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| gboolean | hard-resync | Read / Write |
| gboolean | mark-granule | Read |
| gboolean | perfect-timestamp | Read / Write |
| gint64 | tolerance | Read / Write |
| struct | GstAudioEncoder |
| struct | GstAudioEncoderClass |
| #define | GST_AUDIO_ENCODER_SINK_NAME |
| #define | GST_AUDIO_ENCODER_SRC_NAME |
This base class is for audio encoders turning raw audio samples into encoded audio data.
GstAudioEncoder and subclass should cooperate as follows.
Configuration
Initially, GstAudioEncoder calls start when the encoder element
is activated, which allows subclass to perform any global setup.
GstAudioEncoder calls set_format to inform subclass of the format
of input audio data that it is about to receive. Subclass should
setup for encoding and configure various base class parameters
appropriately, notably those directing desired input data handling.
While unlikely, it might be called more than once, if changing input
parameters require reconfiguration.
GstAudioEncoder calls stop at end of all processing.
Data processing
Base class gathers input sample data (as directed by the context's
frame_samples and frame_max) and provides this to subclass' handle_frame.
If codec processing results in encoded data, subclass should call
gst_audio_encoder_finish_frame() to have encoded data pushed
downstream. Alternatively, it might also call
gst_audio_encoder_finish_frame() (with a NULL buffer and some number of
dropped samples) to indicate dropped (non-encoded) samples.
Just prior to actually pushing a buffer downstream,
it is passed to pre_push.
During the parsing process GstAudioEncoderClass will handle both
srcpad and sinkpad events. Sink events will be passed to subclass
if event callback has been provided.
Shutdown phase
GstAudioEncoder 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_encoder_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 either favor tracking upstream timestamps (at the possible expense of jitter) or aim to arrange for a perfect stream of output timestamps, depending on “perfect-timestamp”. However, in the latter case, the input may not be so perfect or ideal, which is handled as follows. An input timestamp is compared with the expected timestamp as dictated by input sample stream and if the deviation is less than “tolerance”, the deviation is discarded. Otherwise, it is considered a discontuinity and subsequent output timestamp is resynced to the new position after performing configured discontinuity processing. In the non-perfect-timestamp case, an upstream variation exceeding tolerance only leads to marking DISCONT on subsequent outgoing (while timestamps are adjusted to upstream regardless of variation). While DISCONT is also marked in the perfect-timestamp case, this one optionally (see “hard-resync”) performs some additional steps, such as clipping of (early) input samples or draining all currently remaining input data, depending on the direction of the discontuinity.
If perfect timestamps are arranged, it is also possible to request baseclass (usually set by subclass) to provide additional buffer metadata (in OFFSET and OFFSET_END) fields according to granule defined semantics currently needed by oggmux. Specifically, OFFSET is set to granulepos (= sample count including buffer) and OFFSET_END to corresponding timestamp (as determined by same sample count and sample rate).
Things that subclass need to take care of:
Provide pad templates
Set source pad caps when appropriate
Inform base class of buffer processing needs using context's frame_samples and frame_bytes.
Set user-configurable properties to sane defaults for format and implementing codec at hand, e.g. those controlling timestamp behaviour and discontinuity processing.
Accept data in handle_frame and provide encoded results to
gst_audio_encoder_finish_frame().
#define GST_AUDIO_ENCODER_SINK_PAD(obj) (GST_AUDIO_ENCODER_CAST (obj)->sinkpad)
Gives the pointer to the sink GstPad object of the element.
#define GST_AUDIO_ENCODER_SRC_PAD(obj) (GST_AUDIO_ENCODER_CAST (obj)->srcpad)
Gives the pointer to the source GstPad object of the element.
GstFlowReturn gst_audio_encoder_finish_frame (GstAudioEncoder *enc,GstBuffer *buffer,gint samples);
Collects encoded data and pushes encoded data downstream. Source pad caps must be set when this is called.
If samples
< 0, then best estimate is all samples provided to encoder
(subclass) so far. buf
may be NULL, in which case next number of samples
are considered discarded, e.g. as a result of discontinuous transmission,
and a discontinuity is marked.
Note that samples received in gst_audio_encoder_handle_frame()
may be invalidated by a call to this function.
GstBuffer * gst_audio_encoder_allocate_output_buffer (GstAudioEncoder *enc,gsize size);
Helper function that allocates a buffer to hold an encoded audio frame
for enc
's current output format.
void gst_audio_encoder_get_allocator (GstAudioEncoder *enc,GstAllocator **allocator,GstAllocationParams *params);
Lets GstAudioEncoder sub-classes to know the memory allocator
used by the base class and its params
.
Unref the allocator
after use it.
enc |
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allocator |
the GstAllocator used. |
[out][allow-none][transfer full] |
params |
the
GstAllocatorParams of |
[out][allow-none][transfer full] |
gboolean gst_audio_encoder_set_output_format (GstAudioEncoder *enc,GstCaps *caps);
Configure output caps on the srcpad of enc
.
gboolean
gst_audio_encoder_negotiate (GstAudioEncoder *enc);
Negotiate with downstream elements to currently configured GstCaps. Unmark GST_PAD_FLAG_NEED_RECONFIGURE in any case. But mark it again if negotiate fails.
GstAudioInfo *
gst_audio_encoder_get_audio_info (GstAudioEncoder *enc);
gboolean
gst_audio_encoder_get_drainable (GstAudioEncoder *enc);
Queries encoder drain handling.
gint
gst_audio_encoder_get_frame_samples_min
(GstAudioEncoder *enc);
gint
gst_audio_encoder_get_frame_samples_max
(GstAudioEncoder *enc);
gboolean
gst_audio_encoder_get_hard_min (GstAudioEncoder *enc);
Queries encoder hard minimum handling.
gboolean
gst_audio_encoder_get_hard_resync (GstAudioEncoder *enc);
void gst_audio_encoder_get_latency (GstAudioEncoder *enc,GstClockTime *min,GstClockTime *max);
Sets the variables pointed to by min
and max
to the currently configured
latency.
gboolean
gst_audio_encoder_get_mark_granule (GstAudioEncoder *enc);
Queries if the encoder will handle granule marking.
gboolean
gst_audio_encoder_get_perfect_timestamp
(GstAudioEncoder *enc);
Queries encoder perfect timestamp behaviour.
GstClockTime
gst_audio_encoder_get_tolerance (GstAudioEncoder *enc);
Queries current audio jitter tolerance threshold.
GstCaps * gst_audio_encoder_proxy_getcaps (GstAudioEncoder *enc,GstCaps *caps,GstCaps *filter);
Returns caps that express caps
(or sink template caps if caps
== NULL)
restricted to channel/rate combinations supported by downstream elements
(e.g. muxers).
void gst_audio_encoder_set_drainable (GstAudioEncoder *enc,gboolean enabled);
Configures encoder drain handling. If drainable, subclass might be handed a NULL buffer to have it return any leftover encoded data. Otherwise, it is not considered so capable and will only ever be passed real data.
MT safe.
void gst_audio_encoder_set_frame_max (GstAudioEncoder *enc,gint num);
Sets max number of frames accepted at once (assumed minimally 1).
Requires frame_samples_min
and frame_samples_max
to be the equal.
Note: This value will be reset to 0 every time before
GstAudioEncoder::set_format() is called.
void gst_audio_encoder_set_frame_samples_min (GstAudioEncoder *enc,gint num);
Sets number of samples (per channel) subclass needs to be handed, at least or will be handed all available if 0.
If an exact number of samples is required, gst_audio_encoder_set_frame_samples_max()
must be called with the same number.
Note: This value will be reset to 0 every time before
GstAudioEncoder::set_format() is called.
void gst_audio_encoder_set_frame_samples_max (GstAudioEncoder *enc,gint num);
Sets number of samples (per channel) subclass needs to be handed, at most or will be handed all available if 0.
If an exact number of samples is required, gst_audio_encoder_set_frame_samples_min()
must be called with the same number.
Note: This value will be reset to 0 every time before
GstAudioEncoder::set_format() is called.
void gst_audio_encoder_set_hard_min (GstAudioEncoder *enc,gboolean enabled);
Configures encoder hard minimum handling. If enabled, subclass will never be handed less samples than it configured, which otherwise might occur near end-of-data handling. Instead, the leftover samples will simply be discarded.
MT safe.
void gst_audio_encoder_set_hard_resync (GstAudioEncoder *enc,gboolean enabled);
void gst_audio_encoder_set_headers (GstAudioEncoder *enc,GList *headers);
Set the codec headers to be sent downstream whenever requested.
enc |
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headers |
a list of GstBuffer containing the codec header. |
[transfer full][element-type Gst.Buffer] |
void gst_audio_encoder_set_latency (GstAudioEncoder *enc,GstClockTime min,GstClockTime max);
Sets encoder latency.
void gst_audio_encoder_set_lookahead (GstAudioEncoder *enc,gint num);
Sets encoder lookahead (in units of input rate samples)
Note: This value will be reset to 0 every time before
GstAudioEncoder::set_format() is called.
void gst_audio_encoder_set_mark_granule (GstAudioEncoder *enc,gboolean enabled);
Enable or disable encoder granule handling.
MT safe.
void gst_audio_encoder_set_perfect_timestamp (GstAudioEncoder *enc,gboolean enabled);
Enable or disable encoder perfect output timestamp preference.
MT safe.
void gst_audio_encoder_set_tolerance (GstAudioEncoder *enc,GstClockTime tolerance);
Configures encoder audio jitter tolerance threshold.
MT safe.
void gst_audio_encoder_merge_tags (GstAudioEncoder *enc,const GstTagList *tags,GstTagMergeMode mode);
Sets the audio encoder tags and how they should be merged with any
upstream stream tags. This will override any tags previously-set
with gst_audio_encoder_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.
MT safe.
enc |
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tags |
a GstTagList to merge, or NULL to unset previously-set tags. |
[allow-none] |
mode |
the GstTagMergeMode to use, usually GST_TAG_MERGE_REPLACE |
struct GstAudioEncoderClass {
GstElementClass element_class;
/* virtual methods for subclasses */
gboolean (*start) (GstAudioEncoder *enc);
gboolean (*stop) (GstAudioEncoder *enc);
gboolean (*set_format) (GstAudioEncoder *enc,
GstAudioInfo *info);
GstFlowReturn (*handle_frame) (GstAudioEncoder *enc,
GstBuffer *buffer);
void (*flush) (GstAudioEncoder *enc);
GstFlowReturn (*pre_push) (GstAudioEncoder *enc,
GstBuffer **buffer);
gboolean (*sink_event) (GstAudioEncoder *enc,
GstEvent *event);
gboolean (*src_event) (GstAudioEncoder *enc,
GstEvent *event);
GstCaps * (*getcaps) (GstAudioEncoder *enc, GstCaps *filter);
gboolean (*open) (GstAudioEncoder *enc);
gboolean (*close) (GstAudioEncoder *enc);
gboolean (*negotiate) (GstAudioEncoder *enc);
gboolean (*decide_allocation) (GstAudioEncoder *enc, GstQuery *query);
gboolean (*propose_allocation) (GstAudioEncoder * enc,
GstQuery * query);
gboolean (*transform_meta) (GstAudioEncoder *enc, GstBuffer *outbuf,
GstMeta *meta, GstBuffer *inbuf);
gboolean (*sink_query) (GstAudioEncoder *encoder,
GstQuery *query);
gboolean (*src_query) (GstAudioEncoder *encoder,
GstQuery *query);
};
Subclasses can override any of the available virtual methods or not, as
needed. At minimum set_format
and handle_frame
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. GstAudioInfo contains the format according to provided caps. |
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Provides input samples (or NULL to clear any remaining data)
according to directions as configured by the subclass
using the API. 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 encoded 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. Allows for a custom sink getcaps implementation (e.g. for multichannel input specification). If not implemented, default returns gst_audio_encoder_proxy_getcaps applied to sink template caps. |
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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. 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 |
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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 |
#define GST_AUDIO_ENCODER_SINK_NAME "sink"
the name of the templates for the sink pad
“hard-resync” property“hard-resync” gboolean
Perform clipping and sample flushing upon discontinuity.
Flags: Read / Write
Default value: FALSE
“mark-granule” property“mark-granule” gboolean
Apply granule semantics to buffer metadata (implies perfect-timestamp).
Flags: Read
Default value: FALSE
“perfect-timestamp” property“perfect-timestamp” gboolean
Favour perfect timestamps over tracking upstream timestamps.
Flags: Read / Write
Default value: FALSE
“tolerance” property“tolerance” gint64
Consider discontinuity if timestamp jitter/imperfection exceeds tolerance (ns).
Flags: Read / Write
Allowed values: >= 0
Default value: 40000000