mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-18 15:51:11 +00:00
255 lines
9.4 KiB
C
255 lines
9.4 KiB
C
/* 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.
|
|
*
|
|
* 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, GstCaps *filter);
|
|
|
|
/*< private >*/
|
|
gpointer _gst_reserved[GST_PADDING_LARGE];
|
|
};
|
|
|
|
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_merge_tags (GstAudioEncoder * enc,
|
|
const GstTagList * tags, GstTagMergeMode mode);
|
|
|
|
G_END_DECLS
|
|
|
|
#endif /* __GST_AUDIO_ENCODER_H__ */
|