1 /* GStreamer
2 * Copyright (C) 1999 Erik Walthinsen <omega@cse.ogi.edu>
3 * Copyright (C) 2005 Edgard Lima <edgard.lima@gmail.com>
4 * Copyright (C) 2005 Nokia Corporation <kai.vehmanen@nokia.com>
5 * Copyright (C) 2007,2008 Axis Communications <dev-gstreamer@axis.com>
6 *
7 * This library is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Library General Public
9 * License as published by the Free Software Foundation; either
10 * version 2 of the License, or (at your option) any later version.
11 *
12 * This library is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Library General Public License for more details.
16 *
17 * You should have received a copy of the GNU Library General Public
18 * License along with this library; if not, write to the
19 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
20 * Boston, MA 02110-1301, USA.
21 */
22
23 #ifdef HAVE_CONFIG_H
24 # include "config.h"
25 #endif
26
27 #include <stdlib.h>
28 #include <string.h>
29 #include <gst/rtp/gstrtpbuffer.h>
30
31 #include "gstrtpg726pay.h"
32
33 GST_DEBUG_CATEGORY_STATIC (rtpg726pay_debug);
34 #define GST_CAT_DEFAULT (rtpg726pay_debug)
35
36 #define DEFAULT_FORCE_AAL2 TRUE
37
38 enum
39 {
40 PROP_0,
41 PROP_FORCE_AAL2
42 };
43
44 static GstStaticPadTemplate gst_rtp_g726_pay_sink_template =
45 GST_STATIC_PAD_TEMPLATE ("sink",
46 GST_PAD_SINK,
47 GST_PAD_ALWAYS,
48 GST_STATIC_CAPS ("audio/x-adpcm, "
49 "channels = (int) 1, "
50 "rate = (int) 8000, "
51 "bitrate = (int) { 16000, 24000, 32000, 40000 }, "
52 "layout = (string) \"g726\"")
53 );
54
55 static GstStaticPadTemplate gst_rtp_g726_pay_src_template =
56 GST_STATIC_PAD_TEMPLATE ("src",
57 GST_PAD_SRC,
58 GST_PAD_ALWAYS,
59 GST_STATIC_CAPS ("application/x-rtp, "
60 "media = (string) \"audio\", "
61 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
62 "clock-rate = (int) 8000, "
63 "encoding-name = (string) { \"G726-16\", \"G726-24\", \"G726-32\", \"G726-40\", "
64 " \"AAL2-G726-16\", \"AAL2-G726-24\", \"AAL2-G726-32\", \"AAL2-G726-40\" } ")
65 );
66
67 static void gst_rtp_g726_pay_get_property (GObject * object, guint prop_id,
68 GValue * value, GParamSpec * pspec);
69 static void gst_rtp_g726_pay_set_property (GObject * object, guint prop_id,
70 const GValue * value, GParamSpec * pspec);
71
72 static gboolean gst_rtp_g726_pay_setcaps (GstRTPBasePayload * payload,
73 GstCaps * caps);
74 static GstFlowReturn gst_rtp_g726_pay_handle_buffer (GstRTPBasePayload *
75 payload, GstBuffer * buffer);
76
77 #define gst_rtp_g726_pay_parent_class parent_class
78 G_DEFINE_TYPE (GstRtpG726Pay, gst_rtp_g726_pay,
79 GST_TYPE_RTP_BASE_AUDIO_PAYLOAD);
80
81 static void
gst_rtp_g726_pay_class_init(GstRtpG726PayClass * klass)82 gst_rtp_g726_pay_class_init (GstRtpG726PayClass * klass)
83 {
84 GObjectClass *gobject_class;
85 GstElementClass *gstelement_class;
86 GstRTPBasePayloadClass *gstrtpbasepayload_class;
87
88 gobject_class = (GObjectClass *) klass;
89 gstelement_class = (GstElementClass *) klass;
90 gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
91
92 gobject_class->set_property = gst_rtp_g726_pay_set_property;
93 gobject_class->get_property = gst_rtp_g726_pay_get_property;
94
95 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_FORCE_AAL2,
96 g_param_spec_boolean ("force-aal2", "Force AAL2",
97 "Force AAL2 encoding for compatibility with bad depayloaders",
98 DEFAULT_FORCE_AAL2, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
99
100 gst_element_class_add_static_pad_template (gstelement_class,
101 &gst_rtp_g726_pay_sink_template);
102 gst_element_class_add_static_pad_template (gstelement_class,
103 &gst_rtp_g726_pay_src_template);
104
105 gst_element_class_set_static_metadata (gstelement_class,
106 "RTP G.726 payloader", "Codec/Payloader/Network/RTP",
107 "Payload-encodes G.726 audio into a RTP packet",
108 "Axis Communications <dev-gstreamer@axis.com>");
109
110 gstrtpbasepayload_class->set_caps = gst_rtp_g726_pay_setcaps;
111 gstrtpbasepayload_class->handle_buffer = gst_rtp_g726_pay_handle_buffer;
112
113 GST_DEBUG_CATEGORY_INIT (rtpg726pay_debug, "rtpg726pay", 0,
114 "G.726 RTP Payloader");
115 }
116
117 static void
gst_rtp_g726_pay_init(GstRtpG726Pay * rtpg726pay)118 gst_rtp_g726_pay_init (GstRtpG726Pay * rtpg726pay)
119 {
120 GstRTPBaseAudioPayload *rtpbaseaudiopayload;
121
122 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpg726pay);
123
124 GST_RTP_BASE_PAYLOAD (rtpg726pay)->clock_rate = 8000;
125
126 rtpg726pay->force_aal2 = DEFAULT_FORCE_AAL2;
127
128 /* sample based codec */
129 gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload);
130 }
131
132 static gboolean
gst_rtp_g726_pay_setcaps(GstRTPBasePayload * payload,GstCaps * caps)133 gst_rtp_g726_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
134 {
135 gchar *encoding_name;
136 GstStructure *structure;
137 GstRTPBaseAudioPayload *rtpbaseaudiopayload;
138 GstRtpG726Pay *pay;
139 GstCaps *peercaps;
140 gboolean res;
141
142 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (payload);
143 pay = GST_RTP_G726_PAY (payload);
144
145 structure = gst_caps_get_structure (caps, 0);
146
147 if (!gst_structure_get_int (structure, "bitrate", &pay->bitrate))
148 pay->bitrate = 32000;
149
150 GST_DEBUG_OBJECT (payload, "using bitrate %d", pay->bitrate);
151
152 pay->aal2 = FALSE;
153
154 /* first see what we can do with the bitrate */
155 switch (pay->bitrate) {
156 case 16000:
157 encoding_name = g_strdup ("G726-16");
158 gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload,
159 2);
160 break;
161 case 24000:
162 encoding_name = g_strdup ("G726-24");
163 gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload,
164 3);
165 break;
166 case 32000:
167 encoding_name = g_strdup ("G726-32");
168 gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload,
169 4);
170 break;
171 case 40000:
172 encoding_name = g_strdup ("G726-40");
173 gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload,
174 5);
175 break;
176 default:
177 goto invalid_bitrate;
178 }
179
180 GST_DEBUG_OBJECT (payload, "selected base encoding %s", encoding_name);
181
182 /* now see if we need to produce AAL2 or not */
183 peercaps = gst_pad_peer_query_caps (payload->srcpad, NULL);
184 if (peercaps) {
185 GstCaps *filter, *intersect;
186 gchar *capsstr;
187
188 GST_DEBUG_OBJECT (payload, "have peercaps %" GST_PTR_FORMAT, peercaps);
189
190 capsstr = g_strdup_printf ("application/x-rtp, "
191 "media = (string) \"audio\", "
192 "clock-rate = (int) 8000, "
193 "encoding-name = (string) %s; "
194 "application/x-rtp, "
195 "media = (string) \"audio\", "
196 "clock-rate = (int) 8000, "
197 "encoding-name = (string) AAL2-%s", encoding_name, encoding_name);
198 filter = gst_caps_from_string (capsstr);
199 g_free (capsstr);
200 g_free (encoding_name);
201
202 /* intersect to filter */
203 intersect = gst_caps_intersect (peercaps, filter);
204 gst_caps_unref (peercaps);
205 gst_caps_unref (filter);
206
207 GST_DEBUG_OBJECT (payload, "intersected to %" GST_PTR_FORMAT, intersect);
208
209 if (!intersect)
210 goto no_format;
211 if (gst_caps_is_empty (intersect)) {
212 gst_caps_unref (intersect);
213 goto no_format;
214 }
215
216 structure = gst_caps_get_structure (intersect, 0);
217
218 /* now see what encoding name we settled on, we need to dup because the
219 * string goes away when we unref the intersection below. */
220 encoding_name =
221 g_strdup (gst_structure_get_string (structure, "encoding-name"));
222
223 /* if we managed to negotiate to AAL2, we definatly are going to do AAL2
224 * encoding. Else we only encode AAL2 when explicitly set by the
225 * property. */
226 if (g_str_has_prefix (encoding_name, "AAL2-"))
227 pay->aal2 = TRUE;
228 else
229 pay->aal2 = pay->force_aal2;
230
231 GST_DEBUG_OBJECT (payload, "final encoding %s, AAL2 %d", encoding_name,
232 pay->aal2);
233
234 gst_caps_unref (intersect);
235 } else {
236 /* downstream can do anything but we prefer the better supported non-AAL2 */
237 pay->aal2 = pay->force_aal2;
238 GST_DEBUG_OBJECT (payload, "no peer caps, AAL2 %d", pay->aal2);
239 }
240
241 gst_rtp_base_payload_set_options (payload, "audio", TRUE, encoding_name,
242 8000);
243 res = gst_rtp_base_payload_set_outcaps (payload, NULL);
244
245 g_free (encoding_name);
246
247 return res;
248
249 /* ERRORS */
250 invalid_bitrate:
251 {
252 GST_ERROR_OBJECT (payload, "invalid bitrate %d specified", pay->bitrate);
253 return FALSE;
254 }
255 no_format:
256 {
257 GST_ERROR_OBJECT (payload, "could not negotiate format");
258 return FALSE;
259 }
260 }
261
262 static GstFlowReturn
gst_rtp_g726_pay_handle_buffer(GstRTPBasePayload * payload,GstBuffer * buffer)263 gst_rtp_g726_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buffer)
264 {
265 GstFlowReturn res;
266 GstRtpG726Pay *pay;
267
268 pay = GST_RTP_G726_PAY (payload);
269
270 if (!pay->aal2) {
271 GstMapInfo map;
272 guint8 *data, tmp;
273 gsize size;
274
275 /* for non AAL2, we need to reshuffle the bytes, we can do this in-place
276 * when the buffer is writable. */
277 buffer = gst_buffer_make_writable (buffer);
278
279 gst_buffer_map (buffer, &map, GST_MAP_READWRITE);
280 data = map.data;
281 size = map.size;
282
283 GST_LOG_OBJECT (pay, "packing %" G_GSIZE_FORMAT " bytes of data", map.size);
284
285 /* we need to reshuffle the bytes, output is of the form:
286 * A B C D .. with the number of bits depending on the bitrate. */
287 switch (pay->bitrate) {
288 case 16000:
289 {
290 /* 0
291 * 0 1 2 3 4 5 6 7
292 * +-+-+-+-+-+-+-+-+-
293 * |D D|C C|B B|A A| ...
294 * |0 1|0 1|0 1|0 1|
295 * +-+-+-+-+-+-+-+-+-
296 */
297 while (size > 0) {
298 tmp = *data;
299 *data++ = ((tmp & 0xc0) >> 6) |
300 ((tmp & 0x30) >> 2) | ((tmp & 0x0c) << 2) | ((tmp & 0x03) << 6);
301 size--;
302 }
303 break;
304 }
305 case 24000:
306 {
307 /* 0 1 2
308 * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3
309 * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
310 * |C C|B B B|A A A|F|E E E|D D D|C|H H H|G G G|F F| ...
311 * |1 2|0 1 2|0 1 2|2|0 1 2|0 1 2|0|0 1 2|0 1 2|0 1|
312 * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
313 */
314 while (size > 2) {
315 tmp = *data;
316 *data++ = ((tmp & 0xc0) >> 6) |
317 ((tmp & 0x38) >> 1) | ((tmp & 0x07) << 5);
318 tmp = *data;
319 *data++ = ((tmp & 0x80) >> 7) |
320 ((tmp & 0x70) >> 3) | ((tmp & 0x0e) << 4) | ((tmp & 0x01) << 7);
321 tmp = *data;
322 *data++ = ((tmp & 0xe0) >> 5) |
323 ((tmp & 0x1c) >> 2) | ((tmp & 0x03) << 6);
324 size -= 3;
325 }
326 break;
327 }
328 case 32000:
329 {
330 /* 0 1
331 * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
332 * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
333 * |B B B B|A A A A|D D D D|C C C C| ...
334 * |0 1 2 3|0 1 2 3|0 1 2 3|0 1 2 3|
335 * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
336 */
337 while (size > 0) {
338 tmp = *data;
339 *data++ = ((tmp & 0xf0) >> 4) | ((tmp & 0x0f) << 4);
340 size--;
341 }
342 break;
343 }
344 case 40000:
345 {
346 /* 0 1 2 3 4
347 * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0
348 * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
349 * |B B B|A A A A A|D|C C C C C|B B|E E E E|D D D D|G G|F F F F F|E|H H H H H|G G G|
350 * |2 3 4|0 1 2 3 4|4|0 1 2 3 4|0 1|1 2 3 4|0 1 2 3|3 4|0 1 2 3 4|0|0 1 2 3 4|0 1 2|
351 * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
352 */
353 while (size > 4) {
354 tmp = *data;
355 *data++ = ((tmp & 0xe0) >> 5) | ((tmp & 0x1f) << 3);
356 tmp = *data;
357 *data++ = ((tmp & 0x80) >> 7) |
358 ((tmp & 0x7c) >> 2) | ((tmp & 0x03) << 6);
359 tmp = *data;
360 *data++ = ((tmp & 0xf0) >> 4) | ((tmp & 0x0f) << 4);
361 tmp = *data;
362 *data++ = ((tmp & 0xc0) >> 6) |
363 ((tmp & 0x3e) << 2) | ((tmp & 0x01) << 7);
364 tmp = *data;
365 *data++ = ((tmp & 0xf8) >> 3) | ((tmp & 0x07) << 5);
366 size -= 5;
367 }
368 break;
369 }
370 }
371 gst_buffer_unmap (buffer, &map);
372 }
373
374 res =
375 GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->handle_buffer (payload,
376 buffer);
377
378 return res;
379 }
380
381 static void
gst_rtp_g726_pay_set_property(GObject * object,guint prop_id,const GValue * value,GParamSpec * pspec)382 gst_rtp_g726_pay_set_property (GObject * object, guint prop_id,
383 const GValue * value, GParamSpec * pspec)
384 {
385 GstRtpG726Pay *rtpg726pay;
386
387 rtpg726pay = GST_RTP_G726_PAY (object);
388
389 switch (prop_id) {
390 case PROP_FORCE_AAL2:
391 rtpg726pay->force_aal2 = g_value_get_boolean (value);
392 break;
393 default:
394 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
395 break;
396 }
397 }
398
399 static void
gst_rtp_g726_pay_get_property(GObject * object,guint prop_id,GValue * value,GParamSpec * pspec)400 gst_rtp_g726_pay_get_property (GObject * object, guint prop_id,
401 GValue * value, GParamSpec * pspec)
402 {
403 GstRtpG726Pay *rtpg726pay;
404
405 rtpg726pay = GST_RTP_G726_PAY (object);
406
407 switch (prop_id) {
408 case PROP_FORCE_AAL2:
409 g_value_set_boolean (value, rtpg726pay->force_aal2);
410 break;
411 default:
412 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
413 break;
414 }
415 }
416
417 gboolean
gst_rtp_g726_pay_plugin_init(GstPlugin * plugin)418 gst_rtp_g726_pay_plugin_init (GstPlugin * plugin)
419 {
420 return gst_element_register (plugin, "rtpg726pay",
421 GST_RANK_SECONDARY, GST_TYPE_RTP_G726_PAY);
422 }
423