1/* GStreamer
2 * Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
3 * Copyright (C) 2011 Nokia Corporation. All rights reserved.
4 * Contact: Stefan Kost <stefan.kost@nokia.com>
5 *
6 * This library is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Library General Public
8 * License as published by the Free Software Foundation; either
9 * version 2 of the License, or (at your option) any later version.
10 *
11 * This library is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Library General Public License for more details.
15 *
16 * You should have received a copy of the GNU Library General Public
17 * License along with this library; if not, write to the
18 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
19 * Boston, MA 02110-1301, USA.
20 */
21
22#ifndef __GST_AUDIO_AUDIO_H__
23#include <gst/audio/audio.h>
24#endif
25
26#ifndef __GST_AUDIO_ENCODER_H__
27#define __GST_AUDIO_ENCODER_H__
28
29#include <gst/gst.h>
30
31G_BEGIN_DECLS
32
33#define GST_TYPE_AUDIO_ENCODER (gst_audio_encoder_get_type())
34#define GST_AUDIO_ENCODER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_ENCODER,GstAudioEncoder))
35#define GST_AUDIO_ENCODER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_ENCODER,GstAudioEncoderClass))
36#define GST_AUDIO_ENCODER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj),GST_TYPE_AUDIO_ENCODER,GstAudioEncoderClass))
37#define GST_IS_AUDIO_ENCODER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_ENCODER))
38#define GST_IS_AUDIO_ENCODER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_ENCODER))
39#define GST_AUDIO_ENCODER_CAST(obj) ((GstAudioEncoder *)(obj))
40
41/**
42 * GST_AUDIO_ENCODER_SINK_NAME:
43 *
44 * the name of the templates for the sink pad
45 */
46#define GST_AUDIO_ENCODER_SINK_NAME "sink"
47/**
48 * GST_AUDIO_ENCODER_SRC_NAME:
49 *
50 * the name of the templates for the source pad
51 */
52#define GST_AUDIO_ENCODER_SRC_NAME "src"
53
54/**
55 * GST_AUDIO_ENCODER_SRC_PAD:
56 * @obj: audio encoder instance
57 *
58 * Gives the pointer to the source #GstPad object of the element.
59 */
60#define GST_AUDIO_ENCODER_SRC_PAD(obj) (GST_AUDIO_ENCODER_CAST (obj)->srcpad)
61
62/**
63 * GST_AUDIO_ENCODER_SINK_PAD:
64 * @obj: audio encoder instance
65 *
66 * Gives the pointer to the sink #GstPad object of the element.
67 */
68#define GST_AUDIO_ENCODER_SINK_PAD(obj) (GST_AUDIO_ENCODER_CAST (obj)->sinkpad)
69
70/**
71 * GST_AUDIO_ENCODER_INPUT_SEGMENT:
72 * @obj: base parse instance
73 *
74 * Gives the input segment of the element.
75 */
76#define GST_AUDIO_ENCODER_INPUT_SEGMENT(obj) (GST_AUDIO_ENCODER_CAST (obj)->input_segment)
77
78/**
79 * GST_AUDIO_ENCODER_OUTPUT_SEGMENT:
80 * @obj: base parse instance
81 *
82 * Gives the output segment of the element.
83 */
84#define GST_AUDIO_ENCODER_OUTPUT_SEGMENT(obj) (GST_AUDIO_ENCODER_CAST (obj)->output_segment)
85
86#define GST_AUDIO_ENCODER_STREAM_LOCK(enc) g_rec_mutex_lock (&GST_AUDIO_ENCODER (enc)->stream_lock)
87#define GST_AUDIO_ENCODER_STREAM_UNLOCK(enc) g_rec_mutex_unlock (&GST_AUDIO_ENCODER (enc)->stream_lock)
88
89typedef struct _GstAudioEncoder GstAudioEncoder;
90typedef struct _GstAudioEncoderClass GstAudioEncoderClass;
91
92typedef struct _GstAudioEncoderPrivate GstAudioEncoderPrivate;
93
94/**
95 * GstAudioEncoder:
96 *
97 * The opaque #GstAudioEncoder data structure.
98 */
99struct _GstAudioEncoder {
100 GstElement element;
101
102 /*< protected >*/
103 /* source and sink pads */
104 GstPad *sinkpad;
105 GstPad *srcpad;
106
107 /* protects all data processing, i.e. is locked
108 * in the chain function, finish_frame and when
109 * processing serialized events */
110 GRecMutex stream_lock;
111
112 /* MT-protected (with STREAM_LOCK) */
113 GstSegment input_segment;
114 GstSegment output_segment;
115
116 /*< private >*/
117 GstAudioEncoderPrivate *priv;
118
119 gpointer _gst_reserved[GST_PADDING_LARGE];
120};
121
122/**
123 * GstAudioEncoderClass:
124 * @element_class: The parent class structure
125 * @start: Optional.
126 * Called when the element starts processing.
127 * Allows opening external resources.
128 * @stop: Optional.
129 * Called when the element stops processing.
130 * Allows closing external resources.
131 * @set_format: Notifies subclass of incoming data format.
132 * GstAudioInfo contains the format according to provided caps.
133 * @handle_frame: Provides input samples (or NULL to clear any remaining data)
134 * according to directions as configured by the subclass
135 * using the API. Input data ref management is performed
136 * by base class, subclass should not care or intervene,
137 * and input data is only valid until next call to base class,
138 * most notably a call to gst_audio_encoder_finish_frame().
139 * @flush: Optional.
140 * Instructs subclass to clear any codec caches and discard
141 * any pending samples and not yet returned encoded data.
142 * @sink_event: Optional.
143 * Event handler on the sink pad. Subclasses should chain up to
144 * the parent implementation to invoke the default handler.
145 * @src_event: Optional.
146 * Event handler on the src pad. Subclasses should chain up to
147 * the parent implementation to invoke the default handler.
148 * @pre_push: Optional.
149 * Called just prior to pushing (encoded data) buffer downstream.
150 * Subclass has full discretionary access to buffer,
151 * and a not OK flow return will abort downstream pushing.
152 * @getcaps: Optional.
153 * Allows for a custom sink getcaps implementation (e.g.
154 * for multichannel input specification). If not implemented,
155 * default returns gst_audio_encoder_proxy_getcaps
156 * applied to sink template caps.
157 * @open: Optional.
158 * Called when the element changes to GST_STATE_READY.
159 * Allows opening external resources.
160 * @close: Optional.
161 * Called when the element changes to GST_STATE_NULL.
162 * Allows closing external resources.
163 * @negotiate: Optional.
164 * Negotiate with downstream and configure buffer pools, etc.
165 * Subclasses should chain up to the parent implementation to
166 * invoke the default handler.
167 * @decide_allocation: Optional.
168 * Setup the allocation parameters for allocating output
169 * buffers. The passed in query contains the result of the
170 * downstream allocation query.
171 * Subclasses should chain up to the parent implementation to
172 * invoke the default handler.
173 * @propose_allocation: Optional.
174 * Propose buffer allocation parameters for upstream elements.
175 * Subclasses should chain up to the parent implementation to
176 * invoke the default handler.
177 * @transform_meta: Optional. Transform the metadata on the input buffer to the
178 * output buffer. By default this method copies all meta without
179 * tags and meta with only the "audio" tag. subclasses can
180 * implement this method and return %TRUE if the metadata is to be
181 * copied. Since: 1.6
182 * @sink_query: Optional.
183 * Query handler on the sink pad. This function should
184 * return TRUE if the query could be performed. Subclasses
185 * should chain up to the parent implementation to invoke the
186 * default handler. Since: 1.6
187 * @src_query: Optional.
188 * Query handler on the source pad. This function should
189 * return TRUE if the query could be performed. Subclasses
190 * should chain up to the parent implementation to invoke the
191 * default handler. Since: 1.6
192 *
193 * Subclasses can override any of the available virtual methods or not, as
194 * needed. At minimum @set_format and @handle_frame needs to be overridden.
195 */
196struct _GstAudioEncoderClass {
197 GstElementClass element_class;
198
199 /*< public >*/
200 /* virtual methods for subclasses */
201
202 gboolean (*start) (GstAudioEncoder *enc);
203
204 gboolean (*stop) (GstAudioEncoder *enc);
205
206 gboolean (*set_format) (GstAudioEncoder *enc,
207 GstAudioInfo *info);
208
209 GstFlowReturn (*handle_frame) (GstAudioEncoder *enc,
210 GstBuffer *buffer);
211
212 void (*flush) (GstAudioEncoder *enc);
213
214 GstFlowReturn (*pre_push) (GstAudioEncoder *enc,
215 GstBuffer **buffer);
216
217 gboolean (*sink_event) (GstAudioEncoder *enc,
218 GstEvent *event);
219
220 gboolean (*src_event) (GstAudioEncoder *enc,
221 GstEvent *event);
222
223 GstCaps * (*getcaps) (GstAudioEncoder *enc, GstCaps *filter);
224
225 gboolean (*open) (GstAudioEncoder *enc);
226
227 gboolean (*close) (GstAudioEncoder *enc);
228
229 gboolean (*negotiate) (GstAudioEncoder *enc);
230
231 gboolean (*decide_allocation) (GstAudioEncoder *enc, GstQuery *query);
232
233 gboolean (*propose_allocation) (GstAudioEncoder * enc,
234 GstQuery * query);
235
236 gboolean (*transform_meta) (GstAudioEncoder *enc, GstBuffer *outbuf,
237 GstMeta *meta, GstBuffer *inbuf);
238
239 gboolean (*sink_query) (GstAudioEncoder *encoder,
240 GstQuery *query);
241
242 gboolean (*src_query) (GstAudioEncoder *encoder,
243 GstQuery *query);
244
245
246 /*< private >*/
247 gpointer _gst_reserved[GST_PADDING_LARGE-3];
248};
249
250GST_AUDIO_API
251GType gst_audio_encoder_get_type (void);
252
253GST_AUDIO_API
254GstFlowReturn gst_audio_encoder_finish_frame (GstAudioEncoder * enc,
255 GstBuffer * buffer,
256 gint samples);
257
258GST_AUDIO_API
259GstCaps * gst_audio_encoder_proxy_getcaps (GstAudioEncoder * enc,
260 GstCaps * caps,
261 GstCaps * filter);
262
263GST_AUDIO_API
264gboolean gst_audio_encoder_set_output_format (GstAudioEncoder * enc,
265 GstCaps * caps);
266
267GST_AUDIO_API
268gboolean gst_audio_encoder_negotiate (GstAudioEncoder * enc);
269
270GST_AUDIO_API
271GstBuffer * gst_audio_encoder_allocate_output_buffer (GstAudioEncoder * enc,
272 gsize size);
273
274/* context parameters */
275
276GST_AUDIO_API
277GstAudioInfo * gst_audio_encoder_get_audio_info (GstAudioEncoder * enc);
278
279GST_AUDIO_API
280gint gst_audio_encoder_get_frame_samples_min (GstAudioEncoder * enc);
281
282GST_AUDIO_API
283void gst_audio_encoder_set_frame_samples_min (GstAudioEncoder * enc, gint num);
284
285GST_AUDIO_API
286gint gst_audio_encoder_get_frame_samples_max (GstAudioEncoder * enc);
287
288GST_AUDIO_API
289void gst_audio_encoder_set_frame_samples_max (GstAudioEncoder * enc, gint num);
290
291GST_AUDIO_API
292gint gst_audio_encoder_get_frame_max (GstAudioEncoder * enc);
293
294GST_AUDIO_API
295void gst_audio_encoder_set_frame_max (GstAudioEncoder * enc, gint num);
296
297GST_AUDIO_API
298gint gst_audio_encoder_get_lookahead (GstAudioEncoder * enc);
299
300GST_AUDIO_API
301void gst_audio_encoder_set_lookahead (GstAudioEncoder * enc, gint num);
302
303GST_AUDIO_API
304void gst_audio_encoder_get_latency (GstAudioEncoder * enc,
305 GstClockTime * min,
306 GstClockTime * max);
307
308GST_AUDIO_API
309void gst_audio_encoder_set_latency (GstAudioEncoder * enc,
310 GstClockTime min,
311 GstClockTime max);
312
313GST_AUDIO_API
314void gst_audio_encoder_set_headers (GstAudioEncoder * enc,
315 GList * headers);
316
317GST_AUDIO_API
318void gst_audio_encoder_set_allocation_caps (GstAudioEncoder * enc,
319 GstCaps * allocation_caps);
320
321/* object properties */
322
323GST_AUDIO_API
324void gst_audio_encoder_set_mark_granule (GstAudioEncoder * enc,
325 gboolean enabled);
326
327GST_AUDIO_API
328gboolean gst_audio_encoder_get_mark_granule (GstAudioEncoder * enc);
329
330GST_AUDIO_API
331void gst_audio_encoder_set_perfect_timestamp (GstAudioEncoder * enc,
332 gboolean enabled);
333
334GST_AUDIO_API
335gboolean gst_audio_encoder_get_perfect_timestamp (GstAudioEncoder * enc);
336
337GST_AUDIO_API
338void gst_audio_encoder_set_hard_resync (GstAudioEncoder * enc,
339 gboolean enabled);
340
341GST_AUDIO_API
342gboolean gst_audio_encoder_get_hard_resync (GstAudioEncoder * enc);
343
344GST_AUDIO_API
345void gst_audio_encoder_set_tolerance (GstAudioEncoder * enc,
346 GstClockTime tolerance);
347
348GST_AUDIO_API
349GstClockTime gst_audio_encoder_get_tolerance (GstAudioEncoder * enc);
350
351GST_AUDIO_API
352void gst_audio_encoder_set_hard_min (GstAudioEncoder * enc,
353 gboolean enabled);
354
355GST_AUDIO_API
356gboolean gst_audio_encoder_get_hard_min (GstAudioEncoder * enc);
357
358GST_AUDIO_API
359void gst_audio_encoder_set_drainable (GstAudioEncoder * enc,
360 gboolean enabled);
361
362GST_AUDIO_API
363gboolean gst_audio_encoder_get_drainable (GstAudioEncoder * enc);
364
365GST_AUDIO_API
366void gst_audio_encoder_get_allocator (GstAudioEncoder * enc,
367 GstAllocator ** allocator,
368 GstAllocationParams * params);
369
370GST_AUDIO_API
371void gst_audio_encoder_merge_tags (GstAudioEncoder * enc,
372 const GstTagList * tags, GstTagMergeMode mode);
373
374G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstAudioEncoder, gst_object_unref)
375
376G_END_DECLS
377
378#endif /* __GST_AUDIO_ENCODER_H__ */
379

source code of include/gstreamer-1.0/gst/audio/gstaudioencoder.h