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352 lines
14 KiB
C
352 lines
14 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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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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/audio/audio.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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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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* @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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* @propose_allocation: Optional.
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* Propose buffer allocation parameters for upstream elements.
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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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/*< private >*/
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gpointer _gst_reserved[GST_PADDING_LARGE-2];
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};
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GType gst_audio_decoder_get_type (void);
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gboolean gst_audio_decoder_set_output_format (GstAudioDecoder * dec,
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const GstAudioInfo * info);
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gboolean gst_audio_decoder_negotiate (GstAudioDecoder * dec);
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GstFlowReturn gst_audio_decoder_finish_frame (GstAudioDecoder * dec,
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GstBuffer * buf, gint frames);
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GstBuffer * gst_audio_decoder_allocate_output_buffer (GstAudioDecoder * decoder,
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gsize size);
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/* context parameters */
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GstAudioInfo * gst_audio_decoder_get_audio_info (GstAudioDecoder * dec);
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void gst_audio_decoder_set_plc_aware (GstAudioDecoder * dec,
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gboolean plc);
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gint gst_audio_decoder_get_plc_aware (GstAudioDecoder * dec);
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void gst_audio_decoder_set_estimate_rate (GstAudioDecoder * dec,
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gboolean enabled);
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gint gst_audio_decoder_get_estimate_rate (GstAudioDecoder * dec);
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gint gst_audio_decoder_get_delay (GstAudioDecoder * dec);
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void gst_audio_decoder_set_max_errors (GstAudioDecoder * dec,
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gint num);
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gint gst_audio_decoder_get_max_errors (GstAudioDecoder * dec);
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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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void gst_audio_decoder_get_latency (GstAudioDecoder * dec,
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GstClockTime * min,
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GstClockTime * max);
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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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/* object properties */
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void gst_audio_decoder_set_plc (GstAudioDecoder * dec,
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gboolean enabled);
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gboolean gst_audio_decoder_get_plc (GstAudioDecoder * dec);
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void gst_audio_decoder_set_min_latency (GstAudioDecoder * dec,
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gint64 num);
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gint64 gst_audio_decoder_get_min_latency (GstAudioDecoder * dec);
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void gst_audio_decoder_set_tolerance (GstAudioDecoder * dec,
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gint64 tolerance);
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gint64 gst_audio_decoder_get_tolerance (GstAudioDecoder * dec);
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void gst_audio_decoder_set_drainable (GstAudioDecoder * dec,
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gboolean enabled);
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gboolean gst_audio_decoder_get_drainable (GstAudioDecoder * dec);
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void gst_audio_decoder_set_needs_format (GstAudioDecoder * dec,
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gboolean enabled);
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gboolean gst_audio_decoder_get_needs_format (GstAudioDecoder * dec);
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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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G_END_DECLS
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#endif /* _GST_AUDIO_DECODER_H_ */
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