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371e3e460a
We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
444 lines
18 KiB
C
444 lines
18 KiB
C
/* GStreamer
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* Copyright (C) 2009 Igalia S.L.
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* Author: Iago Toral Quiroga <itoral@igalia.com>
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* Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
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* Copyright (C) 2011 Nokia Corporation. All rights reserved.
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* Contact: Stefan Kost <stefan.kost@nokia.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifndef __GST_AUDIO_AUDIO_H__
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#include <gst/audio/audio.h>
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#endif
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#ifndef _GST_AUDIO_DECODER_H_
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#define _GST_AUDIO_DECODER_H_
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#include <gst/gst.h>
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#include <gst/base/gstadapter.h>
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G_BEGIN_DECLS
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#define GST_TYPE_AUDIO_DECODER \
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(gst_audio_decoder_get_type())
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#define GST_AUDIO_DECODER(obj) \
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(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_DECODER,GstAudioDecoder))
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#define GST_AUDIO_DECODER_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_DECODER,GstAudioDecoderClass))
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#define GST_AUDIO_DECODER_GET_CLASS(obj) \
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(G_TYPE_INSTANCE_GET_CLASS((obj),GST_TYPE_AUDIO_DECODER,GstAudioDecoderClass))
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#define GST_IS_AUDIO_DECODER(obj) \
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(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_DECODER))
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#define GST_IS_AUDIO_DECODER_CLASS(obj) \
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(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_DECODER))
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#define GST_AUDIO_DECODER_CAST(obj) \
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((GstAudioDecoder *)(obj))
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/**
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* GST_AUDIO_DECODER_SINK_NAME:
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*
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* The name of the templates for the sink pad.
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*/
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#define GST_AUDIO_DECODER_SINK_NAME "sink"
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/**
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* GST_AUDIO_DECODER_SRC_NAME:
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*
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* The name of the templates for the source pad.
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*/
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#define GST_AUDIO_DECODER_SRC_NAME "src"
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/**
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* GST_AUDIO_DECODER_SRC_PAD:
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* @obj: base audio codec instance
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*
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* Gives the pointer to the source #GstPad object of the element.
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*/
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#define GST_AUDIO_DECODER_SRC_PAD(obj) (((GstAudioDecoder *) (obj))->srcpad)
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/**
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* GST_AUDIO_DECODER_SINK_PAD:
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* @obj: base audio codec instance
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*
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* Gives the pointer to the sink #GstPad object of the element.
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*/
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#define GST_AUDIO_DECODER_SINK_PAD(obj) (((GstAudioDecoder *) (obj))->sinkpad)
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#define GST_AUDIO_DECODER_STREAM_LOCK(dec) g_rec_mutex_lock (&GST_AUDIO_DECODER (dec)->stream_lock)
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#define GST_AUDIO_DECODER_STREAM_UNLOCK(dec) g_rec_mutex_unlock (&GST_AUDIO_DECODER (dec)->stream_lock)
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/**
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* GST_AUDIO_DECODER_INPUT_SEGMENT:
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* @obj: audio decoder instance
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*
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* Gives the input segment of the element.
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*/
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#define GST_AUDIO_DECODER_INPUT_SEGMENT(obj) (GST_AUDIO_DECODER_CAST (obj)->input_segment)
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/**
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* GST_AUDIO_DECODER_OUTPUT_SEGMENT:
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* @obj: audio decoder instance
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*
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* Gives the output segment of the element.
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*/
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#define GST_AUDIO_DECODER_OUTPUT_SEGMENT(obj) (GST_AUDIO_DECODER_CAST (obj)->output_segment)
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typedef struct _GstAudioDecoder GstAudioDecoder;
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typedef struct _GstAudioDecoderClass GstAudioDecoderClass;
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typedef struct _GstAudioDecoderPrivate GstAudioDecoderPrivate;
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/* do not use this one, use macro below */
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GST_AUDIO_API
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GstFlowReturn _gst_audio_decoder_error (GstAudioDecoder *dec, gint weight,
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GQuark domain, gint code,
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gchar *txt, gchar *debug,
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const gchar *file, const gchar *function,
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gint line);
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/**
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* GST_AUDIO_DECODER_ERROR:
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* @el: the base audio decoder element that generates the error
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* @weight: element defined weight of the error, added to error count
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* @domain: like CORE, LIBRARY, RESOURCE or STREAM (see #gstreamer-GstGError)
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* @code: error code defined for that domain (see #gstreamer-GstGError)
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* @text: the message to display (format string and args enclosed in
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* parentheses)
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* @debug: debugging information for the message (format string and args
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* enclosed in parentheses)
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* @ret: variable to receive return value
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*
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* Utility function that audio decoder elements can use in case they encountered
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* a data processing error that may be fatal for the current "data unit" but
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* need not prevent subsequent decoding. Such errors are counted and if there
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* are too many, as configured in the context's max_errors, the pipeline will
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* post an error message and the application will be requested to stop further
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* media processing. Otherwise, it is considered a "glitch" and only a warning
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* is logged. In either case, @ret is set to the proper value to
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* return to upstream/caller (indicating either GST_FLOW_ERROR or GST_FLOW_OK).
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*/
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#define GST_AUDIO_DECODER_ERROR(el, weight, domain, code, text, debug, ret) \
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G_STMT_START { \
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gchar *__txt = _gst_element_error_printf text; \
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gchar *__dbg = _gst_element_error_printf debug; \
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GstAudioDecoder *__dec = GST_AUDIO_DECODER (el); \
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ret = _gst_audio_decoder_error (__dec, weight, GST_ ## domain ## _ERROR, \
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GST_ ## domain ## _ERROR_ ## code, __txt, __dbg, __FILE__, \
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GST_FUNCTION, __LINE__); \
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} G_STMT_END
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/**
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* GST_AUDIO_DECODER_MAX_ERRORS:
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*
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* Default maximum number of errors tolerated before signaling error.
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*/
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#define GST_AUDIO_DECODER_MAX_ERRORS 10
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/**
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* GstAudioDecoder:
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*
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* The opaque #GstAudioDecoder data structure.
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*/
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struct _GstAudioDecoder
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{
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GstElement element;
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/*< protected >*/
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/* source and sink pads */
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GstPad *sinkpad;
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GstPad *srcpad;
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/* protects all data processing, i.e. is locked
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* in the chain function, finish_frame and when
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* processing serialized events */
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GRecMutex stream_lock;
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/* MT-protected (with STREAM_LOCK) */
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GstSegment input_segment;
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GstSegment output_segment;
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/*< private >*/
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GstAudioDecoderPrivate *priv;
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gpointer _gst_reserved[GST_PADDING_LARGE];
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};
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/**
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* GstAudioDecoderClass:
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* @element_class: The parent class structure
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* @start: Optional.
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* Called when the element starts processing.
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* Allows opening external resources.
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* @stop: Optional.
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* Called when the element stops processing.
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* Allows closing external resources.
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* @set_format: Notifies subclass of incoming data format (caps).
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* @parse: Optional.
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* Allows chopping incoming data into manageable units (frames)
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* for subsequent decoding. This division is at subclass
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* discretion and may or may not correspond to 1 (or more)
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* frames as defined by audio format.
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* @handle_frame: Provides input data (or NULL to clear any remaining data)
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* to subclass. Input data ref management is performed by
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* base class, subclass should not care or intervene,
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* and input data is only valid until next call to base class,
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* most notably a call to gst_audio_decoder_finish_frame().
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* @flush: Optional.
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* Instructs subclass to clear any codec caches and discard
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* any pending samples and not yet returned decoded data.
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* @hard indicates whether a FLUSH is being processed,
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* or otherwise a DISCONT (or conceptually similar).
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* @sink_event: Optional.
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* Event handler on the sink pad. Subclasses should chain up to
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* the parent implementation to invoke the default handler.
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* @src_event: Optional.
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* Event handler on the src pad. Subclasses should chain up to
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* the parent implementation to invoke the default handler.
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* @pre_push: Optional.
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* Called just prior to pushing (encoded data) buffer downstream.
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* Subclass has full discretionary access to buffer,
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* and a not OK flow return will abort downstream pushing.
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* @open: Optional.
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* Called when the element changes to GST_STATE_READY.
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* Allows opening external resources.
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* @close: Optional.
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* Called when the element changes to GST_STATE_NULL.
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* Allows closing external resources.
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* @negotiate: Optional.
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* Negotiate with downstream and configure buffer pools, etc.
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* Subclasses should chain up to the parent implementation to
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* invoke the default handler.
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* @decide_allocation: Optional.
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* Setup the allocation parameters for allocating output
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* buffers. The passed in query contains the result of the
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* downstream allocation query.
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* Subclasses should chain up to the parent implementation to
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* invoke the default handler.
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* @propose_allocation: Optional.
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* Propose buffer allocation parameters for upstream elements.
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* Subclasses should chain up to the parent implementation to
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* invoke the default handler.
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* @sink_query: Optional.
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* Query handler on the sink pad. This function should
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* return TRUE if the query could be performed. Subclasses
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* should chain up to the parent implementation to invoke the
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* default handler. Since 1.6
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* @src_query: Optional.
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* Query handler on the source pad. This function should
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* return TRUE if the query could be performed. Subclasses
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* should chain up to the parent implementation to invoke the
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* default handler. Since 1.6
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* @getcaps: Optional.
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* Allows for a custom sink getcaps implementation.
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* If not implemented,
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* default returns gst_audio_decoder_proxy_getcaps
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* applied to sink template caps.
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* @transform_meta: Optional. Transform the metadata on the input buffer to the
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* output buffer. By default this method copies all meta without
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* tags and meta with only the "audio" tag. subclasses can
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* implement this method and return %TRUE if the metadata is to be
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* copied. Since 1.6
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*
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* Subclasses can override any of the available virtual methods or not, as
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* needed. At minimum @handle_frame (and likely @set_format) needs to be
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* overridden.
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*/
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struct _GstAudioDecoderClass
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{
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GstElementClass element_class;
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/*< public >*/
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/* virtual methods for subclasses */
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gboolean (*start) (GstAudioDecoder *dec);
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gboolean (*stop) (GstAudioDecoder *dec);
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gboolean (*set_format) (GstAudioDecoder *dec,
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GstCaps *caps);
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GstFlowReturn (*parse) (GstAudioDecoder *dec,
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GstAdapter *adapter,
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gint *offset, gint *length);
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GstFlowReturn (*handle_frame) (GstAudioDecoder *dec,
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GstBuffer *buffer);
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void (*flush) (GstAudioDecoder *dec, gboolean hard);
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GstFlowReturn (*pre_push) (GstAudioDecoder *dec,
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GstBuffer **buffer);
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gboolean (*sink_event) (GstAudioDecoder *dec,
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GstEvent *event);
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gboolean (*src_event) (GstAudioDecoder *dec,
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GstEvent *event);
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gboolean (*open) (GstAudioDecoder *dec);
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gboolean (*close) (GstAudioDecoder *dec);
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gboolean (*negotiate) (GstAudioDecoder *dec);
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gboolean (*decide_allocation) (GstAudioDecoder *dec, GstQuery *query);
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gboolean (*propose_allocation) (GstAudioDecoder *dec,
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GstQuery * query);
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gboolean (*sink_query) (GstAudioDecoder *dec, GstQuery *query);
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gboolean (*src_query) (GstAudioDecoder *dec, GstQuery *query);
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GstCaps * (*getcaps) (GstAudioDecoder * dec,
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GstCaps * filter);
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gboolean (*transform_meta) (GstAudioDecoder *enc, GstBuffer *outbuf,
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GstMeta *meta, GstBuffer *inbuf);
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/*< private >*/
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gpointer _gst_reserved[GST_PADDING_LARGE - 4];
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};
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GST_AUDIO_API
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GType gst_audio_decoder_get_type (void);
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GST_AUDIO_API
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gboolean gst_audio_decoder_set_output_format (GstAudioDecoder * dec,
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const GstAudioInfo * info);
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GST_AUDIO_API
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GstCaps * gst_audio_decoder_proxy_getcaps (GstAudioDecoder * decoder,
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GstCaps * caps,
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GstCaps * filter);
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GST_AUDIO_API
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gboolean gst_audio_decoder_negotiate (GstAudioDecoder * dec);
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GST_AUDIO_API
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GstFlowReturn gst_audio_decoder_finish_frame (GstAudioDecoder * dec,
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GstBuffer * buf, gint frames);
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GST_AUDIO_API
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GstBuffer * gst_audio_decoder_allocate_output_buffer (GstAudioDecoder * dec,
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gsize size);
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/* context parameters */
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GST_AUDIO_API
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GstAudioInfo * gst_audio_decoder_get_audio_info (GstAudioDecoder * dec);
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GST_AUDIO_API
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void gst_audio_decoder_set_plc_aware (GstAudioDecoder * dec,
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gboolean plc);
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GST_AUDIO_API
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gint gst_audio_decoder_get_plc_aware (GstAudioDecoder * dec);
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GST_AUDIO_API
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void gst_audio_decoder_set_estimate_rate (GstAudioDecoder * dec,
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gboolean enabled);
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GST_AUDIO_API
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gint gst_audio_decoder_get_estimate_rate (GstAudioDecoder * dec);
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GST_AUDIO_API
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gint gst_audio_decoder_get_delay (GstAudioDecoder * dec);
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GST_AUDIO_API
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void gst_audio_decoder_set_max_errors (GstAudioDecoder * dec,
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gint num);
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GST_AUDIO_API
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gint gst_audio_decoder_get_max_errors (GstAudioDecoder * dec);
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GST_AUDIO_API
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void gst_audio_decoder_set_latency (GstAudioDecoder * dec,
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GstClockTime min,
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GstClockTime max);
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GST_AUDIO_API
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void gst_audio_decoder_get_latency (GstAudioDecoder * dec,
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GstClockTime * min,
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GstClockTime * max);
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GST_AUDIO_API
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void gst_audio_decoder_get_parse_state (GstAudioDecoder * dec,
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gboolean * sync,
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gboolean * eos);
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GST_AUDIO_API
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void gst_audio_decoder_set_allocation_caps (GstAudioDecoder * dec,
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GstCaps * allocation_caps);
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/* object properties */
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GST_AUDIO_API
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void gst_audio_decoder_set_plc (GstAudioDecoder * dec,
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gboolean enabled);
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GST_AUDIO_API
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gboolean gst_audio_decoder_get_plc (GstAudioDecoder * dec);
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GST_AUDIO_API
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void gst_audio_decoder_set_min_latency (GstAudioDecoder * dec,
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GstClockTime num);
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GST_AUDIO_API
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GstClockTime gst_audio_decoder_get_min_latency (GstAudioDecoder * dec);
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GST_AUDIO_API
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void gst_audio_decoder_set_tolerance (GstAudioDecoder * dec,
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GstClockTime tolerance);
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GST_AUDIO_API
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GstClockTime gst_audio_decoder_get_tolerance (GstAudioDecoder * dec);
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GST_AUDIO_API
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void gst_audio_decoder_set_drainable (GstAudioDecoder * dec,
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gboolean enabled);
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GST_AUDIO_API
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gboolean gst_audio_decoder_get_drainable (GstAudioDecoder * dec);
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GST_AUDIO_API
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void gst_audio_decoder_set_needs_format (GstAudioDecoder * dec,
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gboolean enabled);
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GST_AUDIO_API
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gboolean gst_audio_decoder_get_needs_format (GstAudioDecoder * dec);
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GST_AUDIO_API
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void gst_audio_decoder_get_allocator (GstAudioDecoder * dec,
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GstAllocator ** allocator,
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GstAllocationParams * params);
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GST_AUDIO_API
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void gst_audio_decoder_merge_tags (GstAudioDecoder * dec,
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const GstTagList * tags, GstTagMergeMode mode);
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GST_AUDIO_API
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void gst_audio_decoder_set_use_default_pad_acceptcaps (GstAudioDecoder * decoder,
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gboolean use);
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#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
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G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstAudioDecoder, gst_object_unref)
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#endif
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G_END_DECLS
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#endif /* _GST_AUDIO_DECODER_H_ */
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