1 /*
2 * Siren Encoder Gst Element
3 *
4 * @author: Youness Alaoui <kakaroto@kakaroto.homelinux.net>
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 * SECTION:element-sirenenc
24 * @title: sirenenc
25 *
26 * This encodes audio buffers into the Siren 16 codec (a 16khz extension of
27 * G.722.1) that is meant to be compatible with the Microsoft Windows Live
28 * Messenger(tm) implementation.
29 *
30 * Ref: http://www.polycom.com/company/about_us/technology/siren_g7221/index.html
31 */
32
33 #ifdef HAVE_CONFIG_H
34 #include "config.h"
35 #endif
36
37 #include "gstsirenenc.h"
38
39 #include <string.h>
40
41 GST_DEBUG_CATEGORY (sirenenc_debug);
42 #define GST_CAT_DEFAULT (sirenenc_debug)
43
44 #define FRAME_DURATION (20 * GST_MSECOND)
45
46 static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
47 GST_PAD_SRC,
48 GST_PAD_ALWAYS,
49 GST_STATIC_CAPS ("audio/x-siren, " "dct-length = (int) 320"));
50
51 static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
52 GST_PAD_SINK,
53 GST_PAD_ALWAYS,
54 GST_STATIC_CAPS ("audio/x-raw, format = (string) \"S16LE\", "
55 "rate = (int) 16000, " "channels = (int) 1"));
56
57 static gboolean gst_siren_enc_start (GstAudioEncoder * enc);
58 static gboolean gst_siren_enc_stop (GstAudioEncoder * enc);
59 static gboolean gst_siren_enc_set_format (GstAudioEncoder * enc,
60 GstAudioInfo * info);
61 static GstFlowReturn gst_siren_enc_handle_frame (GstAudioEncoder * enc,
62 GstBuffer * in_buf);
63
64 G_DEFINE_TYPE (GstSirenEnc, gst_siren_enc, GST_TYPE_AUDIO_ENCODER);
65
66
67 static void
gst_siren_enc_class_init(GstSirenEncClass * klass)68 gst_siren_enc_class_init (GstSirenEncClass * klass)
69 {
70 GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
71 GstAudioEncoderClass *base_class = GST_AUDIO_ENCODER_CLASS (klass);
72
73 GST_DEBUG_CATEGORY_INIT (sirenenc_debug, "sirenenc", 0, "sirenenc");
74
75 gst_element_class_add_static_pad_template (element_class, &srctemplate);
76 gst_element_class_add_static_pad_template (element_class, &sinktemplate);
77
78 gst_element_class_set_static_metadata (element_class, "Siren Encoder element",
79 "Codec/Encoder/Audio ",
80 "Encode 16bit PCM streams into the Siren7 codec",
81 "Youness Alaoui <kakaroto@kakaroto.homelinux.net>");
82
83 base_class->start = GST_DEBUG_FUNCPTR (gst_siren_enc_start);
84 base_class->stop = GST_DEBUG_FUNCPTR (gst_siren_enc_stop);
85 base_class->set_format = GST_DEBUG_FUNCPTR (gst_siren_enc_set_format);
86 base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_siren_enc_handle_frame);
87
88 GST_DEBUG ("Class Init done");
89 }
90
91 static void
gst_siren_enc_init(GstSirenEnc * enc)92 gst_siren_enc_init (GstSirenEnc * enc)
93 {
94 GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (enc));
95 }
96
97 static gboolean
gst_siren_enc_start(GstAudioEncoder * enc)98 gst_siren_enc_start (GstAudioEncoder * enc)
99 {
100 GstSirenEnc *senc = GST_SIREN_ENC (enc);
101
102 GST_DEBUG_OBJECT (enc, "start");
103
104 senc->encoder = Siren7_NewEncoder (16000);
105
106 return TRUE;
107 }
108
109 static gboolean
gst_siren_enc_stop(GstAudioEncoder * enc)110 gst_siren_enc_stop (GstAudioEncoder * enc)
111 {
112 GstSirenEnc *senc = GST_SIREN_ENC (enc);
113
114 GST_DEBUG_OBJECT (senc, "stop");
115
116 Siren7_CloseEncoder (senc->encoder);
117
118 return TRUE;
119 }
120
121 static gboolean
gst_siren_enc_set_format(GstAudioEncoder * benc,GstAudioInfo * info)122 gst_siren_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
123 {
124 gboolean res;
125 GstCaps *outcaps;
126
127 outcaps = gst_static_pad_template_get_caps (&srctemplate);
128 res = gst_audio_encoder_set_output_format (benc, outcaps);
129 gst_caps_unref (outcaps);
130
131 /* report needs to base class */
132 gst_audio_encoder_set_frame_samples_min (benc, 320);
133 gst_audio_encoder_set_frame_samples_max (benc, 320);
134 /* no remainder or flushing please */
135 gst_audio_encoder_set_hard_min (benc, TRUE);
136 gst_audio_encoder_set_drainable (benc, FALSE);
137
138 return res;
139 }
140
141 static GstFlowReturn
gst_siren_enc_handle_frame(GstAudioEncoder * benc,GstBuffer * buf)142 gst_siren_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
143 {
144 GstSirenEnc *enc;
145 GstFlowReturn ret = GST_FLOW_OK;
146 GstBuffer *out_buf;
147 guint8 *in_data, *out_data;
148 guint i, size, num_frames;
149 gint out_size;
150 #ifndef GST_DISABLE_GST_DEBUG
151 gint in_size;
152 #endif
153 gint encode_ret;
154 GstMapInfo inmap, outmap;
155
156 g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR);
157
158 enc = GST_SIREN_ENC (benc);
159
160 size = gst_buffer_get_size (buf);
161
162 GST_LOG_OBJECT (enc, "Received buffer of size %d", size);
163
164 g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
165 g_return_val_if_fail (size % 640 == 0, GST_FLOW_ERROR);
166
167 /* we need to process 640 input bytes to produce 40 output bytes */
168 /* calculate the amount of frames we will handle */
169 num_frames = size / 640;
170
171 /* this is the input/output size */
172 #ifndef GST_DISABLE_GST_DEBUG
173 in_size = num_frames * 640;
174 #endif
175 out_size = num_frames * 40;
176
177 GST_LOG_OBJECT (enc, "we have %u frames, %u in, %u out", num_frames, in_size,
178 out_size);
179
180 /* get a buffer */
181 out_buf = gst_audio_encoder_allocate_output_buffer (benc, out_size);
182 if (out_buf == NULL)
183 goto alloc_failed;
184
185 /* get the input data for all the frames */
186 gst_buffer_map (buf, &inmap, GST_MAP_READ);
187 gst_buffer_map (out_buf, &outmap, GST_MAP_READ);
188 in_data = inmap.data;
189 out_data = outmap.data;
190
191 for (i = 0; i < num_frames; i++) {
192 GST_LOG_OBJECT (enc, "Encoding frame %u/%u", i, num_frames);
193
194 /* encode 640 input bytes to 40 output bytes */
195 encode_ret = Siren7_EncodeFrame (enc->encoder, in_data, out_data);
196 if (encode_ret != 0)
197 goto encode_error;
198
199 /* move to next frame */
200 out_data += 40;
201 in_data += 640;
202 }
203
204 gst_buffer_unmap (buf, &inmap);
205 gst_buffer_unmap (out_buf, &outmap);
206
207 GST_LOG_OBJECT (enc, "Finished encoding");
208
209 /* we encode all we get, pass it along */
210 ret = gst_audio_encoder_finish_frame (benc, out_buf, -1);
211
212 done:
213 return ret;
214
215 /* ERRORS */
216 alloc_failed:
217 {
218 GST_DEBUG_OBJECT (enc, "failed to pad_alloc buffer: %d (%s)", ret,
219 gst_flow_get_name (ret));
220 goto done;
221 }
222 encode_error:
223 {
224 GST_ELEMENT_ERROR (enc, STREAM, ENCODE, (NULL),
225 ("Error encoding frame: %d", encode_ret));
226 ret = GST_FLOW_ERROR;
227 gst_buffer_unref (out_buf);
228 goto done;
229 }
230 }
231
232 gboolean
gst_siren_enc_plugin_init(GstPlugin * plugin)233 gst_siren_enc_plugin_init (GstPlugin * plugin)
234 {
235 return gst_element_register (plugin, "sirenenc",
236 GST_RANK_MARGINAL, GST_TYPE_SIREN_ENC);
237 }
238