state
- Return type:
Return current parsing (sync and eos) state.
- get_plc()¶
-
Queries decoder packet loss concealment handling.
- get_tolerance()¶
- Returns:
decoder audio jitter tolerance threshold.
MT safe.
- Return type:
Queries current audio jitter tolerance threshold.
- merge_tags(tags, mode)¶
- Parameters:
tags (
Gst.TagListorNone) – aGst.TagListto merge, orNonemode (
Gst.TagMergeMode) – theGst.TagMergeModeto use, usuallyGst.TagMergeMode.REPLACE
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.
- negotiate()¶
-
Negotiate with downstream elements to currently configured
GstAudio.AudioInfo. UnmarkGst.PadFlags.NEED_RECONFIGUREin any case. But mark it again if negotiate fails.
- proxy_getcaps(caps, filter)¶
- Parameters:
- Returns:
a
Gst.Capsowned by caller- Return type:
Returns caps that express caps (or sink template caps if caps ==
None) restricted to rate/channels/… combinations supported by downstream elements.New in version 1.6.
- set_allocation_caps(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 toNonethe allocation query will use the caps from the pad.New in version 1.10.
- set_drainable(enabled)¶
- Parameters:
enabled (
bool) – new state
Configures decoder drain handling. If drainable, subclass might be handed a
Nonebuffer 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.
- set_estimate_rate(enabled)¶
- Parameters:
enabled (
bool) – whether to enable byte to time conversion
Allows baseclass to perform byte to time estimated conversion.
- set_latency(min, 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.
- set_max_errors(num)¶
- Parameters:
num (
int) – max tolerated 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
GstAudio.AUDIO_DECODER_MAX_ERRORS.
- set_min_latency(num)¶
- Parameters:
num (
int) – new minimum latency
Sets decoder minimum aggregation latency.
MT safe.
- set_needs_format(enabled)¶
- Parameters:
enabled (
bool) – new state
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.
- set_output_caps(caps)¶
-
Configure output caps on the srcpad of self. Similar to
GstAudio.AudioDecoder.set_output_format(), but allows subclasses to specify output caps that can’t be expressed viaGstAudio.AudioInfoe.g. caps that have caps features.New in version 1.16.
- set_output_format(info)¶
- Parameters:
info (
GstAudio.AudioInfo) –GstAudio.AudioInfo- Returns:
Trueon success.- Return type:
Configure output info on the srcpad of self.
- set_plc(enabled)¶
- Parameters:
enabled (
bool) – new state
Enable or disable decoder packet loss concealment, provided subclass and codec are capable and allow handling plc.
MT safe.
- set_plc_aware(plc)¶
- Parameters:
plc (
bool) – new plc state
Indicates whether or not subclass handles packet loss concealment (plc).
- set_tolerance(tolerance)¶
- Parameters:
tolerance (
int) – new tolerance
Configures decoder audio jitter tolerance threshold.
MT safe.
- set_use_default_pad_acceptcaps(use)¶
- Parameters:
use (
bool) – if the default pad accept-caps query handling should be used
Lets
GstAudio.AudioDecodersub-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
New in version 1.6.
- do_handle_frame(buffer) virtual¶
- Parameters:
buffer (
Gst.Buffer) –- Return type:
- do_negotiate() virtual¶
-
Negotiate with downstream elements to currently configured
GstAudio.AudioInfo. UnmarkGst.PadFlags.NEED_RECONFIGUREin any case. But mark it again if negotiate fails.
- do_parse(adapter) virtual¶
- Parameters:
adapter (
GstBase.Adapter) –- Return type:
(
Gst.FlowReturn, offset:int, length:int)
- do_pre_push(buffer) virtual¶
- Parameters:
buffer (
Gst.Buffer) –- Return type:
- do_transform_meta(outbuf, meta, inbuf) virtual¶
- Parameters:
outbuf (
Gst.Buffer) –meta (
Gst.Meta) –inbuf (
Gst.Buffer) –
- Return type:
Property Details¶
- GstAudio.AudioDecoder.props.max_errors¶
-
Maximum number of tolerated consecutive decode errors. See
GstAudio.AudioDecoder.set_max_errors() for more details.New in version 1.18.
- GstAudio.AudioDecoder.props.min_latency¶
-
Aggregate output data to a minimum of latency time (ns)
- GstAudio.AudioDecoder.props.plc¶
-
Perform packet loss concealment (if supported)