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
|
/*
* Farsight
* GStreamer GSM encoder
* Copyright (C) 2005 Philippe Khalaf <burger@speedy.org>
*
* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include "gstgsmenc.h"
GST_DEBUG_CATEGORY_STATIC (gsmenc_debug);
#define GST_CAT_DEFAULT (gsmenc_debug)
/* GSMEnc signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
/* FILL ME */
ARG_0
};
static gboolean gst_gsmenc_start (GstAudioEncoder * enc);
static gboolean gst_gsmenc_stop (GstAudioEncoder * enc);
static gboolean gst_gsmenc_set_format (GstAudioEncoder * enc,
GstAudioInfo * info);
static GstFlowReturn gst_gsmenc_handle_frame (GstAudioEncoder * enc,
GstBuffer * in_buf);
static GstStaticPadTemplate gsmenc_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-gsm, " "rate = (int) 8000, " "channels = (int) 1")
);
static GstStaticPadTemplate gsmenc_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " GST_AUDIO_NE (S16) ", "
"layout = (string) interleaved, "
"rate = (int) 8000, channels = (int) 1")
);
G_DEFINE_TYPE (GstGSMEnc, gst_gsmenc, GST_TYPE_AUDIO_ENCODER);
static void
gst_gsmenc_class_init (GstGSMEncClass * klass)
{
GstElementClass *element_class;
GstAudioEncoderClass *base_class;
element_class = (GstElementClass *) klass;
base_class = (GstAudioEncoderClass *) klass;
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gsmenc_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gsmenc_src_template));
gst_element_class_set_static_metadata (element_class, "GSM audio encoder",
"Codec/Encoder/Audio",
"Encodes GSM audio", "Philippe Khalaf <burger@speedy.org>");
base_class->start = GST_DEBUG_FUNCPTR (gst_gsmenc_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_gsmenc_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_gsmenc_set_format);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_gsmenc_handle_frame);
GST_DEBUG_CATEGORY_INIT (gsmenc_debug, "gsmenc", 0, "GSM Encoder");
}
static void
gst_gsmenc_init (GstGSMEnc * gsmenc)
{
}
static gboolean
gst_gsmenc_start (GstAudioEncoder * enc)
{
GstGSMEnc *gsmenc = GST_GSMENC (enc);
gint use_wav49;
GST_DEBUG_OBJECT (enc, "start");
gsmenc->state = gsm_create ();
/* turn off WAV49 handling */
use_wav49 = 0;
gsm_option (gsmenc->state, GSM_OPT_WAV49, &use_wav49);
return TRUE;
}
static gboolean
gst_gsmenc_stop (GstAudioEncoder * enc)
{
GstGSMEnc *gsmenc = GST_GSMENC (enc);
GST_DEBUG_OBJECT (enc, "stop");
gsm_destroy (gsmenc->state);
return TRUE;
}
static gboolean
gst_gsmenc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
{
GstCaps *srccaps;
srccaps = gst_static_pad_template_get_caps (&gsmenc_src_template);
gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (benc), srccaps);
gst_caps_unref (srccaps);
/* report needs to base class */
gst_audio_encoder_set_frame_samples_min (benc, 160);
gst_audio_encoder_set_frame_samples_max (benc, 160);
gst_audio_encoder_set_frame_max (benc, 1);
return TRUE;
}
static GstFlowReturn
gst_gsmenc_handle_frame (GstAudioEncoder * benc, GstBuffer * buffer)
{
GstGSMEnc *gsmenc;
gsm_signal *data;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *outbuf;
GstMapInfo map, omap;
gsmenc = GST_GSMENC (benc);
/* we don't deal with squeezing remnants, so simply discard those */
if (G_UNLIKELY (buffer == NULL)) {
GST_DEBUG_OBJECT (gsmenc, "no data");
goto done;
}
gst_buffer_map (buffer, &map, GST_MAP_READ);
if (G_UNLIKELY (map.size < 320)) {
GST_DEBUG_OBJECT (gsmenc, "discarding trailing data %d", (gint) map.size);
gst_buffer_unmap (buffer, &map);
ret = gst_audio_encoder_finish_frame (benc, NULL, -1);
goto done;
}
outbuf = gst_buffer_new_and_alloc (33 * sizeof (gsm_byte));
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
/* encode 160 16-bit samples into 33 bytes */
data = (gsm_signal *) map.data;
gsm_encode (gsmenc->state, data, (gsm_byte *) omap.data);
GST_LOG_OBJECT (gsmenc, "encoded to %d bytes", (gint) omap.size);
gst_buffer_unmap (buffer, &map);
gst_buffer_unmap (buffer, &omap);
ret = gst_audio_encoder_finish_frame (benc, outbuf, 160);
done:
return ret;
}
|
