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| struct | GstAudioDecoder |
| struct | GstAudioDecoderClass |
| #define | GST_AUDIO_DECODER_SINK_NAME |
| #define | GST_AUDIO_DECODER_SRC_NAME |
| #define | GST_AUDIO_DECODER_MAX_ERRORS |
This base class is for audio decoders turning encoded data into raw audio samples.
GstAudioDecoder and subclass should cooperate as follows.
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.
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.
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.
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.
#define GST_AUDIO_DECODER_ERROR(el, weight, domain, code, text, debug, ret)
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).
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 gstreamer-GstGError) |
|
code |
error code defined for that domain (see gstreamer-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 |
#define GST_AUDIO_DECODER_SINK_PAD(obj) (((GstAudioDecoder *) (obj))->sinkpad)
Gives the pointer to the sink GstPad object of the element.
#define GST_AUDIO_DECODER_SRC_PAD(obj) (((GstAudioDecoder *) (obj))->srcpad)
Gives the pointer to the source GstPad object of the element.
#define GST_AUDIO_DECODER_INPUT_SEGMENT(obj) (GST_AUDIO_DECODER_CAST (obj)->input_segment)
Gives the input segment of the element.
#define GST_AUDIO_DECODER_OUTPUT_SEGMENT(obj) (GST_AUDIO_DECODER_CAST (obj)->output_segment)
Gives the output segment of the element.
#define GST_AUDIO_DECODER_STREAM_LOCK(dec) g_rec_mutex_lock (&GST_AUDIO_DECODER (dec)->stream_lock)
#define GST_AUDIO_DECODER_STREAM_UNLOCK(dec) g_rec_mutex_unlock (&GST_AUDIO_DECODER (dec)->stream_lock)
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 gst_audio_decoder_handle_frame() may be
invalidated by a call to this function.
gboolean gst_audio_decoder_set_output_format (GstAudioDecoder *dec,const GstAudioInfo *info);
Configure output info on the srcpad of dec
.
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.
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.
void 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.
dec |
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allocator |
the GstAllocator used. |
[out][allow-none][transfer full] |
params |
the
GstAllocatorParams of |
[out][allow-none][transfer full] |
GstAudioInfo *
gst_audio_decoder_get_audio_info (GstAudioDecoder *dec);
gint
gst_audio_decoder_get_estimate_rate (GstAudioDecoder *dec);
gboolean
gst_audio_decoder_get_drainable (GstAudioDecoder *dec);
Queries decoder drain handling.
void 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.
GstClockTime
gst_audio_decoder_get_min_latency (GstAudioDecoder *dec);
Queries decoder's latency aggregation.
gboolean
gst_audio_decoder_get_needs_format (GstAudioDecoder *dec);
Queries decoder required format handling.
void gst_audio_decoder_get_parse_state (GstAudioDecoder *dec,gboolean *sync,gboolean *eos);
Return current parsing (sync and eos) state.
gboolean
gst_audio_decoder_get_plc (GstAudioDecoder *dec);
Queries decoder packet loss concealment handling.
GstClockTime
gst_audio_decoder_get_tolerance (GstAudioDecoder *dec);
Queries current audio jitter tolerance threshold.
void gst_audio_decoder_set_estimate_rate (GstAudioDecoder *dec,gboolean enabled);
Allows baseclass to perform byte to time estimated conversion.
void 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.
void gst_audio_decoder_set_latency (GstAudioDecoder *dec,GstClockTime min,GstClockTime max);
Sets decoder latency.
void 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.
void gst_audio_decoder_set_min_latency (GstAudioDecoder *dec,GstClockTime num);
Sets decoder minimum aggregation latency.
MT safe.
void 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.
void 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.
void gst_audio_decoder_set_plc_aware (GstAudioDecoder *dec,gboolean plc);
Indicates whether or not subclass handles packet loss concealment (plc).
void gst_audio_decoder_set_tolerance (GstAudioDecoder *dec,GstClockTime tolerance);
Configures decoder audio jitter tolerance threshold.
MT safe.
void 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
void 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
Since: 1.6
void 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.
dec |
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tags |
a GstTagList to merge, or NULL. |
[allow-none] |
mode |
the GstTagMergeMode to use, usually GST_TAG_MERGE_REPLACE |
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.
Since: 1.6
struct GstAudioDecoderClass {
GstElementClass element_class;
/* virtual methods for subclasses */
gboolean (*start) (GstAudioDecoder *dec);
gboolean (*stop) (GstAudioDecoder *dec);
gboolean (*set_format) (GstAudioDecoder *dec,
GstCaps *caps);
GstFlowReturn (*parse) (GstAudioDecoder *dec,
GstAdapter *adapter,
gint *offset, gint *length);
GstFlowReturn (*handle_frame) (GstAudioDecoder *dec,
GstBuffer *buffer);
void (*flush) (GstAudioDecoder *dec, gboolean hard);
GstFlowReturn (*pre_push) (GstAudioDecoder *dec,
GstBuffer **buffer);
gboolean (*sink_event) (GstAudioDecoder *dec,
GstEvent *event);
gboolean (*src_event) (GstAudioDecoder *dec,
GstEvent *event);
gboolean (*open) (GstAudioDecoder *dec);
gboolean (*close) (GstAudioDecoder *dec);
gboolean (*negotiate) (GstAudioDecoder *dec);
gboolean (*decide_allocation) (GstAudioDecoder *dec, GstQuery *query);
gboolean (*propose_allocation) (GstAudioDecoder *dec,
GstQuery * query);
gboolean (*sink_query) (GstAudioDecoder *dec, GstQuery *query);
gboolean (*src_query) (GstAudioDecoder *dec, GstQuery *query);
GstCaps * (*getcaps) (GstAudioDecoder * dec,
GstCaps * filter);
gboolean (*transform_meta) (GstAudioDecoder *enc, GstBuffer *outbuf,
GstMeta *meta, GstBuffer *inbuf);
};
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.
GstElementClass |
The parent class structure |
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Optional. Called when the element starts processing. Allows opening external resources. |
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Optional. Called when the element stops processing. Allows closing external resources. |
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Notifies subclass of incoming data format (caps). |
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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. |
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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 |
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Optional.
Instructs subclass to clear any codec caches and discard
any pending samples and not yet returned decoded data.
|
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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. |
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Optional. Event handler on the sink pad. Subclasses should chain up to the parent implementation to invoke the default handler. |
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Optional. Event handler on the src pad. Subclasses should chain up to the parent implementation to invoke the default handler. |
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Optional. Called when the element changes to GST_STATE_READY. Allows opening external resources. |
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Optional. Called when the element changes to GST_STATE_NULL. Allows closing external resources. |
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Optional. Negotiate with downstream and configure buffer pools, etc. Subclasses should chain up to the parent implementation to invoke the default handler. |
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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. |
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Optional. Propose buffer allocation parameters for upstream elements. Subclasses should chain up to the parent implementation to invoke the default handler. |
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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 |
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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 |
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Optional. Allows for a custom sink getcaps implementation. If not implemented, default returns gst_audio_decoder_proxy_getcaps applied to sink template caps. |
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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 |
#define GST_AUDIO_DECODER_SINK_NAME "sink"
The name of the templates for the sink pad.
#define GST_AUDIO_DECODER_SRC_NAME "src"
The name of the templates for the source pad.
“min-latency” property“min-latency” gint64
Aggregate output data to a minimum of latency time (ns).
Flags: Read / Write
Allowed values: >= 0
Default value: 0
“plc” property“plc” gboolean
Perform packet loss concealment (if supported).
Flags: Read / Write
Default value: FALSE
“tolerance” property“tolerance” gint64
Perfect ts while timestamp jitter/imperfection within tolerance (ns).
Flags: Read / Write
Allowed values: >= 0
Default value: 0