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 | |
58 | G_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 | */ |
94 | typedef 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 | |
102 | typedef struct _GstAudioBaseSink GstAudioBaseSink; |
103 | typedef struct _GstAudioBaseSinkClass GstAudioBaseSinkClass; |
104 | typedef 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 | */ |
120 | typedef 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 | */ |
165 | typedef 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 | */ |
172 | struct _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 | */ |
210 | struct _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 | |
223 | GST_AUDIO_API |
224 | GType gst_audio_base_sink_get_type(void); |
225 | |
226 | GST_AUDIO_API |
227 | GstAudioRingBuffer * |
228 | gst_audio_base_sink_create_ringbuffer (GstAudioBaseSink *sink); |
229 | |
230 | GST_AUDIO_API |
231 | void gst_audio_base_sink_set_provide_clock (GstAudioBaseSink *sink, gboolean provide); |
232 | |
233 | GST_AUDIO_API |
234 | gboolean gst_audio_base_sink_get_provide_clock (GstAudioBaseSink *sink); |
235 | |
236 | GST_AUDIO_API |
237 | void gst_audio_base_sink_set_slave_method (GstAudioBaseSink *sink, |
238 | GstAudioBaseSinkSlaveMethod method); |
239 | GST_AUDIO_API |
240 | GstAudioBaseSinkSlaveMethod |
241 | gst_audio_base_sink_get_slave_method (GstAudioBaseSink *sink); |
242 | |
243 | GST_AUDIO_API |
244 | void gst_audio_base_sink_set_drift_tolerance (GstAudioBaseSink *sink, |
245 | gint64 drift_tolerance); |
246 | GST_AUDIO_API |
247 | gint64 gst_audio_base_sink_get_drift_tolerance (GstAudioBaseSink *sink); |
248 | |
249 | GST_AUDIO_API |
250 | void gst_audio_base_sink_set_alignment_threshold (GstAudioBaseSink * sink, |
251 | GstClockTime alignment_threshold); |
252 | GST_AUDIO_API |
253 | GstClockTime |
254 | gst_audio_base_sink_get_alignment_threshold (GstAudioBaseSink * sink); |
255 | |
256 | GST_AUDIO_API |
257 | void gst_audio_base_sink_set_discont_wait (GstAudioBaseSink * sink, |
258 | GstClockTime discont_wait); |
259 | GST_AUDIO_API |
260 | GstClockTime |
261 | gst_audio_base_sink_get_discont_wait (GstAudioBaseSink * sink); |
262 | |
263 | GST_AUDIO_API |
264 | void |
265 | gst_audio_base_sink_set_custom_slaving_callback (GstAudioBaseSink * sink, |
266 | GstAudioBaseSinkCustomSlavingCallback callback, |
267 | gpointer user_data, |
268 | GDestroyNotify notify); |
269 | |
270 | GST_AUDIO_API |
271 | void gst_audio_base_sink_report_device_failure (GstAudioBaseSink * sink); |
272 | |
273 | G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstAudioBaseSink, gst_object_unref) |
274 | |
275 | G_END_DECLS |
276 | |
277 | #endif /* __GST_AUDIO_BASE_SINK_H__ */ |
278 | |