1 /*
2 * Farsight
3 * GStreamer GSM encoder
4 * Copyright (C) 2005 Philippe Khalaf <burger@speedy.org>
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
23 #ifdef HAVE_CONFIG_H
24 #include "config.h"
25 #endif
26 #include <string.h>
27
28 #include "gstgsmenc.h"
29
30 GST_DEBUG_CATEGORY_STATIC (gsmenc_debug);
31 #define GST_CAT_DEFAULT (gsmenc_debug)
32
33 /* GSMEnc signals and args */
34 enum
35 {
36 /* FILL ME */
37 LAST_SIGNAL
38 };
39
40 enum
41 {
42 /* FILL ME */
43 ARG_0
44 };
45
46 static gboolean gst_gsmenc_start (GstAudioEncoder * enc);
47 static gboolean gst_gsmenc_stop (GstAudioEncoder * enc);
48 static gboolean gst_gsmenc_set_format (GstAudioEncoder * enc,
49 GstAudioInfo * info);
50 static GstFlowReturn gst_gsmenc_handle_frame (GstAudioEncoder * enc,
51 GstBuffer * in_buf);
52
53 static GstStaticPadTemplate gsmenc_src_template =
54 GST_STATIC_PAD_TEMPLATE ("src",
55 GST_PAD_SRC,
56 GST_PAD_ALWAYS,
57 GST_STATIC_CAPS ("audio/x-gsm, " "rate = (int) 8000, " "channels = (int) 1")
58 );
59
60 static GstStaticPadTemplate gsmenc_sink_template =
61 GST_STATIC_PAD_TEMPLATE ("sink",
62 GST_PAD_SINK,
63 GST_PAD_ALWAYS,
64 GST_STATIC_CAPS ("audio/x-raw, "
65 "format = (string) " GST_AUDIO_NE (S16) ", "
66 "layout = (string) interleaved, "
67 "rate = (int) 8000, channels = (int) 1")
68 );
69
70 G_DEFINE_TYPE (GstGSMEnc, gst_gsmenc, GST_TYPE_AUDIO_ENCODER);
71
72 static void
gst_gsmenc_class_init(GstGSMEncClass * klass)73 gst_gsmenc_class_init (GstGSMEncClass * klass)
74 {
75 GstElementClass *element_class;
76 GstAudioEncoderClass *base_class;
77
78 element_class = (GstElementClass *) klass;
79 base_class = (GstAudioEncoderClass *) klass;
80
81 gst_element_class_add_static_pad_template (element_class,
82 &gsmenc_sink_template);
83 gst_element_class_add_static_pad_template (element_class,
84 &gsmenc_src_template);
85 gst_element_class_set_static_metadata (element_class, "GSM audio encoder",
86 "Codec/Encoder/Audio", "Encodes GSM audio",
87 "Philippe Khalaf <burger@speedy.org>");
88
89 base_class->start = GST_DEBUG_FUNCPTR (gst_gsmenc_start);
90 base_class->stop = GST_DEBUG_FUNCPTR (gst_gsmenc_stop);
91 base_class->set_format = GST_DEBUG_FUNCPTR (gst_gsmenc_set_format);
92 base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_gsmenc_handle_frame);
93
94 GST_DEBUG_CATEGORY_INIT (gsmenc_debug, "gsmenc", 0, "GSM Encoder");
95 }
96
97 static void
gst_gsmenc_init(GstGSMEnc * gsmenc)98 gst_gsmenc_init (GstGSMEnc * gsmenc)
99 {
100 GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (gsmenc));
101 }
102
103 static gboolean
gst_gsmenc_start(GstAudioEncoder * enc)104 gst_gsmenc_start (GstAudioEncoder * enc)
105 {
106 GstGSMEnc *gsmenc = GST_GSMENC (enc);
107 gint use_wav49;
108
109 GST_DEBUG_OBJECT (enc, "start");
110
111 gsmenc->state = gsm_create ();
112
113 /* turn off WAV49 handling */
114 use_wav49 = 0;
115 gsm_option (gsmenc->state, GSM_OPT_WAV49, &use_wav49);
116
117 return TRUE;
118 }
119
120 static gboolean
gst_gsmenc_stop(GstAudioEncoder * enc)121 gst_gsmenc_stop (GstAudioEncoder * enc)
122 {
123 GstGSMEnc *gsmenc = GST_GSMENC (enc);
124
125 GST_DEBUG_OBJECT (enc, "stop");
126 gsm_destroy (gsmenc->state);
127
128 return TRUE;
129 }
130
131 static gboolean
gst_gsmenc_set_format(GstAudioEncoder * benc,GstAudioInfo * info)132 gst_gsmenc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
133 {
134 GstCaps *srccaps;
135
136 srccaps = gst_static_pad_template_get_caps (&gsmenc_src_template);
137 gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (benc), srccaps);
138 gst_caps_unref (srccaps);
139
140 /* report needs to base class */
141 gst_audio_encoder_set_frame_samples_min (benc, 160);
142 gst_audio_encoder_set_frame_samples_max (benc, 160);
143 gst_audio_encoder_set_frame_max (benc, 1);
144
145 return TRUE;
146 }
147
148 static GstFlowReturn
gst_gsmenc_handle_frame(GstAudioEncoder * benc,GstBuffer * buffer)149 gst_gsmenc_handle_frame (GstAudioEncoder * benc, GstBuffer * buffer)
150 {
151 GstGSMEnc *gsmenc;
152 gsm_signal *data;
153 GstFlowReturn ret = GST_FLOW_OK;
154 GstBuffer *outbuf;
155 GstMapInfo map, omap;
156
157 gsmenc = GST_GSMENC (benc);
158
159 /* we don't deal with squeezing remnants, so simply discard those */
160 if (G_UNLIKELY (buffer == NULL)) {
161 GST_DEBUG_OBJECT (gsmenc, "no data");
162 goto done;
163 }
164
165 gst_buffer_map (buffer, &map, GST_MAP_READ);
166 if (G_UNLIKELY (map.size < 320)) {
167 GST_DEBUG_OBJECT (gsmenc, "discarding trailing data %d", (gint) map.size);
168 gst_buffer_unmap (buffer, &map);
169 ret = gst_audio_encoder_finish_frame (benc, NULL, -1);
170 goto done;
171 }
172
173 outbuf = gst_buffer_new_and_alloc (33 * sizeof (gsm_byte));
174 gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
175
176 /* encode 160 16-bit samples into 33 bytes */
177 data = (gsm_signal *) map.data;
178 gsm_encode (gsmenc->state, data, (gsm_byte *) omap.data);
179
180 GST_LOG_OBJECT (gsmenc, "encoded to %d bytes", (gint) omap.size);
181 gst_buffer_unmap (buffer, &map);
182 gst_buffer_unmap (outbuf, &omap);
183
184 ret = gst_audio_encoder_finish_frame (benc, outbuf, 160);
185
186 done:
187 return ret;
188 }
189