1 /* GStreamer
2 * Copyright (C) <2008> Wim Taymans <wim.taymans@gmail.com>
3 *
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
8 *
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
13 *
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
17 * Boston, MA 02110-1301, USA.
18 */
19
20 #ifdef HAVE_CONFIG_H
21 # include "config.h"
22 #endif
23
24 #include <string.h>
25
26 #include <gst/rtp/gstrtpbuffer.h>
27 #include <gst/audio/audio.h>
28
29 #include "gstrtpmp4apay.h"
30 #include "gstrtputils.h"
31
32 GST_DEBUG_CATEGORY_STATIC (rtpmp4apay_debug);
33 #define GST_CAT_DEFAULT (rtpmp4apay_debug)
34
35 /* FIXME: add framed=(boolean)true once our encoders have this field set
36 * on their output caps */
37 static GstStaticPadTemplate gst_rtp_mp4a_pay_sink_template =
38 GST_STATIC_PAD_TEMPLATE ("sink",
39 GST_PAD_SINK,
40 GST_PAD_ALWAYS,
41 GST_STATIC_CAPS ("audio/mpeg, mpegversion=(int)4, "
42 "stream-format=(string)raw")
43 );
44
45 static GstStaticPadTemplate gst_rtp_mp4a_pay_src_template =
46 GST_STATIC_PAD_TEMPLATE ("src",
47 GST_PAD_SRC,
48 GST_PAD_ALWAYS,
49 GST_STATIC_CAPS ("application/x-rtp, "
50 "media = (string) \"audio\", "
51 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
52 "clock-rate = (int) [1, MAX ], "
53 "encoding-name = (string) \"MP4A-LATM\""
54 /* All optional parameters
55 *
56 * "cpresent = (string) \"0\""
57 * "config="
58 */
59 )
60 );
61
62 static void gst_rtp_mp4a_pay_finalize (GObject * object);
63
64 static gboolean gst_rtp_mp4a_pay_setcaps (GstRTPBasePayload * payload,
65 GstCaps * caps);
66 static GstFlowReturn gst_rtp_mp4a_pay_handle_buffer (GstRTPBasePayload *
67 payload, GstBuffer * buffer);
68
69 #define gst_rtp_mp4a_pay_parent_class parent_class
70 G_DEFINE_TYPE (GstRtpMP4APay, gst_rtp_mp4a_pay, GST_TYPE_RTP_BASE_PAYLOAD);
71
72 static void
gst_rtp_mp4a_pay_class_init(GstRtpMP4APayClass * klass)73 gst_rtp_mp4a_pay_class_init (GstRtpMP4APayClass * klass)
74 {
75 GObjectClass *gobject_class;
76 GstElementClass *gstelement_class;
77 GstRTPBasePayloadClass *gstrtpbasepayload_class;
78
79 gobject_class = (GObjectClass *) klass;
80 gstelement_class = (GstElementClass *) klass;
81 gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
82
83 gobject_class->finalize = gst_rtp_mp4a_pay_finalize;
84
85 gstrtpbasepayload_class->set_caps = gst_rtp_mp4a_pay_setcaps;
86 gstrtpbasepayload_class->handle_buffer = gst_rtp_mp4a_pay_handle_buffer;
87
88 gst_element_class_add_static_pad_template (gstelement_class,
89 &gst_rtp_mp4a_pay_src_template);
90 gst_element_class_add_static_pad_template (gstelement_class,
91 &gst_rtp_mp4a_pay_sink_template);
92
93 gst_element_class_set_static_metadata (gstelement_class,
94 "RTP MPEG4 audio payloader", "Codec/Payloader/Network/RTP",
95 "Payload MPEG4 audio as RTP packets (RFC 3016)",
96 "Wim Taymans <wim.taymans@gmail.com>");
97
98 GST_DEBUG_CATEGORY_INIT (rtpmp4apay_debug, "rtpmp4apay", 0,
99 "MP4A-LATM RTP Payloader");
100 }
101
102 static void
gst_rtp_mp4a_pay_init(GstRtpMP4APay * rtpmp4apay)103 gst_rtp_mp4a_pay_init (GstRtpMP4APay * rtpmp4apay)
104 {
105 rtpmp4apay->rate = 90000;
106 rtpmp4apay->profile = g_strdup ("1");
107 }
108
109 static void
gst_rtp_mp4a_pay_finalize(GObject * object)110 gst_rtp_mp4a_pay_finalize (GObject * object)
111 {
112 GstRtpMP4APay *rtpmp4apay;
113
114 rtpmp4apay = GST_RTP_MP4A_PAY (object);
115
116 g_free (rtpmp4apay->params);
117 rtpmp4apay->params = NULL;
118
119 if (rtpmp4apay->config)
120 gst_buffer_unref (rtpmp4apay->config);
121 rtpmp4apay->config = NULL;
122
123 g_free (rtpmp4apay->profile);
124 rtpmp4apay->profile = NULL;
125
126 G_OBJECT_CLASS (parent_class)->finalize (object);
127 }
128
129 static const unsigned int sampling_table[16] = {
130 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
131 16000, 12000, 11025, 8000, 7350, 0, 0, 0
132 };
133
134 static gboolean
gst_rtp_mp4a_pay_parse_audio_config(GstRtpMP4APay * rtpmp4apay,GstBuffer * buffer)135 gst_rtp_mp4a_pay_parse_audio_config (GstRtpMP4APay * rtpmp4apay,
136 GstBuffer * buffer)
137 {
138 GstMapInfo map;
139 guint8 *data;
140 gsize size;
141 guint8 objectType;
142 guint8 samplingIdx;
143 guint8 channelCfg;
144
145 gst_buffer_map (buffer, &map, GST_MAP_READ);
146 data = map.data;
147 size = map.size;
148
149 if (size < 2)
150 goto too_short;
151
152 /* any object type is fine, we need to copy it to the profile-level-id field. */
153 objectType = (data[0] & 0xf8) >> 3;
154 if (objectType == 0)
155 goto invalid_object;
156
157 samplingIdx = ((data[0] & 0x07) << 1) | ((data[1] & 0x80) >> 7);
158 /* only fixed values for now */
159 if (samplingIdx > 12 && samplingIdx != 15)
160 goto wrong_freq;
161
162 channelCfg = ((data[1] & 0x78) >> 3);
163 if (channelCfg > 7)
164 goto wrong_channels;
165
166 /* rtp rate depends on sampling rate of the audio */
167 if (samplingIdx == 15) {
168 if (size < 5)
169 goto too_short;
170
171 /* index of 15 means we get the rate in the next 24 bits */
172 rtpmp4apay->rate = ((data[1] & 0x7f) << 17) |
173 ((data[2]) << 9) | ((data[3]) << 1) | ((data[4] & 0x80) >> 7);
174 } else {
175 /* else use the rate from the table */
176 rtpmp4apay->rate = sampling_table[samplingIdx];
177 }
178 /* extra rtp params contain the number of channels */
179 g_free (rtpmp4apay->params);
180 rtpmp4apay->params = g_strdup_printf ("%d", channelCfg);
181 /* audio stream type */
182 rtpmp4apay->streamtype = "5";
183 /* profile */
184 g_free (rtpmp4apay->profile);
185 rtpmp4apay->profile = g_strdup_printf ("%d", objectType);
186
187 GST_DEBUG_OBJECT (rtpmp4apay,
188 "objectType: %d, samplingIdx: %d (%d), channelCfg: %d", objectType,
189 samplingIdx, rtpmp4apay->rate, channelCfg);
190
191 gst_buffer_unmap (buffer, &map);
192
193 return TRUE;
194
195 /* ERROR */
196 too_short:
197 {
198 GST_ELEMENT_ERROR (rtpmp4apay, STREAM, FORMAT,
199 (NULL),
200 ("config string too short, expected 2 bytes, got %" G_GSIZE_FORMAT,
201 size));
202 gst_buffer_unmap (buffer, &map);
203 return FALSE;
204 }
205 invalid_object:
206 {
207 GST_ELEMENT_ERROR (rtpmp4apay, STREAM, FORMAT,
208 (NULL), ("invalid object type 0"));
209 gst_buffer_unmap (buffer, &map);
210 return FALSE;
211 }
212 wrong_freq:
213 {
214 GST_ELEMENT_ERROR (rtpmp4apay, STREAM, NOT_IMPLEMENTED,
215 (NULL), ("unsupported frequency index %d", samplingIdx));
216 gst_buffer_unmap (buffer, &map);
217 return FALSE;
218 }
219 wrong_channels:
220 {
221 GST_ELEMENT_ERROR (rtpmp4apay, STREAM, NOT_IMPLEMENTED,
222 (NULL), ("unsupported number of channels %d, must < 8", channelCfg));
223 gst_buffer_unmap (buffer, &map);
224 return FALSE;
225 }
226 }
227
228 static gboolean
gst_rtp_mp4a_pay_new_caps(GstRtpMP4APay * rtpmp4apay)229 gst_rtp_mp4a_pay_new_caps (GstRtpMP4APay * rtpmp4apay)
230 {
231 gchar *config;
232 GValue v = { 0 };
233 gboolean res;
234
235 g_value_init (&v, GST_TYPE_BUFFER);
236 gst_value_set_buffer (&v, rtpmp4apay->config);
237 config = gst_value_serialize (&v);
238
239 res = gst_rtp_base_payload_set_outcaps (GST_RTP_BASE_PAYLOAD (rtpmp4apay),
240 "cpresent", G_TYPE_STRING, "0", "config", G_TYPE_STRING, config, NULL);
241
242 g_value_unset (&v);
243 g_free (config);
244
245 return res;
246 }
247
248 static gboolean
gst_rtp_mp4a_pay_setcaps(GstRTPBasePayload * payload,GstCaps * caps)249 gst_rtp_mp4a_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
250 {
251 GstRtpMP4APay *rtpmp4apay;
252 GstStructure *structure;
253 const GValue *codec_data;
254 gboolean res, framed = TRUE;
255 const gchar *stream_format;
256
257 rtpmp4apay = GST_RTP_MP4A_PAY (payload);
258
259 structure = gst_caps_get_structure (caps, 0);
260
261 /* this is already handled by the template caps, but it is better
262 * to leave here to have meaningful warning messages when linking
263 * fails */
264 stream_format = gst_structure_get_string (structure, "stream-format");
265 if (stream_format) {
266 if (strcmp (stream_format, "raw") != 0) {
267 GST_WARNING_OBJECT (rtpmp4apay, "AAC's stream-format must be 'raw', "
268 "%s is not supported", stream_format);
269 return FALSE;
270 }
271 } else {
272 GST_WARNING_OBJECT (rtpmp4apay, "AAC's stream-format not specified, "
273 "assuming 'raw'");
274 }
275
276 codec_data = gst_structure_get_value (structure, "codec_data");
277 if (codec_data) {
278 GST_LOG_OBJECT (rtpmp4apay, "got codec_data");
279 if (G_VALUE_TYPE (codec_data) == GST_TYPE_BUFFER) {
280 GstBuffer *buffer, *cbuffer;
281 GstMapInfo map;
282 GstMapInfo cmap;
283 guint i;
284
285 buffer = gst_value_get_buffer (codec_data);
286 GST_LOG_OBJECT (rtpmp4apay, "configuring codec_data");
287
288 /* parse buffer */
289 res = gst_rtp_mp4a_pay_parse_audio_config (rtpmp4apay, buffer);
290
291 if (!res)
292 goto config_failed;
293
294 gst_buffer_map (buffer, &map, GST_MAP_READ);
295
296 /* make the StreamMuxConfig, we need 15 bits for the header */
297 cbuffer = gst_buffer_new_and_alloc (map.size + 2);
298 gst_buffer_map (cbuffer, &cmap, GST_MAP_WRITE);
299
300 memset (cmap.data, 0, map.size + 2);
301
302 /* Create StreamMuxConfig according to ISO/IEC 14496-3:
303 *
304 * audioMuxVersion == 0 (1 bit)
305 * allStreamsSameTimeFraming == 1 (1 bit)
306 * numSubFrames == numSubFrames (6 bits)
307 * numProgram == 0 (4 bits)
308 * numLayer == 0 (3 bits)
309 */
310 cmap.data[0] = 0x40;
311 cmap.data[1] = 0x00;
312
313 /* append the config bits, shifting them 1 bit left */
314 for (i = 0; i < map.size; i++) {
315 cmap.data[i + 1] |= ((map.data[i] & 0x80) >> 7);
316 cmap.data[i + 2] |= ((map.data[i] & 0x7f) << 1);
317 }
318
319 gst_buffer_unmap (cbuffer, &cmap);
320 gst_buffer_unmap (buffer, &map);
321
322 /* now we can configure the buffer */
323 if (rtpmp4apay->config)
324 gst_buffer_unref (rtpmp4apay->config);
325 rtpmp4apay->config = cbuffer;
326 }
327 }
328
329 if (gst_structure_get_boolean (structure, "framed", &framed) && !framed) {
330 GST_WARNING_OBJECT (payload, "Need framed AAC data as input!");
331 }
332
333 gst_rtp_base_payload_set_options (payload, "audio", TRUE, "MP4A-LATM",
334 rtpmp4apay->rate);
335
336 res = gst_rtp_mp4a_pay_new_caps (rtpmp4apay);
337
338 return res;
339
340 /* ERRORS */
341 config_failed:
342 {
343 GST_DEBUG_OBJECT (rtpmp4apay, "failed to parse config");
344 return FALSE;
345 }
346 }
347
348 #define RTP_HEADER_LEN 12
349
350 /* we expect buffers as exactly one complete AU
351 */
352 static GstFlowReturn
gst_rtp_mp4a_pay_handle_buffer(GstRTPBasePayload * basepayload,GstBuffer * buffer)353 gst_rtp_mp4a_pay_handle_buffer (GstRTPBasePayload * basepayload,
354 GstBuffer * buffer)
355 {
356 GstRtpMP4APay *rtpmp4apay;
357 GstFlowReturn ret;
358 GstBufferList *list;
359 guint mtu;
360 guint offset;
361 gsize size;
362 gboolean fragmented;
363 GstClockTime timestamp;
364
365 ret = GST_FLOW_OK;
366
367 rtpmp4apay = GST_RTP_MP4A_PAY (basepayload);
368
369 offset = 0;
370 size = gst_buffer_get_size (buffer);
371
372 timestamp = GST_BUFFER_PTS (buffer);
373
374 fragmented = FALSE;
375 mtu = GST_RTP_BASE_PAYLOAD_MTU (rtpmp4apay);
376
377 list = gst_buffer_list_new_sized (size / (mtu - RTP_HEADER_LEN) + 1);
378
379 while (size > 0) {
380 guint towrite;
381 GstBuffer *outbuf;
382 guint payload_len;
383 guint packet_len;
384 guint header_len;
385 GstBuffer *paybuf;
386 GstRTPBuffer rtp = { NULL };
387
388 header_len = 0;
389 if (!fragmented) {
390 guint count;
391 /* first packet calculate space for the packet including the header */
392 count = size;
393 while (count >= 0xff) {
394 header_len++;
395 count -= 0xff;
396 }
397 header_len++;
398 }
399
400 packet_len = gst_rtp_buffer_calc_packet_len (header_len + size, 0, 0);
401 towrite = MIN (packet_len, mtu);
402 payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
403 payload_len -= header_len;
404
405 GST_DEBUG_OBJECT (rtpmp4apay,
406 "avail %" G_GSIZE_FORMAT
407 ", header_len %d, packet_len %d, payload_len %d", size, header_len,
408 packet_len, payload_len);
409
410 /* create buffer to hold the payload. */
411 outbuf = gst_rtp_buffer_new_allocate (header_len, 0, 0);
412
413 /* copy payload */
414 gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
415
416 if (!fragmented) {
417 guint8 *payload = gst_rtp_buffer_get_payload (&rtp);
418 guint count;
419
420 /* first packet write the header */
421 count = size;
422 while (count >= 0xff) {
423 *payload++ = 0xff;
424 count -= 0xff;
425 }
426 *payload++ = count;
427 }
428
429 /* marker only if the packet is complete */
430 gst_rtp_buffer_set_marker (&rtp, size == payload_len);
431
432 gst_rtp_buffer_unmap (&rtp);
433
434 /* create a new buf to hold the payload */
435 paybuf = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_ALL,
436 offset, payload_len);
437
438 /* join memory parts */
439 gst_rtp_copy_audio_meta (rtpmp4apay, outbuf, paybuf);
440 outbuf = gst_buffer_append (outbuf, paybuf);
441 gst_buffer_list_add (list, outbuf);
442 offset += payload_len;
443 size -= payload_len;
444
445 /* copy incoming timestamp (if any) to outgoing buffers */
446 GST_BUFFER_PTS (outbuf) = timestamp;
447
448 fragmented = TRUE;
449 }
450
451 ret =
452 gst_rtp_base_payload_push_list (GST_RTP_BASE_PAYLOAD (rtpmp4apay), list);
453
454 gst_buffer_unref (buffer);
455
456 return ret;
457 }
458
459 gboolean
gst_rtp_mp4a_pay_plugin_init(GstPlugin * plugin)460 gst_rtp_mp4a_pay_plugin_init (GstPlugin * plugin)
461 {
462 return gst_element_register (plugin, "rtpmp4apay",
463 GST_RANK_SECONDARY, GST_TYPE_RTP_MP4A_PAY);
464 }
465