/usr/include/gstreamer-0.10/gst/audio/gstaudioencoder.h is in libgstreamer-plugins-base0.10-dev 0.10.36-1.1ubuntu2.
This file is owned by root:root, with mode 0o644.
The actual contents of the file can be viewed below.
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 218 219 220 221 222 223 224 225 226 227 228 229 230 231 232 233 234 235 236 237 238 239 240 241 242 243 244 245 246 247 248 249 250 251 252 253 254 255 256 257 258 259 260 261 262 263 264 | /* 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);
/*< 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_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__ */
|