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/* GStreamer
* 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_AUDIO_ENCODER_H__
#define __GST_AUDIO_ENCODER_H__
#include <gst/gst.h>
#include <gst/audio/audio.h>
G_BEGIN_DECLS
#define GST_TYPE_AUDIO_ENCODER (gst_audio_encoder_get_type())
#define GST_AUDIO_ENCODER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_ENCODER,GstAudioEncoder))
#define GST_AUDIO_ENCODER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_ENCODER,GstAudioEncoderClass))
#define GST_AUDIO_ENCODER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj),GST_TYPE_AUDIO_ENCODER,GstAudioEncoderClass))
#define GST_IS_AUDIO_ENCODER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_ENCODER))
#define GST_IS_AUDIO_ENCODER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_ENCODER))
#define GST_AUDIO_ENCODER_CAST(obj) ((GstAudioEncoder *)(obj))
/**
* GST_AUDIO_ENCODER_SINK_NAME:
*
* the name of the templates for the sink pad
*
* Since: 0.10.36
*/
#define GST_AUDIO_ENCODER_SINK_NAME "sink"
/**
* GST_AUDIO_ENCODER_SRC_NAME:
*
* the name of the templates for the source pad
*
* Since: 0.10.36
*/
#define GST_AUDIO_ENCODER_SRC_NAME "src"
/**
* GST_AUDIO_ENCODER_SRC_PAD:
* @obj: base parse instance
*
* Gives the pointer to the source #GstPad object of the element.
*
* Since: 0.10.36
*/
#define GST_AUDIO_ENCODER_SRC_PAD(obj) (GST_AUDIO_ENCODER_CAST (obj)->srcpad)
/**
* GST_AUDIO_ENCODER_SINK_PAD:
* @obj: base parse instance
*
* Gives the pointer to the sink #GstPad object of the element.
*
* Since: 0.10.36
*/
#define GST_AUDIO_ENCODER_SINK_PAD(obj) (GST_AUDIO_ENCODER_CAST (obj)->sinkpad)
/**
* GST_AUDIO_ENCODER_SEGMENT:
* @obj: base parse instance
*
* Gives the segment of the element.
*
* Since: 0.10.36
*/
#define GST_AUDIO_ENCODER_SEGMENT(obj) (GST_AUDIO_ENCODER_CAST (obj)->segment)
#define GST_AUDIO_ENCODER_STREAM_LOCK(enc) g_static_rec_mutex_lock (&GST_AUDIO_ENCODER (enc)->stream_lock)
#define GST_AUDIO_ENCODER_STREAM_UNLOCK(enc) g_static_rec_mutex_unlock (&GST_AUDIO_ENCODER (enc)->stream_lock)
typedef struct _GstAudioEncoder GstAudioEncoder;
typedef struct _GstAudioEncoderClass GstAudioEncoderClass;
typedef struct _GstAudioEncoderPrivate GstAudioEncoderPrivate;
/**
* GstAudioEncoder:
*
* The opaque #GstAudioEncoder data structure.
*
* Since: 0.10.36
*/
struct _GstAudioEncoder {
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;
/*< private >*/
GstAudioEncoderPrivate *priv;
gpointer _gst_reserved[GST_PADDING_LARGE];
};
/**
* GstAudioEncoderClass:
* @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.
* GstAudioInfo contains the format according to provided caps.
* @handle_frame: Provides input samples (or NULL to clear any remaining data)
* according to directions as configured by the subclass
* using the API. 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_encoder_finish_frame().
* @flush: Optional.
* Instructs subclass to clear any codec caches and discard
* any pending samples and not yet returned encoded data.
* @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.
* @getcaps: Optional.
* Allows for a custom sink getcaps implementation (e.g.
* for multichannel input specification). If not implemented,
* default returns gst_audio_encoder_proxy_getcaps
* applied to sink template caps.
* @open: Optional.
* Called when the element changes to GST_STATE_READY.
* Allows opening external resources. Since: 0.10.37.
* @close: Optional.
* Called when the element changes to GST_STATE_NULL.
* Allows closing external resources. Since: 0.10.37.
*
* Subclasses can override any of the available virtual methods or not, as
* needed. At minimum @set_format and @handle_frame needs to be overridden.
*
* Since: 0.10.36
*/
struct _GstAudioEncoderClass {
GstElementClass element_class;
/*< public >*/
/* virtual methods for subclasses */
gboolean (*start) (GstAudioEncoder *enc);
gboolean (*stop) (GstAudioEncoder *enc);
gboolean (*set_format) (GstAudioEncoder *enc,
GstAudioInfo *info);
GstFlowReturn (*handle_frame) (GstAudioEncoder *enc,
GstBuffer *buffer);
void (*flush) (GstAudioEncoder *enc);
GstFlowReturn (*pre_push) (GstAudioEncoder *enc,
GstBuffer **buffer);
gboolean (*event) (GstAudioEncoder *enc,
GstEvent *event);
GstCaps * (*getcaps) (GstAudioEncoder *enc);
gboolean (*open) (GstAudioEncoder *enc);
gboolean (*close) (GstAudioEncoder *enc);
/*< private >*/
gpointer _gst_reserved[GST_PADDING_LARGE-2];
};
GType gst_audio_encoder_get_type (void);
GstFlowReturn gst_audio_encoder_finish_frame (GstAudioEncoder * enc,
GstBuffer * buffer,
gint samples);
GstCaps * gst_audio_encoder_proxy_getcaps (GstAudioEncoder * enc,
GstCaps * caps);
/* context parameters */
GstAudioInfo * gst_audio_encoder_get_audio_info (GstAudioEncoder * enc);
gint gst_audio_encoder_get_frame_samples_min (GstAudioEncoder * enc);
void gst_audio_encoder_set_frame_samples_min (GstAudioEncoder * enc, gint num);
gint gst_audio_encoder_get_frame_samples_max (GstAudioEncoder * enc);
void gst_audio_encoder_set_frame_samples_max (GstAudioEncoder * enc, gint num);
gint gst_audio_encoder_get_frame_max (GstAudioEncoder * enc);
void gst_audio_encoder_set_frame_max (GstAudioEncoder * enc, gint num);
gint gst_audio_encoder_get_lookahead (GstAudioEncoder * enc);
void gst_audio_encoder_set_lookahead (GstAudioEncoder * enc, gint num);
void gst_audio_encoder_get_latency (GstAudioEncoder * enc,
GstClockTime * min,
GstClockTime * max);
void gst_audio_encoder_set_latency (GstAudioEncoder * enc,
GstClockTime min,
GstClockTime max);
/* object properties */
void gst_audio_encoder_set_mark_granule (GstAudioEncoder * enc,
gboolean enabled);
gboolean gst_audio_encoder_get_mark_granule (GstAudioEncoder * enc);
void gst_audio_encoder_set_perfect_timestamp (GstAudioEncoder * enc,
gboolean enabled);
gboolean gst_audio_encoder_get_perfect_timestamp (GstAudioEncoder * enc);
void gst_audio_encoder_set_hard_resync (GstAudioEncoder * enc,
gboolean enabled);
gboolean gst_audio_encoder_get_hard_resync (GstAudioEncoder * enc);
void gst_audio_encoder_set_tolerance (GstAudioEncoder * enc,
gint64 tolerance);
gint64 gst_audio_encoder_get_tolerance (GstAudioEncoder * enc);
void gst_audio_encoder_set_hard_min (GstAudioEncoder * enc,
gboolean enabled);
gboolean gst_audio_encoder_get_hard_min (GstAudioEncoder * enc);
void gst_audio_encoder_set_drainable (GstAudioEncoder * enc,
gboolean enabled);
gboolean gst_audio_encoder_get_drainable (GstAudioEncoder * enc);
void gst_audio_encoder_merge_tags (GstAudioEncoder * enc,
const GstTagList * tags, GstTagMergeMode mode);
G_END_DECLS
#endif /* __GST_AUDIO_ENCODER_H__ */
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