/* 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 */
GST_AUDIO_API
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];
};
GST_AUDIO_API
GType gst_audio_decoder_get_type (void);
GST_AUDIO_API
gboolean gst_audio_decoder_set_output_format (GstAudioDecoder * dec,
const GstAudioInfo * info);
GST_AUDIO_API
GstCaps * gst_audio_decoder_proxy_getcaps (GstAudioDecoder * decoder,
GstCaps * caps,
GstCaps * filter);
GST_AUDIO_API
gboolean gst_audio_decoder_negotiate (GstAudioDecoder * dec);
GST_AUDIO_API
GstFlowReturn gst_audio_decoder_finish_frame (GstAudioDecoder * dec,
GstBuffer * buf, gint frames);
GST_AUDIO_API
GstBuffer * gst_audio_decoder_allocate_output_buffer (GstAudioDecoder * dec,
gsize size);
/* context parameters */
GST_AUDIO_API
GstAudioInfo * gst_audio_decoder_get_audio_info (GstAudioDecoder * dec);
GST_AUDIO_API
void gst_audio_decoder_set_plc_aware (GstAudioDecoder * dec,
gboolean plc);
GST_AUDIO_API
gint gst_audio_decoder_get_plc_aware (GstAudioDecoder * dec);
GST_AUDIO_API
void gst_audio_decoder_set_estimate_rate (GstAudioDecoder * dec,
gboolean enabled);
GST_AUDIO_API
gint gst_audio_decoder_get_estimate_rate (GstAudioDecoder * dec);
GST_AUDIO_API
gint gst_audio_decoder_get_delay (GstAudioDecoder * dec);
GST_AUDIO_API
void gst_audio_decoder_set_max_errors (GstAudioDecoder * dec,
gint num);
GST_AUDIO_API
gint gst_audio_decoder_get_max_errors (GstAudioDecoder * dec);
GST_AUDIO_API
void gst_audio_decoder_set_latency (GstAudioDecoder * dec,
GstClockTime min,
GstClockTime max);
GST_AUDIO_API
void gst_audio_decoder_get_latency (GstAudioDecoder * dec,
GstClockTime * min,
GstClockTime * max);
GST_AUDIO_API
void gst_audio_decoder_get_parse_state (GstAudioDecoder * dec,
gboolean * sync,
gboolean * eos);
GST_AUDIO_API
void gst_audio_decoder_set_allocation_caps (GstAudioDecoder * dec,
GstCaps * allocation_caps);
/* object properties */
GST_AUDIO_API
void gst_audio_decoder_set_plc (GstAudioDecoder * dec,
gboolean enabled);
GST_AUDIO_API
gboolean gst_audio_decoder_get_plc (GstAudioDecoder * dec);
GST_AUDIO_API
void gst_audio_decoder_set_min_latency (GstAudioDecoder * dec,
GstClockTime num);
GST_AUDIO_API
GstClockTime gst_audio_decoder_get_min_latency (GstAudioDecoder * dec);
GST_AUDIO_API
void gst_audio_decoder_set_tolerance (GstAudioDecoder * dec,
GstClockTime tolerance);
GST_AUDIO_API
GstClockTime gst_audio_decoder_get_tolerance (GstAudioDecoder * dec);
GST_AUDIO_API
void gst_audio_decoder_set_drainable (GstAudioDecoder * dec,
gboolean enabled);
GST_AUDIO_API
gboolean gst_audio_decoder_get_drainable (GstAudioDecoder * dec);
GST_AUDIO_API
void gst_audio_decoder_set_needs_format (GstAudioDecoder * dec,
gboolean enabled);
GST_AUDIO_API
gboolean gst_audio_decoder_get_needs_format (GstAudioDecoder * dec);
GST_AUDIO_API
void gst_audio_decoder_get_allocator (GstAudioDecoder * dec,
GstAllocator ** allocator,
GstAllocationParams * params);
GST_AUDIO_API
void gst_audio_decoder_merge_tags (GstAudioDecoder * dec,
const GstTagList * tags, GstTagMergeMode mode);
GST_AUDIO_API
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_ */