gstreamer/omx/gstbaseaudiodecoder.h

270 lines
10 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef _GST_BASE_AUDIO_DECODER_H_
#define _GST_BASE_AUDIO_DECODER_H_
#ifndef GST_USE_UNSTABLE_API
#warning "GstBaseAudioDecoder is unstable API and may change in future."
#warning "You can define GST_USE_UNSTABLE_API to avoid this warning."
#endif
#include <gst/gst.h>
#include <gst/base/gstadapter.h>
#include "gstbaseaudioutils.h"
G_BEGIN_DECLS
#define GST_TYPE_BASE_AUDIO_DECODER \
(gst_base_audio_decoder_get_type())
#define GST_BASE_AUDIO_DECODER(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_AUDIO_DECODER,GstBaseAudioDecoder))
#define GST_BASE_AUDIO_DECODER_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_AUDIO_DECODER,GstBaseAudioDecoderClass))
#define GST_BASE_AUDIO_DECODER_GET_CLASS(obj) \
(G_TYPE_INSTANCE_GET_CLASS((obj),GST_TYPE_BASE_AUDIO_DECODER,GstBaseAudioDecoderClass))
#define GST_IS_BASE_AUDIO_DECODER(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_AUDIO_DECODER))
#define GST_IS_BASE_AUDIO_DECODER_CLASS(obj) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_AUDIO_DECODER))
/**
* GST_BASE_AUDIO_DECODER_SINK_NAME:
*
* The name of the templates for the sink pad.
*/
#define GST_BASE_AUDIO_DECODER_SINK_NAME "sink"
/**
* GST_BASE_AUDIO_DECODER_SRC_NAME:
*
* The name of the templates for the source pad.
*/
#define GST_BASE_AUDIO_DECODER_SRC_NAME "src"
/**
* GST_BASE_AUDIO_DECODER_SRC_PAD:
* @obj: base audio codec instance
*
* Gives the pointer to the source #GstPad object of the element.
*/
#define GST_BASE_AUDIO_DECODER_SRC_PAD(obj) (((GstBaseAudioDecoder *) (obj))->srcpad)
/**
* GST_BASE_AUDIO_DECODER_SINK_PAD:
* @obj: base audio codec instance
*
* Gives the pointer to the sink #GstPad object of the element.
*/
#define GST_BASE_AUDIO_DECODER_SINK_PAD(obj) (((GstBaseAudioDecoder *) (obj))->sinkpad)
#define GST_BASE_AUDIO_DECODER_STREAM_LOCK(dec) g_static_rec_mutex_lock (&GST_BASE_AUDIO_DECODER (dec)->stream_lock)
#define GST_BASE_AUDIO_DECODER_STREAM_UNLOCK(dec) g_static_rec_mutex_unlock (&GST_BASE_AUDIO_DECODER (dec)->stream_lock)
typedef struct _GstBaseAudioDecoder GstBaseAudioDecoder;
typedef struct _GstBaseAudioDecoderClass GstBaseAudioDecoderClass;
typedef struct _GstBaseAudioDecoderPrivate GstBaseAudioDecoderPrivate;
typedef struct _GstBaseAudioDecoderContext GstBaseAudioDecoderContext;
/* do not use this one, use macro below */
GstFlowReturn _gst_base_audio_decoder_error (GstBaseAudioDecoder *dec, gint weight,
GQuark domain, gint code,
gchar *txt, gchar *debug,
const gchar *file, const gchar *function,
gint line);
/**
* GST_BASE_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_BASE_AUDIO_DECODER_ERROR(el, w, domain, code, text, debug, ret) \
G_STMT_START { \
gchar *__txt = _gst_element_error_printf text; \
gchar *__dbg = _gst_element_error_printf debug; \
GstBaseAudioDecoder *dec = GST_BASE_AUDIO_DECODER (el); \
ret = _gst_base_audio_decoder_error (dec, w, GST_ ## domain ## _ERROR, \
GST_ ## domain ## _ERROR_ ## code, __txt, __dbg, __FILE__, \
GST_FUNCTION, __LINE__); \
} G_STMT_END
/**
* GstBaseAudioDecoderContext:
* @state: a #GstAudioState describing input audio format
* @eos: no (immediate) subsequent data in stream
* @sync: stream parsing in sync
* @delay: number of frames pending decoding (typically at least 1 for current)
* @do_plc: whether subclass is prepared to handle (packet) loss concealment
* @min_latency: min latency of element
* @max_latency: max latency of element
* @lookahead: decoder lookahead (in units of input rate samples)
*
* Transparent #GstBaseAudioEncoderContext data structure.
*/
struct _GstBaseAudioDecoderContext {
/* input */
/* (output) audio format */
GstAudioState state;
/* parsing state */
gboolean eos;
gboolean sync;
/* misc */
gint delay;
/* output */
gboolean do_plc;
gboolean do_byte_time;
gint max_errors;
/* MT-protected (with LOCK) */
GstClockTime min_latency;
GstClockTime max_latency;
};
/**
* GstBaseAudioDecoder:
*
* The opaque #GstBaseAudioDecoder data structure.
*/
struct _GstBaseAudioDecoder
{
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 */
GStaticRecMutex stream_lock;
/* MT-protected (with STREAM_LOCK) */
GstSegment segment;
GstBaseAudioDecoderContext *ctx;
/* properties */
GstClockTime latency;
GstClockTime tolerance;
gboolean plc;
/*< private >*/
GstBaseAudioDecoderPrivate *priv;
gpointer _gst_reserved[GST_PADDING_LARGE];
};
/**
* GstBaseAudioDecoderClass:
* @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.
* @flush: Optional.
* Instructs subclass to clear any codec caches and discard
* any pending samples and not yet returned encoded data.
* @hard indicates whether a FLUSH is being processed,
* or otherwise a DISCONT (or conceptually similar).
* @event: Optional.
* Event handler on the sink pad. This function should return
* TRUE if the event was handled and should be discarded
* (i.e. not unref'ed).
* @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.
*
* 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 _GstBaseAudioDecoderClass
{
GstElementClass parent_class;
/*< public >*/
/* virtual methods for subclasses */
gboolean (*start) (GstBaseAudioDecoder *dec);
gboolean (*stop) (GstBaseAudioDecoder *dec);
gboolean (*set_format) (GstBaseAudioDecoder *dec,
GstCaps *caps);
GstFlowReturn (*parse) (GstBaseAudioDecoder *dec,
GstAdapter *adapter,
gint *offset, gint *length);
GstFlowReturn (*handle_frame) (GstBaseAudioDecoder *dec,
GstBuffer *buffer);
void (*flush) (GstBaseAudioDecoder *dec, gboolean hard);
GstFlowReturn (*pre_push) (GstBaseAudioDecoder *dec,
GstBuffer **buffer);
gboolean (*event) (GstBaseAudioDecoder *dec,
GstEvent *event);
/*< private >*/
gpointer _gst_reserved[GST_PADDING_LARGE];
};
GstFlowReturn gst_base_audio_decoder_finish_frame (GstBaseAudioDecoder * dec,
GstBuffer * buf, gint frames);
GType gst_base_audio_decoder_get_type (void);
G_END_DECLS
#endif