1/* GStreamer
2 * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
3 * 2005 Wim Taymans <wim@fluendo.com>
4 *
5 * gstaudiobasesink.h:
6 *
7 * This library is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Library General Public
9 * License as published by the Free Software Foundation; either
10 * version 2 of the License, or (at your option) any later version.
11 *
12 * This library is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Library General Public License for more details.
16 *
17 * You should have received a copy of the GNU Library General Public
18 * License along with this library; if not, write to the
19 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
20 * Boston, MA 02110-1301, USA.
21 */
22
23/* a base class for audio sinks.
24 *
25 * It uses a ringbuffer to schedule playback of samples. This makes
26 * it very easy to drop or insert samples to align incoming
27 * buffers to the exact playback timestamp.
28 *
29 * Subclasses must provide a ringbuffer pointing to either DMA
30 * memory or regular memory. A subclass should also call a callback
31 * function when it has played N segments in the buffer. The subclass
32 * is free to use a thread to signal this callback, use EIO or any
33 * other mechanism.
34 *
35 * The base class is able to operate in push or pull mode. The chain
36 * mode will queue the samples in the ringbuffer as much as possible.
37 * The available space is calculated in the callback function.
38 *
39 * The pull mode will pull_range() a new buffer of N samples with a
40 * configurable latency. This allows for high-end real time
41 * audio processing pipelines driven by the audiosink. The callback
42 * function will be used to perform a pull_range() on the sinkpad.
43 * The thread scheduling the callback can be a real-time thread.
44 *
45 * Subclasses must implement a GstAudioRingBuffer in addition to overriding
46 * the methods in GstBaseSink and this class.
47 */
48
49#ifndef __GST_AUDIO_AUDIO_H__
50#include <gst/audio/audio.h>
51#endif
52
53#ifndef __GST_AUDIO_BASE_SINK_H__
54#define __GST_AUDIO_BASE_SINK_H__
55
56#include <gst/base/gstbasesink.h>
57
58G_BEGIN_DECLS
59
60#define GST_TYPE_AUDIO_BASE_SINK (gst_audio_base_sink_get_type())
61#define GST_AUDIO_BASE_SINK(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_BASE_SINK,GstAudioBaseSink))
62#define GST_AUDIO_BASE_SINK_CAST(obj) ((GstAudioBaseSink*)obj)
63#define GST_AUDIO_BASE_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_BASE_SINK,GstAudioBaseSinkClass))
64#define GST_AUDIO_BASE_SINK_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_AUDIO_BASE_SINK, GstAudioBaseSinkClass))
65#define GST_IS_AUDIO_BASE_SINK(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_BASE_SINK))
66#define GST_IS_AUDIO_BASE_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_BASE_SINK))
67
68/**
69 * GST_AUDIO_BASE_SINK_CLOCK:
70 * @obj: a #GstAudioBaseSink
71 *
72 * Get the #GstClock of @obj.
73 */
74#define GST_AUDIO_BASE_SINK_CLOCK(obj) (GST_AUDIO_BASE_SINK (obj)->clock)
75/**
76 * GST_AUDIO_BASE_SINK_PAD:
77 * @obj: a #GstAudioBaseSink
78 *
79 * Get the sink #GstPad of @obj.
80 */
81#define GST_AUDIO_BASE_SINK_PAD(obj) (GST_BASE_SINK (obj)->sinkpad)
82
83/**
84 * GstAudioBaseSinkSlaveMethod:
85 * @GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE: Resample to match the master clock
86 * @GST_AUDIO_BASE_SINK_SLAVE_SKEW: Adjust playout pointer when master clock
87 * drifts too much.
88 * @GST_AUDIO_BASE_SINK_SLAVE_NONE: No adjustment is done.
89 * @GST_AUDIO_BASE_SINK_SLAVE_CUSTOM: Use custom clock slaving algorithm (Since: 1.6)
90 *
91 * Different possible clock slaving algorithms used when the internal audio
92 * clock is not selected as the pipeline master clock.
93 */
94typedef enum
95{
96 GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE,
97 GST_AUDIO_BASE_SINK_SLAVE_SKEW,
98 GST_AUDIO_BASE_SINK_SLAVE_NONE,
99 GST_AUDIO_BASE_SINK_SLAVE_CUSTOM
100} GstAudioBaseSinkSlaveMethod;
101
102typedef struct _GstAudioBaseSink GstAudioBaseSink;
103typedef struct _GstAudioBaseSinkClass GstAudioBaseSinkClass;
104typedef struct _GstAudioBaseSinkPrivate GstAudioBaseSinkPrivate;
105
106/**
107 * GstAudioBaseSinkDiscontReason:
108 * @GST_AUDIO_BASE_SINK_DISCONT_REASON_NO_DISCONT: No discontinuity occurred
109 * @GST_AUDIO_BASE_SINK_DISCONT_REASON_NEW_CAPS: New caps are set, causing renegotiotion
110 * @GST_AUDIO_BASE_SINK_DISCONT_REASON_FLUSH: Samples have been flushed
111 * @GST_AUDIO_BASE_SINK_DISCONT_REASON_SYNC_LATENCY: Sink was synchronized to the estimated latency (occurs during initialization)
112 * @GST_AUDIO_BASE_SINK_DISCONT_REASON_ALIGNMENT: Aligning buffers failed because the timestamps are too discontinuous
113 * @GST_AUDIO_BASE_SINK_DISCONT_REASON_DEVICE_FAILURE: Audio output device experienced and recovered from an error but introduced latency in the process (see also gst_audio_base_sink_report_device_failure())
114 *
115 * Different possible reasons for discontinuities. This enum is useful for the custom
116 * slave method.
117 *
118 * Since: 1.6
119 */
120typedef enum
121{
122 GST_AUDIO_BASE_SINK_DISCONT_REASON_NO_DISCONT,
123 GST_AUDIO_BASE_SINK_DISCONT_REASON_NEW_CAPS,
124 GST_AUDIO_BASE_SINK_DISCONT_REASON_FLUSH,
125 GST_AUDIO_BASE_SINK_DISCONT_REASON_SYNC_LATENCY,
126 GST_AUDIO_BASE_SINK_DISCONT_REASON_ALIGNMENT,
127 GST_AUDIO_BASE_SINK_DISCONT_REASON_DEVICE_FAILURE
128} GstAudioBaseSinkDiscontReason;
129
130/**
131 * GstAudioBaseSinkCustomSlavingCallback:
132 * @sink: a #GstAudioBaseSink
133 * @etime: external clock time
134 * @itime: internal clock time
135 * @requested_skew: skew amount requested by the callback
136 * @discont_reason: reason for discontinuity (if any)
137 * @user_data: user data
138 *
139 * This function is set with gst_audio_base_sink_set_custom_slaving_callback()
140 * and is called during playback. It receives the current time of external and
141 * internal clocks, which the callback can then use to apply any custom
142 * slaving/synchronization schemes.
143 *
144 * The external clock is the sink's element clock, the internal one is the
145 * internal audio clock. The internal audio clock's calibration is applied to
146 * the timestamps before they are passed to the callback. The difference between
147 * etime and itime is the skew; how much internal and external clock lie apart
148 * from each other. A skew of 0 means both clocks are perfectly in sync.
149 * itime > etime means the external clock is going slower, while itime < etime
150 * means it is going faster than the internal clock. etime and itime are always
151 * valid timestamps, except for when a discontinuity happens.
152 *
153 * requested_skew is an output value the callback can write to. It informs the
154 * sink of whether or not it should move the playout pointer, and if so, by how
155 * much. This pointer is only NULL if a discontinuity occurs; otherwise, it is
156 * safe to write to *requested_skew. The default skew is 0.
157 *
158 * The sink may experience discontinuities. If one happens, discont is TRUE,
159 * itime, etime are set to GST_CLOCK_TIME_NONE, and requested_skew is NULL.
160 * This makes it possible to reset custom clock slaving algorithms when a
161 * discontinuity happens.
162 *
163 * Since: 1.6
164 */
165typedef void (*GstAudioBaseSinkCustomSlavingCallback) (GstAudioBaseSink *sink, GstClockTime etime, GstClockTime itime, GstClockTimeDiff *requested_skew, GstAudioBaseSinkDiscontReason discont_reason, gpointer user_data);
166
167/**
168 * GstAudioBaseSink:
169 *
170 * Opaque #GstAudioBaseSink.
171 */
172struct _GstAudioBaseSink {
173 GstBaseSink element;
174
175 /*< protected >*/ /* with LOCK */
176 /* our ringbuffer */
177 GstAudioRingBuffer *ringbuffer;
178
179 /* required buffer and latency in microseconds */
180 guint64 buffer_time;
181 guint64 latency_time;
182
183 /* the next sample to write */
184 guint64 next_sample;
185
186 /* clock */
187 GstClock *provided_clock;
188
189 /* with g_atomic_; currently rendering eos */
190 gboolean eos_rendering;
191
192 /*< private >*/
193 GstAudioBaseSinkPrivate *priv;
194
195 gpointer _gst_reserved[GST_PADDING];
196};
197
198/**
199 * GstAudioBaseSinkClass:
200 * @parent_class: the parent class.
201 * @create_ringbuffer: create and return a #GstAudioRingBuffer to write to.
202 * @payload: payload data in a format suitable to write to the sink. If no
203 * payloading is required, returns a reffed copy of the original
204 * buffer, else returns the payloaded buffer with all other metadata
205 * copied.
206 *
207 * #GstAudioBaseSink class. Override the vmethod to implement
208 * functionality.
209 */
210struct _GstAudioBaseSinkClass {
211 GstBaseSinkClass parent_class;
212
213 /* subclass ringbuffer allocation */
214 GstAudioRingBuffer* (*create_ringbuffer) (GstAudioBaseSink *sink);
215
216 /* subclass payloader */
217 GstBuffer* (*payload) (GstAudioBaseSink *sink,
218 GstBuffer *buffer);
219 /*< private >*/
220 gpointer _gst_reserved[GST_PADDING];
221};
222
223GST_AUDIO_API
224GType gst_audio_base_sink_get_type(void);
225
226GST_AUDIO_API
227GstAudioRingBuffer *
228 gst_audio_base_sink_create_ringbuffer (GstAudioBaseSink *sink);
229
230GST_AUDIO_API
231void gst_audio_base_sink_set_provide_clock (GstAudioBaseSink *sink, gboolean provide);
232
233GST_AUDIO_API
234gboolean gst_audio_base_sink_get_provide_clock (GstAudioBaseSink *sink);
235
236GST_AUDIO_API
237void gst_audio_base_sink_set_slave_method (GstAudioBaseSink *sink,
238 GstAudioBaseSinkSlaveMethod method);
239GST_AUDIO_API
240GstAudioBaseSinkSlaveMethod
241 gst_audio_base_sink_get_slave_method (GstAudioBaseSink *sink);
242
243GST_AUDIO_API
244void gst_audio_base_sink_set_drift_tolerance (GstAudioBaseSink *sink,
245 gint64 drift_tolerance);
246GST_AUDIO_API
247gint64 gst_audio_base_sink_get_drift_tolerance (GstAudioBaseSink *sink);
248
249GST_AUDIO_API
250void gst_audio_base_sink_set_alignment_threshold (GstAudioBaseSink * sink,
251 GstClockTime alignment_threshold);
252GST_AUDIO_API
253GstClockTime
254 gst_audio_base_sink_get_alignment_threshold (GstAudioBaseSink * sink);
255
256GST_AUDIO_API
257void gst_audio_base_sink_set_discont_wait (GstAudioBaseSink * sink,
258 GstClockTime discont_wait);
259GST_AUDIO_API
260GstClockTime
261 gst_audio_base_sink_get_discont_wait (GstAudioBaseSink * sink);
262
263GST_AUDIO_API
264void
265gst_audio_base_sink_set_custom_slaving_callback (GstAudioBaseSink * sink,
266 GstAudioBaseSinkCustomSlavingCallback callback,
267 gpointer user_data,
268 GDestroyNotify notify);
269
270GST_AUDIO_API
271void gst_audio_base_sink_report_device_failure (GstAudioBaseSink * sink);
272
273G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstAudioBaseSink, gst_object_unref)
274
275G_END_DECLS
276
277#endif /* __GST_AUDIO_BASE_SINK_H__ */
278

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