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
|
/* GStreamer
* Copyright (C) 2010 Sebastian Dröge <sebastian.droege@collabora.co.uk>
*
* 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.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/audio/audio.h>
#include "gstaudiosegmentclip.h"
static GstStaticPadTemplate sink_pad_template =
GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL)));
static GstStaticPadTemplate src_pad_template =
GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL)));
static void gst_audio_segment_clip_reset (GstSegmentClip * self);
static GstFlowReturn gst_audio_segment_clip_clip_buffer (GstSegmentClip * self,
GstBuffer * buffer, GstBuffer ** outbuf);
static gboolean gst_audio_segment_clip_set_caps (GstSegmentClip * self,
GstCaps * caps);
GST_DEBUG_CATEGORY_STATIC (gst_audio_segment_clip_debug);
#define GST_CAT_DEFAULT gst_audio_segment_clip_debug
G_DEFINE_TYPE (GstAudioSegmentClip, gst_audio_segment_clip,
GST_TYPE_SEGMENT_CLIP);
static void
gst_audio_segment_clip_class_init (GstAudioSegmentClipClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstSegmentClipClass *segment_clip_klass = GST_SEGMENT_CLIP_CLASS (klass);
GST_DEBUG_CATEGORY_INIT (gst_audio_segment_clip_debug, "audiosegmentclip", 0,
"audiosegmentclip element");
segment_clip_klass->reset = GST_DEBUG_FUNCPTR (gst_audio_segment_clip_reset);
segment_clip_klass->set_caps =
GST_DEBUG_FUNCPTR (gst_audio_segment_clip_set_caps);
segment_clip_klass->clip_buffer =
GST_DEBUG_FUNCPTR (gst_audio_segment_clip_clip_buffer);
gst_element_class_set_details_simple (element_class,
"Audio buffer segment clipper",
"Filter/Audio",
"Clips audio buffers to the configured segment",
"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_pad_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_pad_template));
}
static void
gst_audio_segment_clip_init (GstAudioSegmentClip * self)
{
}
static void
gst_audio_segment_clip_reset (GstSegmentClip * base)
{
GstAudioSegmentClip *self = GST_AUDIO_SEGMENT_CLIP (base);
GST_DEBUG_OBJECT (self, "Resetting internal state");
self->rate = self->framesize = 0;
}
static GstFlowReturn
gst_audio_segment_clip_clip_buffer (GstSegmentClip * base, GstBuffer * buffer,
GstBuffer ** outbuf)
{
GstAudioSegmentClip *self = GST_AUDIO_SEGMENT_CLIP (base);
GstSegment *segment = &base->segment;
GstClockTime timestamp = GST_BUFFER_TIMESTAMP (buffer);
GstClockTime duration = GST_BUFFER_DURATION (buffer);
guint64 offset = GST_BUFFER_OFFSET (buffer);
guint64 offset_end = GST_BUFFER_OFFSET_END (buffer);
guint size = gst_buffer_get_size (buffer);
if (!self->rate || !self->framesize) {
GST_ERROR_OBJECT (self, "Not negotiated yet");
gst_buffer_unref (buffer);
return GST_FLOW_NOT_NEGOTIATED;
}
if (segment->format != GST_FORMAT_DEFAULT &&
segment->format != GST_FORMAT_TIME) {
GST_DEBUG_OBJECT (self, "Unsupported segment format %s",
gst_format_get_name (segment->format));
*outbuf = buffer;
return GST_FLOW_OK;
}
if (!GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
GST_WARNING_OBJECT (self, "Buffer without valid timestamp");
*outbuf = buffer;
return GST_FLOW_OK;
}
*outbuf =
gst_audio_buffer_clip (buffer, segment, self->rate, self->framesize);
if (!*outbuf) {
GST_DEBUG_OBJECT (self, "Buffer outside the configured segment");
/* Now return unexpected if we're before/after the end */
if (segment->format == GST_FORMAT_TIME) {
if (segment->rate >= 0) {
if (segment->stop != -1 && timestamp >= segment->stop)
return GST_FLOW_EOS;
} else {
if (!GST_CLOCK_TIME_IS_VALID (duration))
duration =
gst_util_uint64_scale_int (size, GST_SECOND,
self->framesize * self->rate);
if (segment->start != -1 && timestamp + duration <= segment->start)
return GST_FLOW_EOS;
}
} else {
if (segment->rate >= 0) {
if (segment->stop != -1 && offset != -1 && offset >= segment->stop)
return GST_FLOW_EOS;
} else if (offset != -1 || offset_end != -1) {
if (offset_end == -1)
offset_end = offset + size / self->framesize;
if (segment->start != -1 && offset_end <= segment->start)
return GST_FLOW_EOS;
}
}
}
return GST_FLOW_OK;
}
static gboolean
gst_audio_segment_clip_set_caps (GstSegmentClip * base, GstCaps * caps)
{
GstAudioSegmentClip *self = GST_AUDIO_SEGMENT_CLIP (base);
gboolean ret;
GstAudioInfo info;
gint rate, channels, width;
gst_audio_info_init (&info);
ret = gst_audio_info_from_caps (&info, caps);
if (ret) {
rate = GST_AUDIO_INFO_RATE (&info);
channels = GST_AUDIO_INFO_CHANNELS (&info);
width = GST_AUDIO_INFO_WIDTH (&info);
GST_DEBUG_OBJECT (self, "Configured: rate %d channels %d width %d",
rate, channels, width);
self->rate = rate;
self->framesize = (width / 8) * channels;
}
return ret;
}
|
