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 GST_ELEMENT_REGISTER_DEFINE (sirenenc, "sirenenc",
66 GST_RANK_MARGINAL, GST_TYPE_SIREN_ENC);
67
68 static void
gst_siren_enc_class_init(GstSirenEncClass * klass)69 gst_siren_enc_class_init (GstSirenEncClass * klass)
70 {
71 GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
72 GstAudioEncoderClass *base_class = GST_AUDIO_ENCODER_CLASS (klass);
73
74 GST_DEBUG_CATEGORY_INIT (sirenenc_debug, "sirenenc", 0, "sirenenc");
75
76 gst_element_class_add_static_pad_template (element_class, &srctemplate);
77 gst_element_class_add_static_pad_template (element_class, &sinktemplate);
78
79 gst_element_class_set_static_metadata (element_class, "Siren Encoder element",
80 "Codec/Encoder/Audio ",
81 "Encode 16bit PCM streams into the Siren7 codec",
82 "Youness Alaoui <kakaroto@kakaroto.homelinux.net>");
83
84 base_class->start = GST_DEBUG_FUNCPTR (gst_siren_enc_start);
85 base_class->stop = GST_DEBUG_FUNCPTR (gst_siren_enc_stop);
86 base_class->set_format = GST_DEBUG_FUNCPTR (gst_siren_enc_set_format);
87 base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_siren_enc_handle_frame);
88
89 GST_DEBUG ("Class Init done");
90 }
91
92 static void
gst_siren_enc_init(GstSirenEnc * enc)93 gst_siren_enc_init (GstSirenEnc * enc)
94 {
95 GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (enc));
96 }
97
98 static gboolean
gst_siren_enc_start(GstAudioEncoder * enc)99 gst_siren_enc_start (GstAudioEncoder * enc)
100 {
101 GstSirenEnc *senc = GST_SIREN_ENC (enc);
102
103 GST_DEBUG_OBJECT (enc, "start");
104
105 senc->encoder = Siren7_NewEncoder (16000);
106
107 return TRUE;
108 }
109
110 static gboolean
gst_siren_enc_stop(GstAudioEncoder * enc)111 gst_siren_enc_stop (GstAudioEncoder * enc)
112 {
113 GstSirenEnc *senc = GST_SIREN_ENC (enc);
114
115 GST_DEBUG_OBJECT (senc, "stop");
116
117 Siren7_CloseEncoder (senc->encoder);
118
119 return TRUE;
120 }
121
122 static gboolean
gst_siren_enc_set_format(GstAudioEncoder * benc,GstAudioInfo * info)123 gst_siren_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
124 {
125 gboolean res;
126 GstCaps *outcaps;
127
128 outcaps = gst_static_pad_template_get_caps (&srctemplate);
129 res = gst_audio_encoder_set_output_format (benc, outcaps);
130 gst_caps_unref (outcaps);
131
132 /* report needs to base class */
133 gst_audio_encoder_set_frame_samples_min (benc, 320);
134 gst_audio_encoder_set_frame_samples_max (benc, 320);
135 /* no remainder or flushing please */
136 gst_audio_encoder_set_hard_min (benc, TRUE);
137 gst_audio_encoder_set_drainable (benc, FALSE);
138
139 return res;
140 }
141
142 static GstFlowReturn
gst_siren_enc_handle_frame(GstAudioEncoder * benc,GstBuffer * buf)143 gst_siren_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
144 {
145 GstSirenEnc *enc;
146 GstFlowReturn ret = GST_FLOW_OK;
147 GstBuffer *out_buf;
148 guint8 *in_data, *out_data;
149 guint i, size, num_frames;
150 gint out_size;
151 #ifndef GST_DISABLE_GST_DEBUG
152 gint in_size;
153 #endif
154 gint encode_ret;
155 GstMapInfo inmap, outmap;
156
157 g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR);
158
159 enc = GST_SIREN_ENC (benc);
160
161 size = gst_buffer_get_size (buf);
162
163 GST_LOG_OBJECT (enc, "Received buffer of size %d", size);
164
165 g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
166 g_return_val_if_fail (size % 640 == 0, GST_FLOW_ERROR);
167
168 /* we need to process 640 input bytes to produce 40 output bytes */
169 /* calculate the amount of frames we will handle */
170 num_frames = size / 640;
171
172 /* this is the input/output size */
173 #ifndef GST_DISABLE_GST_DEBUG
174 in_size = num_frames * 640;
175 #endif
176 out_size = num_frames * 40;
177
178 GST_LOG_OBJECT (enc, "we have %u frames, %u in, %u out", num_frames, in_size,
179 out_size);
180
181 /* get a buffer */
182 out_buf = gst_audio_encoder_allocate_output_buffer (benc, out_size);
183 if (out_buf == NULL)
184 goto alloc_failed;
185
186 /* get the input data for all the frames */
187 gst_buffer_map (buf, &inmap, GST_MAP_READ);
188 gst_buffer_map (out_buf, &outmap, GST_MAP_READ);
189 in_data = inmap.data;
190 out_data = outmap.data;
191
192 for (i = 0; i < num_frames; i++) {
193 GST_LOG_OBJECT (enc, "Encoding frame %u/%u", i, num_frames);
194
195 /* encode 640 input bytes to 40 output bytes */
196 encode_ret = Siren7_EncodeFrame (enc->encoder, in_data, out_data);
197 if (encode_ret != 0)
198 goto encode_error;
199
200 /* move to next frame */
201 out_data += 40;
202 in_data += 640;
203 }
204
205 gst_buffer_unmap (buf, &inmap);
206 gst_buffer_unmap (out_buf, &outmap);
207
208 GST_LOG_OBJECT (enc, "Finished encoding");
209
210 /* we encode all we get, pass it along */
211 ret = gst_audio_encoder_finish_frame (benc, out_buf, -1);
212
213 done:
214 return ret;
215
216 /* ERRORS */
217 alloc_failed:
218 {
219 GST_DEBUG_OBJECT (enc, "failed to pad_alloc buffer: %d (%s)", ret,
220 gst_flow_get_name (ret));
221 goto done;
222 }
223 encode_error:
224 {
225 GST_ELEMENT_ERROR (enc, STREAM, ENCODE, (NULL),
226 ("Error encoding frame: %d", encode_ret));
227 ret = GST_FLOW_ERROR;
228 gst_buffer_unref (out_buf);
229 goto done;
230 }
231 }
232