GstAudioEncoder

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.

As of configuration stage, and throughout processing, GstAudioEncoder maintains various parameters that provide required context, e.g. describing the format of input audio data. Conversely, subclass can and should configure these context parameters to inform base class of its expectation w.r.t. buffer handling.

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.

GstAudioEncoder

GObject
    ╰──GInitiallyUnowned
        ╰──GstObject
            ╰──GstElement
                ╰──GstAudioEncoder

The opaque GstAudioEncoder data structure.

Members

element (GstElement) –
No description available
sinkpad (GstPad *) –
No description available
srcpad (GstPad *) –
No description available
stream_lock (GRecMutex) –
No description available
input_segment (GstSegment) –
No description available
output_segment (GstSegment) –
No description available

Class structure

GstAudioEncoderClass

Subclasses can override any of the available virtual methods or not, as needed. At minimum set_format and handle_frame needs to be overridden.

Fields
element_class (GstElementClass) –

The parent class structure


GstAudio.AudioEncoderClass

Subclasses can override any of the available virtual methods or not, as needed. At minimum set_format and handle_frame needs to be overridden.

Attributes
element_class (Gst.ElementClass) –

The parent class structure


GstAudio.AudioEncoderClass

Subclasses can override any of the available virtual methods or not, as needed. At minimum set_format and handle_frame needs to be overridden.

Attributes
element_class (Gst.ElementClass) –

The parent class structure


GstAudioEncoder

GObject
    ╰──GInitiallyUnowned
        ╰──GstObject
            ╰──GstElement
                ╰──GstAudioEncoder

The opaque GstAudioEncoder data structure.

Members

element (GstElement) –
No description available
sinkpad (GstPad) –
No description available
srcpad (GstPad) –
No description available
stream_lock (GRecMutex) –
No description available
input_segment (GstSegment) –
No description available
output_segment (GstSegment) –
No description available

GstAudioEncoder

GObject
    ╰──GInitiallyUnowned
        ╰──GstObject
            ╰──GstElement
                ╰──GstAudioEncoder

The opaque GstAudioEncoder data structure.

Members

element (GstElement) –
No description available
sinkpad (GstPad) –
No description available
srcpad (GstPad) –
No description available
stream_lock (GRecMutex) –
No description available
input_segment (GstSegment) –
No description available
output_segment (GstSegment) –
No description available

Methods

gst_audio_encoder_allocate_output_buffer

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.

Parameters:

enc

a GstAudioEncoder

size

size of the buffer

Returns ( [transfer: full] ) –

allocated buffer


GstAudio.AudioEncoder.allocate_output_buffer

def GstAudio.AudioEncoder.allocate_output_buffer (self, size):
    #python wrapper for 'gst_audio_encoder_allocate_output_buffer'

Helper function that allocates a buffer to hold an encoded audio frame for enc's current output format.

Parameters:

size ( int ) –

size of the buffer

Returns ( Gst.Buffer ) –

allocated buffer


GstAudio.AudioEncoder.prototype.allocate_output_buffer

function GstAudio.AudioEncoder.prototype.allocate_output_buffer(size: Number): {
    // javascript wrapper for 'gst_audio_encoder_allocate_output_buffer'
}

Helper function that allocates a buffer to hold an encoded audio frame for enc's current output format.

Parameters:

size ( Number ) –

size of the buffer

Returns ( Gst.Buffer ) –

allocated buffer


gst_audio_encoder_finish_frame

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 handle_frame() may be invalidated by a call to this function.

Parameters:

enc

a GstAudioEncoder

buffer

encoded data

samples

number of samples (per channel) represented by encoded data

Returns

a GstFlowReturn that should be escalated to caller (of caller)


GstAudio.AudioEncoder.finish_frame

def GstAudio.AudioEncoder.finish_frame (self, buffer, samples):
    #python wrapper for 'gst_audio_encoder_finish_frame'

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 do_handle_frame() may be invalidated by a call to this function.

Parameters:

buffer ( Gst.Buffer ) –

encoded data

samples ( int ) –

number of samples (per channel) represented by encoded data

Returns ( Gst.FlowReturn ) –

a Gst.FlowReturn that should be escalated to caller (of caller)


GstAudio.AudioEncoder.prototype.finish_frame

function GstAudio.AudioEncoder.prototype.finish_frame(buffer: Gst.Buffer, samples: Number): {
    // javascript wrapper for 'gst_audio_encoder_finish_frame'
}

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 vfunc_handle_frame() may be invalidated by a call to this function.

Parameters:

buffer ( Gst.Buffer ) –

encoded data

samples ( Number ) –

number of samples (per channel) represented by encoded data

Returns ( Gst.FlowReturn ) –

a Gst.FlowReturn that should be escalated to caller (of caller)


gst_audio_encoder_get_allocator

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.

Parameters:

enc

a GstAudioEncoder

allocator ( [out] [allow-none] [transfer: full] ) –

the GstAllocator used

params ( [out] [allow-none] [transfer: full] ) –

the GstAllocationParams of allocator


GstAudio.AudioEncoder.get_allocator

def GstAudio.AudioEncoder.get_allocator (self):
    #python wrapper for 'gst_audio_encoder_get_allocator'

Lets GstAudio.AudioEncoder sub-classes to know the memory allocator used by the base class and its params.

Unref the allocator after use it.

Parameters:


GstAudio.AudioEncoder.prototype.get_allocator

function GstAudio.AudioEncoder.prototype.get_allocator(): {
    // javascript wrapper for 'gst_audio_encoder_get_allocator'
}

Lets GstAudio.AudioEncoder sub-classes to know the memory allocator used by the base class and its params.

Unref the allocator after use it.

Parameters:


gst_audio_encoder_get_audio_info

GstAudioInfo *
gst_audio_encoder_get_audio_info (GstAudioEncoder * enc)

Parameters:

enc

a GstAudioEncoder

Returns

a GstAudioInfo describing the input audio format


GstAudio.AudioEncoder.get_audio_info

def GstAudio.AudioEncoder.get_audio_info (self):
    #python wrapper for 'gst_audio_encoder_get_audio_info'

Parameters:

Returns ( GstAudio.AudioInfo ) –

a GstAudio.AudioInfo describing the input audio format


GstAudio.AudioEncoder.prototype.get_audio_info

function GstAudio.AudioEncoder.prototype.get_audio_info(): {
    // javascript wrapper for 'gst_audio_encoder_get_audio_info'
}

Parameters:

Returns ( GstAudio.AudioInfo ) –

a GstAudio.AudioInfo describing the input audio format


gst_audio_encoder_get_drainable

gboolean
gst_audio_encoder_get_drainable (GstAudioEncoder * enc)

Queries encoder drain handling.

Parameters:

enc

a GstAudioEncoder

Returns

TRUE if drainable handling is enabled.

MT safe.


GstAudio.AudioEncoder.get_drainable

def GstAudio.AudioEncoder.get_drainable (self):
    #python wrapper for 'gst_audio_encoder_get_drainable'

Queries encoder drain handling.

Parameters:

Returns ( bool ) –

TRUE if drainable handling is enabled.

MT safe.


GstAudio.AudioEncoder.prototype.get_drainable

function GstAudio.AudioEncoder.prototype.get_drainable(): {
    // javascript wrapper for 'gst_audio_encoder_get_drainable'
}

Queries encoder drain handling.

Parameters:

Returns ( Number ) –

TRUE if drainable handling is enabled.

MT safe.


gst_audio_encoder_get_frame_max

gint
gst_audio_encoder_get_frame_max (GstAudioEncoder * enc)

Parameters:

enc

a GstAudioEncoder

Returns

currently configured maximum handled frames


GstAudio.AudioEncoder.get_frame_max

def GstAudio.AudioEncoder.get_frame_max (self):
    #python wrapper for 'gst_audio_encoder_get_frame_max'

Parameters:

Returns ( int ) –

currently configured maximum handled frames


GstAudio.AudioEncoder.prototype.get_frame_max

function GstAudio.AudioEncoder.prototype.get_frame_max(): {
    // javascript wrapper for 'gst_audio_encoder_get_frame_max'
}

Parameters:

Returns ( Number ) –

currently configured maximum handled frames


gst_audio_encoder_get_frame_samples_max

gint
gst_audio_encoder_get_frame_samples_max (GstAudioEncoder * enc)

Parameters:

enc

a GstAudioEncoder

Returns

currently maximum requested samples per frame


GstAudio.AudioEncoder.get_frame_samples_max

def GstAudio.AudioEncoder.get_frame_samples_max (self):
    #python wrapper for 'gst_audio_encoder_get_frame_samples_max'

Parameters:

Returns ( int ) –

currently maximum requested samples per frame


GstAudio.AudioEncoder.prototype.get_frame_samples_max

function GstAudio.AudioEncoder.prototype.get_frame_samples_max(): {
    // javascript wrapper for 'gst_audio_encoder_get_frame_samples_max'
}

Parameters:

Returns ( Number ) –

currently maximum requested samples per frame


gst_audio_encoder_get_frame_samples_min

gint
gst_audio_encoder_get_frame_samples_min (GstAudioEncoder * enc)

Parameters:

enc

a GstAudioEncoder

Returns

currently minimum requested samples per frame


GstAudio.AudioEncoder.get_frame_samples_min

def GstAudio.AudioEncoder.get_frame_samples_min (self):
    #python wrapper for 'gst_audio_encoder_get_frame_samples_min'

Parameters:

Returns ( int ) –

currently minimum requested samples per frame


GstAudio.AudioEncoder.prototype.get_frame_samples_min

function GstAudio.AudioEncoder.prototype.get_frame_samples_min(): {
    // javascript wrapper for 'gst_audio_encoder_get_frame_samples_min'
}

Parameters:

Returns ( Number ) –

currently minimum requested samples per frame


gst_audio_encoder_get_hard_min

gboolean
gst_audio_encoder_get_hard_min (GstAudioEncoder * enc)

Queries encoder hard minimum handling.

Parameters:

enc

a GstAudioEncoder

Returns

TRUE if hard minimum handling is enabled.

MT safe.


GstAudio.AudioEncoder.get_hard_min

def GstAudio.AudioEncoder.get_hard_min (self):
    #python wrapper for 'gst_audio_encoder_get_hard_min'

Queries encoder hard minimum handling.

Parameters:

Returns ( bool ) –

TRUE if hard minimum handling is enabled.

MT safe.


GstAudio.AudioEncoder.prototype.get_hard_min

function GstAudio.AudioEncoder.prototype.get_hard_min(): {
    // javascript wrapper for 'gst_audio_encoder_get_hard_min'
}

Queries encoder hard minimum handling.

Parameters:

Returns ( Number ) –

TRUE if hard minimum handling is enabled.

MT safe.


gst_audio_encoder_get_hard_resync

gboolean
gst_audio_encoder_get_hard_resync (GstAudioEncoder * enc)

Parameters:

enc
No description available
Returns
No description available

GstAudio.AudioEncoder.get_hard_resync

def GstAudio.AudioEncoder.get_hard_resync (self):
    #python wrapper for 'gst_audio_encoder_get_hard_resync'

Parameters:

enc ( GstAudio.AudioEncoder ) –
No description available
Returns ( bool ) –
No description available

GstAudio.AudioEncoder.prototype.get_hard_resync

function GstAudio.AudioEncoder.prototype.get_hard_resync(): {
    // javascript wrapper for 'gst_audio_encoder_get_hard_resync'
}

Parameters:

enc ( GstAudio.AudioEncoder ) –
No description available
Returns ( Number ) –
No description available

gst_audio_encoder_get_latency

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.

Parameters:

enc

a GstAudioEncoder

min ( [out] [allow-none] ) –

a pointer to storage to hold minimum latency

max ( [out] [allow-none] ) –

a pointer to storage to hold maximum latency


GstAudio.AudioEncoder.get_latency

def GstAudio.AudioEncoder.get_latency (self):
    #python wrapper for 'gst_audio_encoder_get_latency'

Sets the variables pointed to by min and max to the currently configured latency.

Parameters:


GstAudio.AudioEncoder.prototype.get_latency

function GstAudio.AudioEncoder.prototype.get_latency(): {
    // javascript wrapper for 'gst_audio_encoder_get_latency'
}

Sets the variables pointed to by min and max to the currently configured latency.

Parameters:


gst_audio_encoder_get_lookahead

gint
gst_audio_encoder_get_lookahead (GstAudioEncoder * enc)

Parameters:

enc

a GstAudioEncoder

Returns

currently configured encoder lookahead


GstAudio.AudioEncoder.get_lookahead

def GstAudio.AudioEncoder.get_lookahead (self):
    #python wrapper for 'gst_audio_encoder_get_lookahead'

Parameters:

Returns ( int ) –

currently configured encoder lookahead


GstAudio.AudioEncoder.prototype.get_lookahead

function GstAudio.AudioEncoder.prototype.get_lookahead(): {
    // javascript wrapper for 'gst_audio_encoder_get_lookahead'
}

Parameters:

Returns ( Number ) –

currently configured encoder lookahead


gst_audio_encoder_get_mark_granule

gboolean
gst_audio_encoder_get_mark_granule (GstAudioEncoder * enc)

Queries if the encoder will handle granule marking.

Parameters:

enc

a GstAudioEncoder

Returns

TRUE if granule marking is enabled.

MT safe.


GstAudio.AudioEncoder.get_mark_granule

def GstAudio.AudioEncoder.get_mark_granule (self):
    #python wrapper for 'gst_audio_encoder_get_mark_granule'

Queries if the encoder will handle granule marking.

Parameters:

Returns ( bool ) –

TRUE if granule marking is enabled.

MT safe.


GstAudio.AudioEncoder.prototype.get_mark_granule

function GstAudio.AudioEncoder.prototype.get_mark_granule(): {
    // javascript wrapper for 'gst_audio_encoder_get_mark_granule'
}

Queries if the encoder will handle granule marking.

Parameters:

Returns ( Number ) –

TRUE if granule marking is enabled.

MT safe.


gst_audio_encoder_get_perfect_timestamp

gboolean
gst_audio_encoder_get_perfect_timestamp (GstAudioEncoder * enc)

Queries encoder perfect timestamp behaviour.

Parameters:

enc

a GstAudioEncoder

Returns

TRUE if perfect timestamp setting enabled.

MT safe.


GstAudio.AudioEncoder.get_perfect_timestamp

def GstAudio.AudioEncoder.get_perfect_timestamp (self):
    #python wrapper for 'gst_audio_encoder_get_perfect_timestamp'

Queries encoder perfect timestamp behaviour.

Parameters:

Returns ( bool ) –

TRUE if perfect timestamp setting enabled.

MT safe.


GstAudio.AudioEncoder.prototype.get_perfect_timestamp

function GstAudio.AudioEncoder.prototype.get_perfect_timestamp(): {
    // javascript wrapper for 'gst_audio_encoder_get_perfect_timestamp'
}

Queries encoder perfect timestamp behaviour.

Parameters:

Returns ( Number ) –

TRUE if perfect timestamp setting enabled.

MT safe.


gst_audio_encoder_get_tolerance

GstClockTime
gst_audio_encoder_get_tolerance (GstAudioEncoder * enc)

Queries current audio jitter tolerance threshold.

Parameters:

enc

a GstAudioEncoder

Returns

encoder audio jitter tolerance threshold.

MT safe.


GstAudio.AudioEncoder.get_tolerance

def GstAudio.AudioEncoder.get_tolerance (self):
    #python wrapper for 'gst_audio_encoder_get_tolerance'

Queries current audio jitter tolerance threshold.

Parameters:

Returns ( int ) –

encoder audio jitter tolerance threshold.

MT safe.


GstAudio.AudioEncoder.prototype.get_tolerance

function GstAudio.AudioEncoder.prototype.get_tolerance(): {
    // javascript wrapper for 'gst_audio_encoder_get_tolerance'
}

Queries current audio jitter tolerance threshold.

Parameters:

Returns ( Number ) –

encoder audio jitter tolerance threshold.

MT safe.


gst_audio_encoder_merge_tags

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.

Parameters:

enc

a GstAudioEncoder

tags ( [allow-none] ) –

a GstTagList to merge, or NULL to unset previously-set tags

mode

the GstTagMergeMode to use, usually GST_TAG_MERGE_REPLACE


GstAudio.AudioEncoder.merge_tags

def GstAudio.AudioEncoder.merge_tags (self, tags, mode):
    #python wrapper for 'gst_audio_encoder_merge_tags'

Sets the audio encoder tags and how they should be merged with any upstream stream tags. This will override any tags previously-set with GstAudio.AudioEncoder.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.

Parameters:

tags ( Gst.TagList ) –

a Gst.TagList to merge, or NULL to unset previously-set tags


GstAudio.AudioEncoder.prototype.merge_tags

function GstAudio.AudioEncoder.prototype.merge_tags(tags: Gst.TagList, mode: Gst.TagMergeMode): {
    // javascript wrapper for 'gst_audio_encoder_merge_tags'
}

Sets the audio encoder tags and how they should be merged with any upstream stream tags. This will override any tags previously-set with GstAudio.AudioEncoder.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.

MT safe.

Parameters:

tags ( Gst.TagList ) –

a Gst.TagList to merge, or NULL to unset previously-set tags


gst_audio_encoder_negotiate

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.

Parameters:

enc

a GstAudioEncoder

Returns

TRUE if the negotiation succeeded, else FALSE.


GstAudio.AudioEncoder.negotiate

def GstAudio.AudioEncoder.negotiate (self):
    #python wrapper for 'gst_audio_encoder_negotiate'

Negotiate with downstream elements to currently configured Gst.Caps. Unmark GST_PAD_FLAG_NEED_RECONFIGURE in any case. But mark it again if negotiate fails.

Parameters:

Returns ( bool ) –

True if the negotiation succeeded, else False.


GstAudio.AudioEncoder.prototype.negotiate

function GstAudio.AudioEncoder.prototype.negotiate(): {
    // javascript wrapper for 'gst_audio_encoder_negotiate'
}

Negotiate with downstream elements to currently configured Gst.Caps. Unmark GST_PAD_FLAG_NEED_RECONFIGURE in any case. But mark it again if negotiate fails.

Parameters:

Returns ( Number ) –

true if the negotiation succeeded, else false.


gst_audio_encoder_proxy_getcaps

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).

Parameters:

enc

a GstAudioEncoder

caps ( [allow-none] ) –

initial caps

filter ( [allow-none] ) –

filter caps

Returns ( [transfer: full] ) –

a GstCaps owned by caller


GstAudio.AudioEncoder.proxy_getcaps

def GstAudio.AudioEncoder.proxy_getcaps (self, caps, filter):
    #python wrapper for 'gst_audio_encoder_proxy_getcaps'

Returns caps that express caps (or sink template caps if caps == NULL) restricted to channel/rate combinations supported by downstream elements (e.g. muxers).

Parameters:

caps ( Gst.Caps ) –

initial caps

filter ( Gst.Caps ) –

filter caps

Returns ( Gst.Caps ) –

a Gst.Caps owned by caller


GstAudio.AudioEncoder.prototype.proxy_getcaps

function GstAudio.AudioEncoder.prototype.proxy_getcaps(caps: Gst.Caps, filter: Gst.Caps): {
    // javascript wrapper for 'gst_audio_encoder_proxy_getcaps'
}

Returns caps that express caps (or sink template caps if caps == NULL) restricted to channel/rate combinations supported by downstream elements (e.g. muxers).

Parameters:

caps ( Gst.Caps ) –

initial caps

filter ( Gst.Caps ) –

filter caps

Returns ( Gst.Caps ) –

a Gst.Caps owned by caller


gst_audio_encoder_set_allocation_caps

gst_audio_encoder_set_allocation_caps (GstAudioEncoder * enc,
                                       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_encoder_negotiate. Setting to NULL the allocation query will use the caps from the pad.

Parameters:

enc

a GstAudioEncoder

allocation_caps ( [allow-none] ) –

a GstCaps or NULL

Since : 1.10


GstAudio.AudioEncoder.set_allocation_caps

def GstAudio.AudioEncoder.set_allocation_caps (self, allocation_caps):
    #python wrapper for 'gst_audio_encoder_set_allocation_caps'

Sets a caps in allocation query which are different from the set pad's caps. Use this function before calling GstAudio.AudioEncoder.negotiate. Setting to None the allocation query will use the caps from the pad.

Parameters:

allocation_caps ( Gst.Caps ) –

a Gst.Caps or None

Since : 1.10


GstAudio.AudioEncoder.prototype.set_allocation_caps

function GstAudio.AudioEncoder.prototype.set_allocation_caps(allocation_caps: Gst.Caps): {
    // javascript wrapper for 'gst_audio_encoder_set_allocation_caps'
}

Sets a caps in allocation query which are different from the set pad's caps. Use this function before calling GstAudio.AudioEncoder.prototype.negotiate. Setting to null the allocation query will use the caps from the pad.

Parameters:

allocation_caps ( Gst.Caps ) –

a Gst.Caps or null

Since : 1.10


gst_audio_encoder_set_drainable

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.

Parameters:

enc

a GstAudioEncoder

enabled

new state


GstAudio.AudioEncoder.set_drainable

def GstAudio.AudioEncoder.set_drainable (self, enabled):
    #python wrapper for 'gst_audio_encoder_set_drainable'

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.

Parameters:

enabled ( bool ) –

new state


GstAudio.AudioEncoder.prototype.set_drainable

function GstAudio.AudioEncoder.prototype.set_drainable(enabled: Number): {
    // javascript wrapper for 'gst_audio_encoder_set_drainable'
}

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.

Parameters:

enabled ( Number ) –

new state


gst_audio_encoder_set_frame_max

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 set_format() is called.

Parameters:

enc

a GstAudioEncoder

num

number of frames


GstAudio.AudioEncoder.set_frame_max

def GstAudio.AudioEncoder.set_frame_max (self, num):
    #python wrapper for 'gst_audio_encoder_set_frame_max'

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 do_set_format() is called.

Parameters:

num ( int ) –

number of frames


GstAudio.AudioEncoder.prototype.set_frame_max

function GstAudio.AudioEncoder.prototype.set_frame_max(num: Number): {
    // javascript wrapper for 'gst_audio_encoder_set_frame_max'
}

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 vfunc_set_format() is called.

Parameters:

num ( Number ) –

number of frames


gst_audio_encoder_set_frame_samples_max

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 set_format() is called.

Parameters:

enc

a GstAudioEncoder

num

number of samples per frame


GstAudio.AudioEncoder.set_frame_samples_max

def GstAudio.AudioEncoder.set_frame_samples_max (self, num):
    #python wrapper for 'gst_audio_encoder_set_frame_samples_max'

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, GstAudio.AudioEncoder.set_frame_samples_min must be called with the same number.

Note: This value will be reset to 0 every time before do_set_format() is called.

Parameters:

num ( int ) –

number of samples per frame


GstAudio.AudioEncoder.prototype.set_frame_samples_max

function GstAudio.AudioEncoder.prototype.set_frame_samples_max(num: Number): {
    // javascript wrapper for 'gst_audio_encoder_set_frame_samples_max'
}

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, GstAudio.AudioEncoder.prototype.set_frame_samples_min must be called with the same number.

Note: This value will be reset to 0 every time before vfunc_set_format() is called.

Parameters:

num ( Number ) –

number of samples per frame


gst_audio_encoder_set_frame_samples_min

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 set_format() is called.

Parameters:

enc

a GstAudioEncoder

num

number of samples per frame


GstAudio.AudioEncoder.set_frame_samples_min

def GstAudio.AudioEncoder.set_frame_samples_min (self, num):
    #python wrapper for 'gst_audio_encoder_set_frame_samples_min'

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, GstAudio.AudioEncoder.set_frame_samples_max must be called with the same number.

Note: This value will be reset to 0 every time before do_set_format() is called.

Parameters:

num ( int ) –

number of samples per frame


GstAudio.AudioEncoder.prototype.set_frame_samples_min

function GstAudio.AudioEncoder.prototype.set_frame_samples_min(num: Number): {
    // javascript wrapper for 'gst_audio_encoder_set_frame_samples_min'
}

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, GstAudio.AudioEncoder.prototype.set_frame_samples_max must be called with the same number.

Note: This value will be reset to 0 every time before vfunc_set_format() is called.

Parameters:

num ( Number ) –

number of samples per frame


gst_audio_encoder_set_hard_min

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.

Parameters:

enc

a GstAudioEncoder

enabled

new state


GstAudio.AudioEncoder.set_hard_min

def GstAudio.AudioEncoder.set_hard_min (self, enabled):
    #python wrapper for 'gst_audio_encoder_set_hard_min'

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.

Parameters:

enabled ( bool ) –

new state


GstAudio.AudioEncoder.prototype.set_hard_min

function GstAudio.AudioEncoder.prototype.set_hard_min(enabled: Number): {
    // javascript wrapper for 'gst_audio_encoder_set_hard_min'
}

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.

Parameters:

enabled ( Number ) –

new state


gst_audio_encoder_set_hard_resync

gst_audio_encoder_set_hard_resync (GstAudioEncoder * enc,
                                   gboolean enabled)

Parameters:

enc
No description available
enabled
No description available

GstAudio.AudioEncoder.set_hard_resync

def GstAudio.AudioEncoder.set_hard_resync (self, enabled):
    #python wrapper for 'gst_audio_encoder_set_hard_resync'

Parameters:

enc ( GstAudio.AudioEncoder ) –
No description available
enabled ( bool ) –
No description available

GstAudio.AudioEncoder.prototype.set_hard_resync

function GstAudio.AudioEncoder.prototype.set_hard_resync(enabled: Number): {
    // javascript wrapper for 'gst_audio_encoder_set_hard_resync'
}

Parameters:

enc ( GstAudio.AudioEncoder ) –
No description available
enabled ( Number ) –
No description available

gst_audio_encoder_set_headers

gst_audio_encoder_set_headers (GstAudioEncoder * enc,
                               GList * headers)

Set the codec headers to be sent downstream whenever requested.

Parameters:

enc

a GstAudioEncoder

headers ( [transfer: full] [element-type Gst.Buffer] ) –

a list of GstBuffer containing the codec header


GstAudio.AudioEncoder.set_headers

def GstAudio.AudioEncoder.set_headers (self, headers):
    #python wrapper for 'gst_audio_encoder_set_headers'

Set the codec headers to be sent downstream whenever requested.

Parameters:

headers ( [ Gst.Buffer ] ) –

a list of Gst.Buffer containing the codec header


GstAudio.AudioEncoder.prototype.set_headers

function GstAudio.AudioEncoder.prototype.set_headers(headers: [ Gst.Buffer ]): {
    // javascript wrapper for 'gst_audio_encoder_set_headers'
}

Set the codec headers to be sent downstream whenever requested.

Parameters:

headers ( [ Gst.Buffer ] ) –

a list of Gst.Buffer containing the codec header


gst_audio_encoder_set_latency

gst_audio_encoder_set_latency (GstAudioEncoder * enc,
                               GstClockTime min,
                               GstClockTime max)

Sets encoder latency.

Parameters:

enc

a GstAudioEncoder

min

minimum latency

max

maximum latency


GstAudio.AudioEncoder.set_latency

def GstAudio.AudioEncoder.set_latency (self, min, max):
    #python wrapper for 'gst_audio_encoder_set_latency'

Sets encoder latency.

Parameters:

min ( int ) –

minimum latency

max ( int ) –

maximum latency


GstAudio.AudioEncoder.prototype.set_latency

function GstAudio.AudioEncoder.prototype.set_latency(min: Number, max: Number): {
    // javascript wrapper for 'gst_audio_encoder_set_latency'
}

Sets encoder latency.

Parameters:

min ( Number ) –

minimum latency

max ( Number ) –

maximum latency


gst_audio_encoder_set_lookahead

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 set_format() is called.

Parameters:

enc

a GstAudioEncoder

num

lookahead


GstAudio.AudioEncoder.set_lookahead

def GstAudio.AudioEncoder.set_lookahead (self, num):
    #python wrapper for 'gst_audio_encoder_set_lookahead'

Sets encoder lookahead (in units of input rate samples)

Note: This value will be reset to 0 every time before do_set_format() is called.

Parameters:

num ( int ) –

lookahead


GstAudio.AudioEncoder.prototype.set_lookahead

function GstAudio.AudioEncoder.prototype.set_lookahead(num: Number): {
    // javascript wrapper for 'gst_audio_encoder_set_lookahead'
}

Sets encoder lookahead (in units of input rate samples)

Note: This value will be reset to 0 every time before vfunc_set_format() is called.

Parameters:

num ( Number ) –

lookahead


gst_audio_encoder_set_mark_granule

gst_audio_encoder_set_mark_granule (GstAudioEncoder * enc,
                                    gboolean enabled)

Enable or disable encoder granule handling.

MT safe.

Parameters:

enc

a GstAudioEncoder

enabled

new state


GstAudio.AudioEncoder.set_mark_granule

def GstAudio.AudioEncoder.set_mark_granule (self, enabled):
    #python wrapper for 'gst_audio_encoder_set_mark_granule'

Enable or disable encoder granule handling.

MT safe.

Parameters:

enabled ( bool ) –

new state


GstAudio.AudioEncoder.prototype.set_mark_granule

function GstAudio.AudioEncoder.prototype.set_mark_granule(enabled: Number): {
    // javascript wrapper for 'gst_audio_encoder_set_mark_granule'
}

Enable or disable encoder granule handling.

MT safe.

Parameters:

enabled ( Number ) –

new state


gst_audio_encoder_set_output_format

gboolean
gst_audio_encoder_set_output_format (GstAudioEncoder * enc,
                                     GstCaps * caps)

Configure output caps on the srcpad of enc.

Parameters:

enc

a GstAudioEncoder

caps ( [transfer: none] ) –

GstCaps

Returns

TRUE on success.


GstAudio.AudioEncoder.set_output_format

def GstAudio.AudioEncoder.set_output_format (self, caps):
    #python wrapper for 'gst_audio_encoder_set_output_format'

Configure output caps on the srcpad of enc.

Parameters:

caps ( Gst.Caps ) –

Gst.Caps

Returns ( bool ) –

True on success.


GstAudio.AudioEncoder.prototype.set_output_format

function GstAudio.AudioEncoder.prototype.set_output_format(caps: Gst.Caps): {
    // javascript wrapper for 'gst_audio_encoder_set_output_format'
}

Configure output caps on the srcpad of enc.

Parameters:

caps ( Gst.Caps ) –

Gst.Caps

Returns ( Number ) –

true on success.


gst_audio_encoder_set_perfect_timestamp

gst_audio_encoder_set_perfect_timestamp (GstAudioEncoder * enc,
                                         gboolean enabled)

Enable or disable encoder perfect output timestamp preference.

MT safe.

Parameters:

enc

a GstAudioEncoder

enabled

new state


GstAudio.AudioEncoder.set_perfect_timestamp

def GstAudio.AudioEncoder.set_perfect_timestamp (self, enabled):
    #python wrapper for 'gst_audio_encoder_set_perfect_timestamp'

Enable or disable encoder perfect output timestamp preference.

MT safe.

Parameters:

enabled ( bool ) –

new state


GstAudio.AudioEncoder.prototype.set_perfect_timestamp

function GstAudio.AudioEncoder.prototype.set_perfect_timestamp(enabled: Number): {
    // javascript wrapper for 'gst_audio_encoder_set_perfect_timestamp'
}

Enable or disable encoder perfect output timestamp preference.

MT safe.

Parameters:

enabled ( Number ) –

new state


gst_audio_encoder_set_tolerance

gst_audio_encoder_set_tolerance (GstAudioEncoder * enc,
                                 GstClockTime tolerance)

Configures encoder audio jitter tolerance threshold.

MT safe.

Parameters:

enc

a GstAudioEncoder

tolerance

new tolerance


GstAudio.AudioEncoder.set_tolerance

def GstAudio.AudioEncoder.set_tolerance (self, tolerance):
    #python wrapper for 'gst_audio_encoder_set_tolerance'

Configures encoder audio jitter tolerance threshold.

MT safe.

Parameters:

tolerance ( int ) –

new tolerance


GstAudio.AudioEncoder.prototype.set_tolerance

function GstAudio.AudioEncoder.prototype.set_tolerance(tolerance: Number): {
    // javascript wrapper for 'gst_audio_encoder_set_tolerance'
}

Configures encoder audio jitter tolerance threshold.

MT safe.

Parameters:

tolerance ( Number ) –

new tolerance


Properties

hard-resync

“hard-resync” gboolean

Flags : Read / Write


hard_resync

“self.props.hard_resync” bool

Flags : Read / Write


hard-resync

“hard-resync” Number

Flags : Read / Write


mark-granule

“mark-granule” gboolean

Flags : Read


mark_granule

“self.props.mark_granule” bool

Flags : Read


mark-granule

“mark-granule” Number

Flags : Read


perfect-timestamp

“perfect-timestamp” gboolean

Flags : Read / Write


perfect_timestamp

“self.props.perfect_timestamp” bool

Flags : Read / Write


perfect-timestamp

“perfect-timestamp” Number

Flags : Read / Write


tolerance

“tolerance” gint64

Flags : Read / Write


tolerance

“self.props.tolerance” int

Flags : Read / Write


tolerance

“tolerance” Number

Flags : Read / Write


Virtual Methods

close

gboolean
close (GstAudioEncoder * enc)

Optional. Called when the element changes to GST_STATE_NULL. Allows closing external resources.

Parameters:

enc
No description available
Returns
No description available

do_close

def do_close (enc):
    #python implementation of the 'close' virtual method

Optional. Called when the element changes to GST_STATE_NULL. Allows closing external resources.

Parameters:

enc ( GstAudio.AudioEncoder ) –
No description available
Returns ( bool ) –
No description available

vfunc_close

function vfunc_close(enc: GstAudio.AudioEncoder): {
    // javascript implementation of the 'close' virtual method
}

Optional. Called when the element changes to GST_STATE_NULL. Allows closing external resources.

Parameters:

enc ( GstAudio.AudioEncoder ) –
No description available
Returns ( Number ) –
No description available

decide_allocation

gboolean
decide_allocation (GstAudioEncoder * enc,
                   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:

enc
No description available
query
No description available
Returns
No description available

do_decide_allocation

def do_decide_allocation (enc, 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:

enc ( GstAudio.AudioEncoder ) –
No description available
query ( Gst.Query ) –
No description available
Returns ( bool ) –
No description available

vfunc_decide_allocation

function vfunc_decide_allocation(enc: GstAudio.AudioEncoder, 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:

enc ( GstAudio.AudioEncoder ) –
No description available
query ( Gst.Query ) –
No description available
Returns ( Number ) –
No description available

flush

flush (GstAudioEncoder * enc)

Optional. Instructs subclass to clear any codec caches and discard any pending samples and not yet returned encoded data.

Parameters:

enc
No description available

do_flush

def do_flush (enc):
    #python implementation of the 'flush' virtual method

Optional. Instructs subclass to clear any codec caches and discard any pending samples and not yet returned encoded data.

Parameters:

enc ( GstAudio.AudioEncoder ) –
No description available

vfunc_flush

function vfunc_flush(enc: GstAudio.AudioEncoder): {
    // javascript implementation of the 'flush' virtual method
}

Optional. Instructs subclass to clear any codec caches and discard any pending samples and not yet returned encoded data.

Parameters:

enc ( GstAudio.AudioEncoder ) –
No description available

getcaps

GstCaps *
getcaps (GstAudioEncoder * enc,
         GstCaps * filter)

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.

Parameters:

enc
No description available
filter
No description available
Returns
No description available

do_getcaps

def do_getcaps (enc, filter):
    #python implementation of the 'getcaps' virtual method

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.

Parameters:

enc ( GstAudio.AudioEncoder ) –
No description available
filter ( Gst.Caps ) –
No description available
Returns ( Gst.Caps ) –
No description available

vfunc_getcaps

function vfunc_getcaps(enc: GstAudio.AudioEncoder, filter: Gst.Caps): {
    // javascript implementation of the 'getcaps' virtual method
}

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.

Parameters:

enc ( GstAudio.AudioEncoder ) –
No description available
filter ( Gst.Caps ) –
No description available
Returns ( Gst.Caps ) –
No description available

handle_frame

GstFlowReturn
handle_frame (GstAudioEncoder * enc,
              GstBuffer * buffer)

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 gst_audio_encoder_finish_frame.

Parameters:

enc
No description available
buffer
No description available
Returns
No description available

do_handle_frame

def do_handle_frame (enc, buffer):
    #python implementation of the 'handle_frame' virtual method

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 GstAudio.AudioEncoder.finish_frame.

Parameters:

enc ( GstAudio.AudioEncoder ) –
No description available
buffer ( Gst.Buffer ) –
No description available
Returns ( Gst.FlowReturn ) –
No description available

vfunc_handle_frame

function vfunc_handle_frame(enc: GstAudio.AudioEncoder, buffer: Gst.Buffer): {
    // javascript implementation of the 'handle_frame' virtual method
}

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 GstAudio.AudioEncoder.prototype.finish_frame.

Parameters:

enc ( GstAudio.AudioEncoder ) –
No description available
buffer ( Gst.Buffer ) –
No description available
Returns ( Gst.FlowReturn ) –
No description available

negotiate

gboolean
negotiate (GstAudioEncoder * enc)

Optional. Negotiate with downstream and configure buffer pools, etc. Subclasses should chain up to the parent implementation to invoke the default handler.

Parameters:

enc
No description available
Returns
No description available

do_negotiate

def do_negotiate (enc):
    #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:

enc ( GstAudio.AudioEncoder ) –
No description available
Returns ( bool ) –
No description available

vfunc_negotiate

function vfunc_negotiate(enc: GstAudio.AudioEncoder): {
    // 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:

enc ( GstAudio.AudioEncoder ) –
No description available
Returns ( Number ) –
No description available

open

gboolean
open (GstAudioEncoder * enc)

Optional. Called when the element changes to GST_STATE_READY. Allows opening external resources.

Parameters:

enc
No description available
Returns
No description available

do_open

def do_open (enc):
    #python implementation of the 'open' virtual method

Optional. Called when the element changes to GST_STATE_READY. Allows opening external resources.

Parameters:

enc ( GstAudio.AudioEncoder ) –
No description available
Returns ( bool ) –
No description available

vfunc_open

function vfunc_open(enc: GstAudio.AudioEncoder): {
    // javascript implementation of the 'open' virtual method
}

Optional. Called when the element changes to GST_STATE_READY. Allows opening external resources.

Parameters:

enc ( GstAudio.AudioEncoder ) –
No description available
Returns ( Number ) –
No description available

pre_push

GstFlowReturn
pre_push (GstAudioEncoder * enc,
          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:

enc
No description available
buffer
No description available
Returns
No description available

do_pre_push

def do_pre_push (enc, 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:

enc ( GstAudio.AudioEncoder ) –
No description available
buffer ( Gst.Buffer ) –
No description available
Returns ( Gst.FlowReturn ) –
No description available

vfunc_pre_push

function vfunc_pre_push(enc: GstAudio.AudioEncoder, 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:

enc ( GstAudio.AudioEncoder ) –
No description available
buffer ( Gst.Buffer ) –
No description available
Returns ( Gst.FlowReturn ) –
No description available

propose_allocation

gboolean
propose_allocation (GstAudioEncoder * enc,
                    GstQuery * query)

Optional. Propose buffer allocation parameters for upstream elements. Subclasses should chain up to the parent implementation to invoke the default handler.

Parameters:

enc
No description available
query
No description available
Returns
No description available

do_propose_allocation

def do_propose_allocation (enc, 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:

enc ( GstAudio.AudioEncoder ) –
No description available
query ( Gst.Query ) –
No description available
Returns ( bool ) –
No description available

vfunc_propose_allocation

function vfunc_propose_allocation(enc: GstAudio.AudioEncoder, 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:

enc ( GstAudio.AudioEncoder ) –
No description available
query ( Gst.Query ) –
No description available
Returns ( Number ) –
No description available

set_format

gboolean
set_format (GstAudioEncoder * enc,
            GstAudioInfo * info)

Notifies subclass of incoming data format. GstAudioInfo contains the format according to provided caps.

Parameters:

enc
No description available
info
No description available
Returns
No description available

do_set_format

def do_set_format (enc, info):
    #python implementation of the 'set_format' virtual method

Notifies subclass of incoming data format. GstAudioInfo contains the format according to provided caps.

Parameters:

enc ( GstAudio.AudioEncoder ) –
No description available
info ( GstAudio.AudioInfo ) –
No description available
Returns ( bool ) –
No description available

vfunc_set_format

function vfunc_set_format(enc: GstAudio.AudioEncoder, info: GstAudio.AudioInfo): {
    // javascript implementation of the 'set_format' virtual method
}

Notifies subclass of incoming data format. GstAudioInfo contains the format according to provided caps.

Parameters:

enc ( GstAudio.AudioEncoder ) –
No description available
info ( GstAudio.AudioInfo ) –
No description available
Returns ( Number ) –
No description available

sink_event

gboolean
sink_event (GstAudioEncoder * enc,
            GstEvent * event)

Optional. Event handler on the sink pad. Subclasses should chain up to the parent implementation to invoke the default handler.

Parameters:

enc
No description available
event
No description available
Returns
No description available

do_sink_event

def do_sink_event (enc, 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:

enc ( GstAudio.AudioEncoder ) –
No description available
event ( Gst.Event ) –
No description available
Returns ( bool ) –
No description available

vfunc_sink_event

function vfunc_sink_event(enc: GstAudio.AudioEncoder, 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:

enc ( GstAudio.AudioEncoder ) –
No description available
event ( Gst.Event ) –
No description available
Returns ( Number ) –
No description available

sink_query

gboolean
sink_query (GstAudioEncoder * encoder,
            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:

encoder
No description available
query
No description available
Returns
No description available

do_sink_query

def do_sink_query (encoder, 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:

encoder ( GstAudio.AudioEncoder ) –
No description available
query ( Gst.Query ) –
No description available
Returns ( bool ) –
No description available

vfunc_sink_query

function vfunc_sink_query(encoder: GstAudio.AudioEncoder, 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:

encoder ( GstAudio.AudioEncoder ) –
No description available
query ( Gst.Query ) –
No description available
Returns ( Number ) –
No description available

src_event

gboolean
src_event (GstAudioEncoder * enc,
           GstEvent * event)

Optional. Event handler on the src pad. Subclasses should chain up to the parent implementation to invoke the default handler.

Parameters:

enc
No description available
event
No description available
Returns
No description available

do_src_event

def do_src_event (enc, 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:

enc ( GstAudio.AudioEncoder ) –
No description available
event ( Gst.Event ) –
No description available
Returns ( bool ) –
No description available

vfunc_src_event

function vfunc_src_event(enc: GstAudio.AudioEncoder, 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:

enc ( GstAudio.AudioEncoder ) –
No description available
event ( Gst.Event ) –
No description available
Returns ( Number ) –
No description available

src_query

gboolean
src_query (GstAudioEncoder * encoder,
           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:

encoder
No description available
query
No description available
Returns
No description available

do_src_query

def do_src_query (encoder, 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:

encoder ( GstAudio.AudioEncoder ) –
No description available
query ( Gst.Query ) –
No description available
Returns ( bool ) –
No description available

vfunc_src_query

function vfunc_src_query(encoder: GstAudio.AudioEncoder, 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:

encoder ( GstAudio.AudioEncoder ) –
No description available
query ( Gst.Query ) –
No description available
Returns ( Number ) –
No description available

start

gboolean
start (GstAudioEncoder * enc)

Optional. Called when the element starts processing. Allows opening external resources.

Parameters:

enc
No description available
Returns
No description available

do_start

def do_start (enc):
    #python implementation of the 'start' virtual method

Optional. Called when the element starts processing. Allows opening external resources.

Parameters:

enc ( GstAudio.AudioEncoder ) –
No description available
Returns ( bool ) –
No description available

vfunc_start

function vfunc_start(enc: GstAudio.AudioEncoder): {
    // javascript implementation of the 'start' virtual method
}

Optional. Called when the element starts processing. Allows opening external resources.

Parameters:

enc ( GstAudio.AudioEncoder ) –
No description available
Returns ( Number ) –
No description available

stop

gboolean
stop (GstAudioEncoder * enc)

Optional. Called when the element stops processing. Allows closing external resources.

Parameters:

enc
No description available
Returns
No description available

do_stop

def do_stop (enc):
    #python implementation of the 'stop' virtual method

Optional. Called when the element stops processing. Allows closing external resources.

Parameters:

enc ( GstAudio.AudioEncoder ) –
No description available
Returns ( bool ) –
No description available

vfunc_stop

function vfunc_stop(enc: GstAudio.AudioEncoder): {
    // javascript implementation of the 'stop' virtual method
}

Optional. Called when the element stops processing. Allows closing external resources.

Parameters:

enc ( GstAudio.AudioEncoder ) –
No description available
Returns ( Number ) –
No description available

transform_meta

gboolean
transform_meta (GstAudioEncoder * 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
No description available
outbuf
No description available
meta
No description available
inbuf
No description available
Returns
No description available

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:

enc ( GstAudio.AudioEncoder ) –
No description available
outbuf ( Gst.Buffer ) –
No description available
meta ( Gst.Meta ) –
No description available
inbuf ( Gst.Buffer ) –
No description available
Returns ( bool ) –
No description available

vfunc_transform_meta

function vfunc_transform_meta(enc: GstAudio.AudioEncoder, 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:

enc ( GstAudio.AudioEncoder ) –
No description available
outbuf ( Gst.Buffer ) –
No description available
meta ( Gst.Meta ) –
No description available
inbuf ( Gst.Buffer ) –
No description available
Returns ( Number ) –
No description available

Function Macros

GST_AUDIO_ENCODER_CAST

#define GST_AUDIO_ENCODER_CAST(obj)	((GstAudioEncoder *)(obj))

GST_AUDIO_ENCODER_INPUT_SEGMENT

#define GST_AUDIO_ENCODER_INPUT_SEGMENT(obj)     (GST_AUDIO_ENCODER_CAST (obj)->input_segment)

Gives the input segment of the element.

Parameters:

obj

base parse instance


GST_AUDIO_ENCODER_OUTPUT_SEGMENT

#define GST_AUDIO_ENCODER_OUTPUT_SEGMENT(obj)     (GST_AUDIO_ENCODER_CAST (obj)->output_segment)

Gives the output segment of the element.

Parameters:

obj

base parse instance


GST_AUDIO_ENCODER_SINK_PAD

#define GST_AUDIO_ENCODER_SINK_PAD(obj)	(GST_AUDIO_ENCODER_CAST (obj)->sinkpad)

Gives the pointer to the sink GstPad object of the element.

Parameters:

obj

audio encoder instance


GST_AUDIO_ENCODER_SRC_PAD

#define GST_AUDIO_ENCODER_SRC_PAD(obj)	(GST_AUDIO_ENCODER_CAST (obj)->srcpad)

Gives the pointer to the source GstPad object of the element.

Parameters:

obj

audio encoder instance


GST_AUDIO_ENCODER_STREAM_LOCK

#define GST_AUDIO_ENCODER_STREAM_LOCK(enc)   g_rec_mutex_lock (&GST_AUDIO_ENCODER (enc)->stream_lock)

GST_AUDIO_ENCODER_STREAM_UNLOCK

#define GST_AUDIO_ENCODER_STREAM_UNLOCK(enc) g_rec_mutex_unlock (&GST_AUDIO_ENCODER (enc)->stream_lock)

Constants

GST_AUDIO_ENCODER_SINK_NAME

#define GST_AUDIO_ENCODER_SINK_NAME	"sink"

the name of the templates for the sink pad


GstAudio.AUDIO_ENCODER_SINK_NAME

the name of the templates for the sink pad


GstAudio.AUDIO_ENCODER_SINK_NAME

the name of the templates for the sink pad


GST_AUDIO_ENCODER_SRC_NAME

#define GST_AUDIO_ENCODER_SRC_NAME	        "src"

the name of the templates for the source pad


GstAudio.AUDIO_ENCODER_SRC_NAME

the name of the templates for the source pad


GstAudio.AUDIO_ENCODER_SRC_NAME

the name of the templates for the source pad


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