mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-09-08 13:18:52 +00:00
37c4915109
Since the allocation query caps contains memory size and the pad's caps
contains the display size, an audio encoder or decoder might need to allocate
a different buffer size than the size negotiated in the caps.
This patch splits this logic distinction for audiodecoder and audioencoder.
Thus the user, if needs a different allocation caps, should set it through
gst_audio_{encoder,decoder}_set_allocation_cap() before calling the negotiate()
vmethod. Otherwise the allocation_caps will be the same as the caps in the
src pad.
https://bugzilla.gnome.org/show_bug.cgi?id=764421
408 lines
17 KiB
C
408 lines
17 KiB
C
/* GStreamer
|
|
* Copyright (C) 2009 Igalia S.L.
|
|
* Author: Iago Toral Quiroga <itoral@igalia.com>
|
|
* Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
|
|
* Copyright (C) 2011 Nokia Corporation. All rights reserved.
|
|
* Contact: Stefan Kost <stefan.kost@nokia.com>
|
|
*
|
|
* This library is free software; you can redistribute it and/or
|
|
* modify it under the terms of the GNU Library General Public
|
|
* License as published by the Free Software Foundation; either
|
|
* version 2 of the License, or (at your option) any later version.
|
|
*
|
|
* This library is distributed in the hope that it will be useful,
|
|
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
* Library General Public License for more details.
|
|
*
|
|
* You should have received a copy of the GNU Library General Public
|
|
* License along with this library; if not, write to the
|
|
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
|
|
* Boston, MA 02110-1301, USA.
|
|
*/
|
|
|
|
#ifndef __GST_AUDIO_AUDIO_H__
|
|
#include <gst/audio/audio.h>
|
|
#endif
|
|
|
|
#ifndef _GST_AUDIO_DECODER_H_
|
|
#define _GST_AUDIO_DECODER_H_
|
|
|
|
#include <gst/gst.h>
|
|
#include <gst/base/gstadapter.h>
|
|
|
|
G_BEGIN_DECLS
|
|
|
|
#define GST_TYPE_AUDIO_DECODER \
|
|
(gst_audio_decoder_get_type())
|
|
#define GST_AUDIO_DECODER(obj) \
|
|
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_DECODER,GstAudioDecoder))
|
|
#define GST_AUDIO_DECODER_CLASS(klass) \
|
|
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_DECODER,GstAudioDecoderClass))
|
|
#define GST_AUDIO_DECODER_GET_CLASS(obj) \
|
|
(G_TYPE_INSTANCE_GET_CLASS((obj),GST_TYPE_AUDIO_DECODER,GstAudioDecoderClass))
|
|
#define GST_IS_AUDIO_DECODER(obj) \
|
|
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_DECODER))
|
|
#define GST_IS_AUDIO_DECODER_CLASS(obj) \
|
|
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_DECODER))
|
|
#define GST_AUDIO_DECODER_CAST(obj) \
|
|
((GstAudioDecoder *)(obj))
|
|
|
|
/**
|
|
* GST_AUDIO_DECODER_SINK_NAME:
|
|
*
|
|
* The name of the templates for the sink pad.
|
|
*/
|
|
#define GST_AUDIO_DECODER_SINK_NAME "sink"
|
|
/**
|
|
* GST_AUDIO_DECODER_SRC_NAME:
|
|
*
|
|
* The name of the templates for the source pad.
|
|
*/
|
|
#define GST_AUDIO_DECODER_SRC_NAME "src"
|
|
|
|
/**
|
|
* GST_AUDIO_DECODER_SRC_PAD:
|
|
* @obj: base audio codec instance
|
|
*
|
|
* Gives the pointer to the source #GstPad object of the element.
|
|
*/
|
|
#define GST_AUDIO_DECODER_SRC_PAD(obj) (((GstAudioDecoder *) (obj))->srcpad)
|
|
|
|
/**
|
|
* GST_AUDIO_DECODER_SINK_PAD:
|
|
* @obj: base audio codec instance
|
|
*
|
|
* Gives the pointer to the sink #GstPad object of the element.
|
|
*/
|
|
#define GST_AUDIO_DECODER_SINK_PAD(obj) (((GstAudioDecoder *) (obj))->sinkpad)
|
|
|
|
#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)
|
|
|
|
/**
|
|
* GST_AUDIO_DECODER_INPUT_SEGMENT:
|
|
* @obj: audio decoder instance
|
|
*
|
|
* Gives the input segment of the element.
|
|
*/
|
|
#define GST_AUDIO_DECODER_INPUT_SEGMENT(obj) (GST_AUDIO_DECODER_CAST (obj)->input_segment)
|
|
|
|
/**
|
|
* GST_AUDIO_DECODER_OUTPUT_SEGMENT:
|
|
* @obj: audio decoder instance
|
|
*
|
|
* Gives the output segment of the element.
|
|
*/
|
|
#define GST_AUDIO_DECODER_OUTPUT_SEGMENT(obj) (GST_AUDIO_DECODER_CAST (obj)->output_segment)
|
|
|
|
typedef struct _GstAudioDecoder GstAudioDecoder;
|
|
typedef struct _GstAudioDecoderClass GstAudioDecoderClass;
|
|
|
|
typedef struct _GstAudioDecoderPrivate GstAudioDecoderPrivate;
|
|
|
|
/* do not use this one, use macro below */
|
|
GstFlowReturn _gst_audio_decoder_error (GstAudioDecoder *dec, gint weight,
|
|
GQuark domain, gint code,
|
|
gchar *txt, gchar *debug,
|
|
const gchar *file, const gchar *function,
|
|
gint line);
|
|
|
|
/**
|
|
* GST_AUDIO_DECODER_ERROR:
|
|
* @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
|
|
*
|
|
* 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).
|
|
*/
|
|
#define GST_AUDIO_DECODER_ERROR(el, weight, domain, code, text, debug, ret) \
|
|
G_STMT_START { \
|
|
gchar *__txt = _gst_element_error_printf text; \
|
|
gchar *__dbg = _gst_element_error_printf debug; \
|
|
GstAudioDecoder *__dec = GST_AUDIO_DECODER (el); \
|
|
ret = _gst_audio_decoder_error (__dec, weight, GST_ ## domain ## _ERROR, \
|
|
GST_ ## domain ## _ERROR_ ## code, __txt, __dbg, __FILE__, \
|
|
GST_FUNCTION, __LINE__); \
|
|
} G_STMT_END
|
|
|
|
|
|
/**
|
|
* GST_AUDIO_DECODER_MAX_ERRORS:
|
|
*
|
|
* Default maximum number of errors tolerated before signaling error.
|
|
*/
|
|
#define GST_AUDIO_DECODER_MAX_ERRORS 10
|
|
|
|
/**
|
|
* GstAudioDecoder:
|
|
*
|
|
* The opaque #GstAudioDecoder data structure.
|
|
*/
|
|
struct _GstAudioDecoder
|
|
{
|
|
GstElement element;
|
|
|
|
/*< protected >*/
|
|
/* source and sink pads */
|
|
GstPad *sinkpad;
|
|
GstPad *srcpad;
|
|
|
|
/* protects all data processing, i.e. is locked
|
|
* in the chain function, finish_frame and when
|
|
* processing serialized events */
|
|
GRecMutex stream_lock;
|
|
|
|
/* MT-protected (with STREAM_LOCK) */
|
|
GstSegment input_segment;
|
|
GstSegment output_segment;
|
|
|
|
/*< private >*/
|
|
GstAudioDecoderPrivate *priv;
|
|
|
|
gpointer _gst_reserved[GST_PADDING_LARGE];
|
|
};
|
|
|
|
/**
|
|
* GstAudioDecoderClass:
|
|
* @element_class: The parent class structure
|
|
* @start: Optional.
|
|
* Called when the element starts processing.
|
|
* Allows opening external resources.
|
|
* @stop: Optional.
|
|
* Called when the element stops processing.
|
|
* Allows closing external resources.
|
|
* @set_format: Notifies subclass of incoming data format (caps).
|
|
* @parse: 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.
|
|
* @handle_frame: Provides input data (or NULL to clear any remaining data)
|
|
* to subclass. Input data ref management is performed by
|
|
* base class, subclass should not care or intervene,
|
|
* and input data is only valid until next call to base class,
|
|
* most notably a call to gst_audio_decoder_finish_frame().
|
|
* @flush: Optional.
|
|
* Instructs subclass to clear any codec caches and discard
|
|
* any pending samples and not yet returned decoded data.
|
|
* @hard indicates whether a FLUSH is being processed,
|
|
* or otherwise a DISCONT (or conceptually similar).
|
|
* @sink_event: Optional.
|
|
* Event handler on the sink pad. Subclasses should chain up to
|
|
* the parent implementation to invoke the default handler.
|
|
* @src_event: Optional.
|
|
* Event handler on the src pad. Subclasses should chain up to
|
|
* the parent implementation to invoke the default handler.
|
|
* @pre_push: 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.
|
|
* @open: Optional.
|
|
* Called when the element changes to GST_STATE_READY.
|
|
* Allows opening external resources.
|
|
* @close: Optional.
|
|
* Called when the element changes to GST_STATE_NULL.
|
|
* Allows closing external resources.
|
|
* @negotiate: Optional.
|
|
* Negotiate with downstream and configure buffer pools, etc.
|
|
* Subclasses should chain up to the parent implementation to
|
|
* invoke the default handler.
|
|
* @decide_allocation: 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.
|
|
* @propose_allocation: Optional.
|
|
* Propose buffer allocation parameters for upstream elements.
|
|
* Subclasses should chain up to the parent implementation to
|
|
* invoke the default handler.
|
|
* @sink_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
|
|
* @src_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
|
|
* @getcaps: Optional.
|
|
* Allows for a custom sink getcaps implementation.
|
|
* If not implemented,
|
|
* default returns gst_audio_decoder_proxy_getcaps
|
|
* applied to sink template caps.
|
|
* @transform_meta: 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
|
|
*
|
|
* 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.
|
|
*/
|
|
struct _GstAudioDecoderClass
|
|
{
|
|
GstElementClass element_class;
|
|
|
|
/*< public >*/
|
|
/* 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);
|
|
|
|
/*< private >*/
|
|
gpointer _gst_reserved[GST_PADDING_LARGE - 4];
|
|
};
|
|
|
|
GType gst_audio_decoder_get_type (void);
|
|
|
|
gboolean gst_audio_decoder_set_output_format (GstAudioDecoder * dec,
|
|
const GstAudioInfo * info);
|
|
|
|
GstCaps * gst_audio_decoder_proxy_getcaps (GstAudioDecoder * decoder,
|
|
GstCaps * caps,
|
|
GstCaps * filter);
|
|
gboolean gst_audio_decoder_negotiate (GstAudioDecoder * dec);
|
|
|
|
GstFlowReturn gst_audio_decoder_finish_frame (GstAudioDecoder * dec,
|
|
GstBuffer * buf, gint frames);
|
|
|
|
GstBuffer * gst_audio_decoder_allocate_output_buffer (GstAudioDecoder * dec,
|
|
gsize size);
|
|
|
|
/* context parameters */
|
|
GstAudioInfo * gst_audio_decoder_get_audio_info (GstAudioDecoder * dec);
|
|
|
|
void gst_audio_decoder_set_plc_aware (GstAudioDecoder * dec,
|
|
gboolean plc);
|
|
|
|
gint gst_audio_decoder_get_plc_aware (GstAudioDecoder * dec);
|
|
|
|
void gst_audio_decoder_set_estimate_rate (GstAudioDecoder * dec,
|
|
gboolean enabled);
|
|
|
|
gint gst_audio_decoder_get_estimate_rate (GstAudioDecoder * dec);
|
|
|
|
gint gst_audio_decoder_get_delay (GstAudioDecoder * dec);
|
|
|
|
void gst_audio_decoder_set_max_errors (GstAudioDecoder * dec,
|
|
gint num);
|
|
|
|
gint gst_audio_decoder_get_max_errors (GstAudioDecoder * dec);
|
|
|
|
void gst_audio_decoder_set_latency (GstAudioDecoder * dec,
|
|
GstClockTime min,
|
|
GstClockTime max);
|
|
|
|
void gst_audio_decoder_get_latency (GstAudioDecoder * dec,
|
|
GstClockTime * min,
|
|
GstClockTime * max);
|
|
|
|
void gst_audio_decoder_get_parse_state (GstAudioDecoder * dec,
|
|
gboolean * sync,
|
|
gboolean * eos);
|
|
|
|
void gst_audio_decoder_set_allocation_caps (GstAudioDecoder * dec,
|
|
GstCaps * allocation_caps);
|
|
|
|
/* object properties */
|
|
void gst_audio_decoder_set_plc (GstAudioDecoder * dec,
|
|
gboolean enabled);
|
|
|
|
gboolean gst_audio_decoder_get_plc (GstAudioDecoder * dec);
|
|
|
|
void gst_audio_decoder_set_min_latency (GstAudioDecoder * dec,
|
|
GstClockTime num);
|
|
|
|
GstClockTime gst_audio_decoder_get_min_latency (GstAudioDecoder * dec);
|
|
|
|
void gst_audio_decoder_set_tolerance (GstAudioDecoder * dec,
|
|
GstClockTime tolerance);
|
|
|
|
GstClockTime gst_audio_decoder_get_tolerance (GstAudioDecoder * dec);
|
|
|
|
void gst_audio_decoder_set_drainable (GstAudioDecoder * dec,
|
|
gboolean enabled);
|
|
|
|
gboolean gst_audio_decoder_get_drainable (GstAudioDecoder * dec);
|
|
|
|
void gst_audio_decoder_set_needs_format (GstAudioDecoder * dec,
|
|
gboolean enabled);
|
|
|
|
gboolean gst_audio_decoder_get_needs_format (GstAudioDecoder * dec);
|
|
|
|
void gst_audio_decoder_get_allocator (GstAudioDecoder * dec,
|
|
GstAllocator ** allocator,
|
|
GstAllocationParams * params);
|
|
|
|
void gst_audio_decoder_merge_tags (GstAudioDecoder * dec,
|
|
const GstTagList * tags, GstTagMergeMode mode);
|
|
|
|
void gst_audio_decoder_set_use_default_pad_acceptcaps (GstAudioDecoder * decoder,
|
|
gboolean use);
|
|
|
|
#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
|
|
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstAudioDecoder, gst_object_unref)
|
|
#endif
|
|
|
|
G_END_DECLS
|
|
|
|
#endif /* _GST_AUDIO_DECODER_H_ */
|