2 * Copyright (C) <2007> Nokia Corporation (contact <stefan.kost@nokia.com>)
3 * <2007> Wim Taymans <wim.taymans@gmail.com>
5 * This library is free software; you can redistribute it and/or
6 * modify it under the terms of the GNU Library General Public
7 * License version 2 as published by the Free Software Foundation.
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.
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.
24 #include <gst/base/gstbitreader.h>
25 #include <gst/rtp/gstrtpbuffer.h>
28 #include "gstrtpmp4adepay.h"
30 GST_DEBUG_CATEGORY_STATIC (rtpmp4adepay_debug);
31 #define GST_CAT_DEFAULT (rtpmp4adepay_debug)
33 static GstStaticPadTemplate gst_rtp_mp4a_depay_src_template =
34 GST_STATIC_PAD_TEMPLATE ("src",
37 GST_STATIC_CAPS ("audio/mpeg,"
38 "mpegversion = (int) 4," "framed = (boolean) true, "
39 "stream-format = (string) raw")
42 static GstStaticPadTemplate gst_rtp_mp4a_depay_sink_template =
43 GST_STATIC_PAD_TEMPLATE ("sink",
46 GST_STATIC_CAPS ("application/x-rtp, "
47 "media = (string) \"audio\", "
48 "clock-rate = (int) [1, MAX ], "
49 "encoding-name = (string) \"MP4A-LATM\""
50 /* All optional parameters
52 * "profile-level-id=[1,MAX]"
58 #define gst_rtp_mp4a_depay_parent_class parent_class
59 G_DEFINE_TYPE (GstRtpMP4ADepay, gst_rtp_mp4a_depay,
60 GST_TYPE_RTP_BASE_DEPAYLOAD);
62 static void gst_rtp_mp4a_depay_finalize (GObject * object);
64 static gboolean gst_rtp_mp4a_depay_setcaps (GstRTPBaseDepayload * depayload,
66 static GstBuffer *gst_rtp_mp4a_depay_process (GstRTPBaseDepayload * depayload,
69 static GstStateChangeReturn gst_rtp_mp4a_depay_change_state (GstElement *
70 element, GstStateChange transition);
74 gst_rtp_mp4a_depay_class_init (GstRtpMP4ADepayClass * klass)
76 GObjectClass *gobject_class;
77 GstElementClass *gstelement_class;
78 GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
80 gobject_class = (GObjectClass *) klass;
81 gstelement_class = (GstElementClass *) klass;
82 gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
84 gobject_class->finalize = gst_rtp_mp4a_depay_finalize;
86 gstelement_class->change_state = gst_rtp_mp4a_depay_change_state;
88 gstrtpbasedepayload_class->process = gst_rtp_mp4a_depay_process;
89 gstrtpbasedepayload_class->set_caps = gst_rtp_mp4a_depay_setcaps;
91 gst_element_class_add_pad_template (gstelement_class,
92 gst_static_pad_template_get (&gst_rtp_mp4a_depay_src_template));
93 gst_element_class_add_pad_template (gstelement_class,
94 gst_static_pad_template_get (&gst_rtp_mp4a_depay_sink_template));
96 gst_element_class_set_static_metadata (gstelement_class,
97 "RTP MPEG4 audio depayloader", "Codec/Depayloader/Network/RTP",
98 "Extracts MPEG4 audio from RTP packets (RFC 3016)",
99 "Nokia Corporation (contact <stefan.kost@nokia.com>), "
100 "Wim Taymans <wim.taymans@gmail.com>");
102 GST_DEBUG_CATEGORY_INIT (rtpmp4adepay_debug, "rtpmp4adepay", 0,
103 "MPEG4 audio RTP Depayloader");
107 gst_rtp_mp4a_depay_init (GstRtpMP4ADepay * rtpmp4adepay)
109 rtpmp4adepay->adapter = gst_adapter_new ();
113 gst_rtp_mp4a_depay_finalize (GObject * object)
115 GstRtpMP4ADepay *rtpmp4adepay;
117 rtpmp4adepay = GST_RTP_MP4A_DEPAY (object);
119 g_object_unref (rtpmp4adepay->adapter);
120 rtpmp4adepay->adapter = NULL;
122 G_OBJECT_CLASS (parent_class)->finalize (object);
125 static const guint aac_sample_rates[] = { 96000, 88200, 64000, 48000,
126 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000, 7350
130 gst_rtp_mp4a_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
132 GstStructure *structure;
133 GstRtpMP4ADepay *rtpmp4adepay;
138 gint channels = 2; /* default */
141 rtpmp4adepay = GST_RTP_MP4A_DEPAY (depayload);
143 structure = gst_caps_get_structure (caps, 0);
145 if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
146 clock_rate = 90000; /* default */
147 depayload->clock_rate = clock_rate;
149 if (!gst_structure_get_int (structure, "object", &object_type))
150 object_type = 2; /* AAC LC default */
152 srccaps = gst_caps_new_simple ("audio/mpeg",
153 "mpegversion", G_TYPE_INT, 4,
154 "framed", G_TYPE_BOOLEAN, TRUE, "channels", G_TYPE_INT, channels,
155 "stream-format", G_TYPE_STRING, "raw", NULL);
157 if ((str = gst_structure_get_string (structure, "config"))) {
160 g_value_init (&v, GST_TYPE_BUFFER);
161 if (gst_value_deserialize (&v, str)) {
168 guint8 obj_type = 0, sr_idx = 0, channels = 0;
171 buffer = gst_value_get_buffer (&v);
172 gst_buffer_ref (buffer);
175 gst_buffer_map (buffer, &map, GST_MAP_READ);
180 GST_WARNING_OBJECT (depayload, "config too short (%d < 2)",
185 /* Parse StreamMuxConfig according to ISO/IEC 14496-3:
187 * audioMuxVersion == 0 (1 bit)
188 * allStreamsSameTimeFraming == 1 (1 bit)
189 * numSubFrames == rtpmp4adepay->numSubFrames (6 bits)
190 * numProgram == 0 (4 bits)
191 * numLayer == 0 (3 bits)
193 * We only require audioMuxVersion == 0;
195 * The remaining bit of the second byte and the rest of the bits are used
196 * for audioSpecificConfig which we need to set in codec_info.
198 if ((data[0] & 0x80) != 0x00) {
199 GST_WARNING_OBJECT (depayload, "unknown audioMuxVersion 1");
203 rtpmp4adepay->numSubFrames = (data[0] & 0x3F);
205 GST_LOG_OBJECT (rtpmp4adepay, "numSubFrames %d",
206 rtpmp4adepay->numSubFrames);
208 /* shift rest of string 15 bits down */
210 for (i = 0; i < size; i++) {
211 data[i] = ((data[i + 1] & 1) << 7) | ((data[i + 2] & 0xfe) >> 1);
214 gst_bit_reader_init (&br, data, size);
216 /* any object type is fine, we need to copy it to the profile-level-id field. */
217 if (!gst_bit_reader_get_bits_uint8 (&br, &obj_type, 5))
220 GST_WARNING_OBJECT (depayload, "invalid object type 0");
224 if (!gst_bit_reader_get_bits_uint8 (&br, &sr_idx, 4))
226 if (sr_idx >= G_N_ELEMENTS (aac_sample_rates) && sr_idx != 15) {
227 GST_WARNING_OBJECT (depayload, "invalid sample rate index %d", sr_idx);
230 GST_LOG_OBJECT (rtpmp4adepay, "sample rate index %u", sr_idx);
232 if (!gst_bit_reader_get_bits_uint8 (&br, &channels, 4))
235 GST_WARNING_OBJECT (depayload, "invalid channels %u", (guint) channels);
239 /* rtp rate depends on sampling rate of the audio */
241 /* index of 15 means we get the rate in the next 24 bits */
242 if (!gst_bit_reader_get_bits_uint32 (&br, &rate, 24))
244 } else if (sr_idx >= G_N_ELEMENTS (aac_sample_rates)) {
247 /* else use the rate from the table */
248 rate = aac_sample_rates[sr_idx];
251 rtpmp4adepay->frame_len = 1024;
261 guint8 frameLenFlag = 0;
263 if (gst_bit_reader_get_bits_uint8 (&br, &frameLenFlag, 1))
265 rtpmp4adepay->frame_len = 960;
272 /* ignore remaining bit, we're only interested in full bytes */
273 gst_buffer_resize (buffer, 0, size);
274 gst_buffer_unmap (buffer, &map);
277 gst_caps_set_simple (srccaps,
278 "channels", G_TYPE_INT, (gint) channels,
279 "rate", G_TYPE_INT, (gint) rate,
280 "codec_data", GST_TYPE_BUFFER, buffer, NULL);
283 gst_buffer_unmap (buffer, &map);
284 gst_buffer_unref (buffer);
286 g_warning ("cannot convert config to buffer");
289 res = gst_pad_set_caps (depayload->srcpad, srccaps);
290 gst_caps_unref (srccaps);
296 gst_rtp_mp4a_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
298 GstRtpMP4ADepay *rtpmp4adepay;
300 GstRTPBuffer rtp = { NULL };
303 rtpmp4adepay = GST_RTP_MP4A_DEPAY (depayload);
305 /* flush remaining data on discont */
306 if (GST_BUFFER_IS_DISCONT (buf)) {
307 gst_adapter_clear (rtpmp4adepay->adapter);
310 gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
311 outbuf = gst_rtp_buffer_get_payload_buffer (&rtp);
313 outbuf = gst_buffer_make_writable (outbuf);
314 GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (buf);
315 gst_adapter_push (rtpmp4adepay->adapter, outbuf);
317 /* RTP marker bit indicates the last packet of the AudioMuxElement => create
318 * and push a buffer */
319 if (gst_rtp_buffer_get_marker (&rtp)) {
324 GstClockTime timestamp;
326 avail = gst_adapter_available (rtpmp4adepay->adapter);
327 timestamp = gst_adapter_prev_pts (rtpmp4adepay->adapter, NULL);
329 GST_LOG_OBJECT (rtpmp4adepay, "have marker and %u available", avail);
331 outbuf = gst_adapter_take_buffer (rtpmp4adepay->adapter, avail);
332 gst_buffer_map (outbuf, &map, GST_MAP_READ);
334 /* position in data we are at */
337 /* looping through the number of sub-frames in the audio payload */
338 for (i = 0; i <= rtpmp4adepay->numSubFrames; i++) {
339 /* determine payload length and set buffer data pointer accordingly */
342 GstBuffer *tmp = NULL;
344 /* each subframe starts with a variable length encoding */
346 for (skip = 0; skip < avail; skip++) {
347 data_len += data[skip];
348 if (data[skip] != 0xff)
353 /* this can not be possible, we have not enough data or the length
354 * decoding failed because we ran out of data. */
355 if (skip + data_len > avail)
358 GST_LOG_OBJECT (rtpmp4adepay,
359 "subframe %u, header len %u, data len %u, left %u", i, skip, data_len,
362 /* take data out, skip the header */
364 tmp = gst_buffer_copy_region (outbuf, GST_BUFFER_COPY_MEMORY, pos,
371 /* update our pointers whith what we consumed */
375 GST_BUFFER_TIMESTAMP (tmp) = timestamp;
376 gst_rtp_base_depayload_push (depayload, tmp);
378 /* shift ts for next buffers */
379 if (rtpmp4adepay->frame_len && timestamp != -1
380 && depayload->clock_rate != 0) {
382 gst_util_uint64_scale_int (rtpmp4adepay->frame_len, GST_SECOND,
383 depayload->clock_rate);
387 /* just a check that lengths match */
389 GST_ELEMENT_WARNING (depayload, STREAM, DECODE,
390 ("Packet invalid"), ("Not all payload consumed: "
391 "possible wrongly encoded packet."));
394 gst_buffer_unmap (outbuf, &map);
395 gst_buffer_unref (outbuf);
397 gst_rtp_buffer_unmap (&rtp);
403 GST_ELEMENT_WARNING (rtpmp4adepay, STREAM, DECODE,
404 ("Packet did not validate"), ("wrong packet size"));
405 gst_buffer_unmap (outbuf, &map);
406 gst_buffer_unref (outbuf);
407 gst_rtp_buffer_unmap (&rtp);
412 static GstStateChangeReturn
413 gst_rtp_mp4a_depay_change_state (GstElement * element,
414 GstStateChange transition)
416 GstRtpMP4ADepay *rtpmp4adepay;
417 GstStateChangeReturn ret;
419 rtpmp4adepay = GST_RTP_MP4A_DEPAY (element);
421 switch (transition) {
422 case GST_STATE_CHANGE_READY_TO_PAUSED:
423 gst_adapter_clear (rtpmp4adepay->adapter);
424 rtpmp4adepay->frame_len = 0;
425 rtpmp4adepay->numSubFrames = 0;
431 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
433 switch (transition) {
441 gst_rtp_mp4a_depay_plugin_init (GstPlugin * plugin)
443 return gst_element_register (plugin, "rtpmp4adepay",
444 GST_RANK_SECONDARY, GST_TYPE_RTP_MP4A_DEPAY);