mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-07-28 01:05:07 +00:00
371e3e460a
We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
381 lines
15 KiB
C
381 lines
15 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., 51 Franklin St, Fifth Floor,
|
|
* Boston, MA 02110-1301, USA.
|
|
*/
|
|
|
|
#ifndef __GST_AUDIO_AUDIO_H__
|
|
#include <gst/audio/audio.h>
|
|
#endif
|
|
|
|
#ifndef __GST_AUDIO_ENCODER_H__
|
|
#define __GST_AUDIO_ENCODER_H__
|
|
|
|
#include <gst/gst.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
|
|
*/
|
|
#define GST_AUDIO_ENCODER_SINK_NAME "sink"
|
|
/**
|
|
* GST_AUDIO_ENCODER_SRC_NAME:
|
|
*
|
|
* the name of the templates for the source pad
|
|
*/
|
|
#define GST_AUDIO_ENCODER_SRC_NAME "src"
|
|
|
|
/**
|
|
* GST_AUDIO_ENCODER_SRC_PAD:
|
|
* @obj: audio encoder instance
|
|
*
|
|
* Gives the pointer to the source #GstPad object of the element.
|
|
*/
|
|
#define GST_AUDIO_ENCODER_SRC_PAD(obj) (GST_AUDIO_ENCODER_CAST (obj)->srcpad)
|
|
|
|
/**
|
|
* GST_AUDIO_ENCODER_SINK_PAD:
|
|
* @obj: audio encoder instance
|
|
*
|
|
* Gives the pointer to the sink #GstPad object of the element.
|
|
*/
|
|
#define GST_AUDIO_ENCODER_SINK_PAD(obj) (GST_AUDIO_ENCODER_CAST (obj)->sinkpad)
|
|
|
|
/**
|
|
* GST_AUDIO_ENCODER_INPUT_SEGMENT:
|
|
* @obj: base parse instance
|
|
*
|
|
* Gives the input segment of the element.
|
|
*/
|
|
#define GST_AUDIO_ENCODER_INPUT_SEGMENT(obj) (GST_AUDIO_ENCODER_CAST (obj)->input_segment)
|
|
|
|
/**
|
|
* GST_AUDIO_ENCODER_OUTPUT_SEGMENT:
|
|
* @obj: base parse instance
|
|
*
|
|
* Gives the output segment of the element.
|
|
*/
|
|
#define GST_AUDIO_ENCODER_OUTPUT_SEGMENT(obj) (GST_AUDIO_ENCODER_CAST (obj)->output_segment)
|
|
|
|
#define GST_AUDIO_ENCODER_STREAM_LOCK(enc) g_rec_mutex_lock (&GST_AUDIO_ENCODER (enc)->stream_lock)
|
|
#define GST_AUDIO_ENCODER_STREAM_UNLOCK(enc) g_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.
|
|
*/
|
|
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 */
|
|
GRecMutex stream_lock;
|
|
|
|
/* MT-protected (with STREAM_LOCK) */
|
|
GstSegment input_segment;
|
|
GstSegment output_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.
|
|
* @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.
|
|
* @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.
|
|
* @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.
|
|
* @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
|
|
* @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
|
|
*
|
|
* Subclasses can override any of the available virtual methods or not, as
|
|
* needed. At minimum @set_format and @handle_frame needs to be overridden.
|
|
*/
|
|
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 (*sink_event) (GstAudioEncoder *enc,
|
|
GstEvent *event);
|
|
|
|
gboolean (*src_event) (GstAudioEncoder *enc,
|
|
GstEvent *event);
|
|
|
|
GstCaps * (*getcaps) (GstAudioEncoder *enc, GstCaps *filter);
|
|
|
|
gboolean (*open) (GstAudioEncoder *enc);
|
|
|
|
gboolean (*close) (GstAudioEncoder *enc);
|
|
|
|
gboolean (*negotiate) (GstAudioEncoder *enc);
|
|
|
|
gboolean (*decide_allocation) (GstAudioEncoder *enc, GstQuery *query);
|
|
|
|
gboolean (*propose_allocation) (GstAudioEncoder * enc,
|
|
GstQuery * query);
|
|
|
|
gboolean (*transform_meta) (GstAudioEncoder *enc, GstBuffer *outbuf,
|
|
GstMeta *meta, GstBuffer *inbuf);
|
|
|
|
gboolean (*sink_query) (GstAudioEncoder *encoder,
|
|
GstQuery *query);
|
|
|
|
gboolean (*src_query) (GstAudioEncoder *encoder,
|
|
GstQuery *query);
|
|
|
|
|
|
/*< private >*/
|
|
gpointer _gst_reserved[GST_PADDING_LARGE-3];
|
|
};
|
|
|
|
GST_AUDIO_API
|
|
GType gst_audio_encoder_get_type (void);
|
|
|
|
GST_AUDIO_API
|
|
GstFlowReturn gst_audio_encoder_finish_frame (GstAudioEncoder * enc,
|
|
GstBuffer * buffer,
|
|
gint samples);
|
|
|
|
GST_AUDIO_API
|
|
GstCaps * gst_audio_encoder_proxy_getcaps (GstAudioEncoder * enc,
|
|
GstCaps * caps,
|
|
GstCaps * filter);
|
|
|
|
GST_AUDIO_API
|
|
gboolean gst_audio_encoder_set_output_format (GstAudioEncoder * enc,
|
|
GstCaps * caps);
|
|
|
|
GST_AUDIO_API
|
|
gboolean gst_audio_encoder_negotiate (GstAudioEncoder * enc);
|
|
|
|
GST_AUDIO_API
|
|
GstBuffer * gst_audio_encoder_allocate_output_buffer (GstAudioEncoder * enc,
|
|
gsize size);
|
|
|
|
/* context parameters */
|
|
|
|
GST_AUDIO_API
|
|
GstAudioInfo * gst_audio_encoder_get_audio_info (GstAudioEncoder * enc);
|
|
|
|
GST_AUDIO_API
|
|
gint gst_audio_encoder_get_frame_samples_min (GstAudioEncoder * enc);
|
|
|
|
GST_AUDIO_API
|
|
void gst_audio_encoder_set_frame_samples_min (GstAudioEncoder * enc, gint num);
|
|
|
|
GST_AUDIO_API
|
|
gint gst_audio_encoder_get_frame_samples_max (GstAudioEncoder * enc);
|
|
|
|
GST_AUDIO_API
|
|
void gst_audio_encoder_set_frame_samples_max (GstAudioEncoder * enc, gint num);
|
|
|
|
GST_AUDIO_API
|
|
gint gst_audio_encoder_get_frame_max (GstAudioEncoder * enc);
|
|
|
|
GST_AUDIO_API
|
|
void gst_audio_encoder_set_frame_max (GstAudioEncoder * enc, gint num);
|
|
|
|
GST_AUDIO_API
|
|
gint gst_audio_encoder_get_lookahead (GstAudioEncoder * enc);
|
|
|
|
GST_AUDIO_API
|
|
void gst_audio_encoder_set_lookahead (GstAudioEncoder * enc, gint num);
|
|
|
|
GST_AUDIO_API
|
|
void gst_audio_encoder_get_latency (GstAudioEncoder * enc,
|
|
GstClockTime * min,
|
|
GstClockTime * max);
|
|
|
|
GST_AUDIO_API
|
|
void gst_audio_encoder_set_latency (GstAudioEncoder * enc,
|
|
GstClockTime min,
|
|
GstClockTime max);
|
|
|
|
GST_AUDIO_API
|
|
void gst_audio_encoder_set_headers (GstAudioEncoder * enc,
|
|
GList * headers);
|
|
|
|
GST_AUDIO_API
|
|
void gst_audio_encoder_set_allocation_caps (GstAudioEncoder * enc,
|
|
GstCaps * allocation_caps);
|
|
|
|
/* object properties */
|
|
|
|
GST_AUDIO_API
|
|
void gst_audio_encoder_set_mark_granule (GstAudioEncoder * enc,
|
|
gboolean enabled);
|
|
|
|
GST_AUDIO_API
|
|
gboolean gst_audio_encoder_get_mark_granule (GstAudioEncoder * enc);
|
|
|
|
GST_AUDIO_API
|
|
void gst_audio_encoder_set_perfect_timestamp (GstAudioEncoder * enc,
|
|
gboolean enabled);
|
|
|
|
GST_AUDIO_API
|
|
gboolean gst_audio_encoder_get_perfect_timestamp (GstAudioEncoder * enc);
|
|
|
|
GST_AUDIO_API
|
|
void gst_audio_encoder_set_hard_resync (GstAudioEncoder * enc,
|
|
gboolean enabled);
|
|
|
|
GST_AUDIO_API
|
|
gboolean gst_audio_encoder_get_hard_resync (GstAudioEncoder * enc);
|
|
|
|
GST_AUDIO_API
|
|
void gst_audio_encoder_set_tolerance (GstAudioEncoder * enc,
|
|
GstClockTime tolerance);
|
|
|
|
GST_AUDIO_API
|
|
GstClockTime gst_audio_encoder_get_tolerance (GstAudioEncoder * enc);
|
|
|
|
GST_AUDIO_API
|
|
void gst_audio_encoder_set_hard_min (GstAudioEncoder * enc,
|
|
gboolean enabled);
|
|
|
|
GST_AUDIO_API
|
|
gboolean gst_audio_encoder_get_hard_min (GstAudioEncoder * enc);
|
|
|
|
GST_AUDIO_API
|
|
void gst_audio_encoder_set_drainable (GstAudioEncoder * enc,
|
|
gboolean enabled);
|
|
|
|
GST_AUDIO_API
|
|
gboolean gst_audio_encoder_get_drainable (GstAudioEncoder * enc);
|
|
|
|
GST_AUDIO_API
|
|
void gst_audio_encoder_get_allocator (GstAudioEncoder * enc,
|
|
GstAllocator ** allocator,
|
|
GstAllocationParams * params);
|
|
|
|
GST_AUDIO_API
|
|
void gst_audio_encoder_merge_tags (GstAudioEncoder * enc,
|
|
const GstTagList * tags, GstTagMergeMode mode);
|
|
|
|
#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
|
|
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstAudioEncoder, gst_object_unref)
|
|
#endif
|
|
|
|
G_END_DECLS
|
|
|
|
#endif /* __GST_AUDIO_ENCODER_H__ */
|