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>
26 #include <gst/audio/audio.h>
29 #include "gstrtpmp4adepay.h"
30 #include "gstrtputils.h"
32 GST_DEBUG_CATEGORY_STATIC (rtpmp4adepay_debug);
33 #define GST_CAT_DEFAULT (rtpmp4adepay_debug)
35 static GstStaticPadTemplate gst_rtp_mp4a_depay_src_template =
36 GST_STATIC_PAD_TEMPLATE ("src",
39 GST_STATIC_CAPS ("audio/mpeg,"
40 "mpegversion = (int) 4," "framed = (boolean) true, "
41 "stream-format = (string) raw")
44 static GstStaticPadTemplate gst_rtp_mp4a_depay_sink_template =
45 GST_STATIC_PAD_TEMPLATE ("sink",
48 GST_STATIC_CAPS ("application/x-rtp, "
49 "media = (string) \"audio\", "
50 "clock-rate = (int) [1, MAX ], "
51 "encoding-name = (string) \"MP4A-LATM\""
52 /* All optional parameters
54 * "profile-level-id=[1,MAX]"
60 #define gst_rtp_mp4a_depay_parent_class parent_class
61 G_DEFINE_TYPE (GstRtpMP4ADepay, gst_rtp_mp4a_depay,
62 GST_TYPE_RTP_BASE_DEPAYLOAD);
64 static void gst_rtp_mp4a_depay_finalize (GObject * object);
66 static gboolean gst_rtp_mp4a_depay_setcaps (GstRTPBaseDepayload * depayload,
68 static GstBuffer *gst_rtp_mp4a_depay_process (GstRTPBaseDepayload * depayload,
71 static GstStateChangeReturn gst_rtp_mp4a_depay_change_state (GstElement *
72 element, GstStateChange transition);
76 gst_rtp_mp4a_depay_class_init (GstRtpMP4ADepayClass * klass)
78 GObjectClass *gobject_class;
79 GstElementClass *gstelement_class;
80 GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
82 gobject_class = (GObjectClass *) klass;
83 gstelement_class = (GstElementClass *) klass;
84 gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
86 gobject_class->finalize = gst_rtp_mp4a_depay_finalize;
88 gstelement_class->change_state = gst_rtp_mp4a_depay_change_state;
90 gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_mp4a_depay_process;
91 gstrtpbasedepayload_class->set_caps = gst_rtp_mp4a_depay_setcaps;
93 gst_element_class_add_static_pad_template (gstelement_class,
94 &gst_rtp_mp4a_depay_src_template);
95 gst_element_class_add_static_pad_template (gstelement_class,
96 &gst_rtp_mp4a_depay_sink_template);
98 gst_element_class_set_static_metadata (gstelement_class,
99 "RTP MPEG4 audio depayloader", "Codec/Depayloader/Network/RTP",
100 "Extracts MPEG4 audio from RTP packets (RFC 3016)",
101 "Nokia Corporation (contact <stefan.kost@nokia.com>), "
102 "Wim Taymans <wim.taymans@gmail.com>");
104 GST_DEBUG_CATEGORY_INIT (rtpmp4adepay_debug, "rtpmp4adepay", 0,
105 "MPEG4 audio RTP Depayloader");
109 gst_rtp_mp4a_depay_init (GstRtpMP4ADepay * rtpmp4adepay)
111 rtpmp4adepay->adapter = gst_adapter_new ();
115 gst_rtp_mp4a_depay_finalize (GObject * object)
117 GstRtpMP4ADepay *rtpmp4adepay;
119 rtpmp4adepay = GST_RTP_MP4A_DEPAY (object);
121 g_object_unref (rtpmp4adepay->adapter);
122 rtpmp4adepay->adapter = NULL;
124 G_OBJECT_CLASS (parent_class)->finalize (object);
127 static const guint aac_sample_rates[] = { 96000, 88200, 64000, 48000,
128 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000, 7350
132 gst_rtp_mp4a_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
134 GstStructure *structure;
135 GstRtpMP4ADepay *rtpmp4adepay;
140 gint channels = 2; /* default */
143 rtpmp4adepay = GST_RTP_MP4A_DEPAY (depayload);
145 structure = gst_caps_get_structure (caps, 0);
147 if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
148 clock_rate = 90000; /* default */
149 depayload->clock_rate = clock_rate;
151 if (!gst_structure_get_int (structure, "object", &object_type))
152 object_type = 2; /* AAC LC default */
154 srccaps = gst_caps_new_simple ("audio/mpeg",
155 "mpegversion", G_TYPE_INT, 4,
156 "framed", G_TYPE_BOOLEAN, TRUE, "channels", G_TYPE_INT, channels,
157 "stream-format", G_TYPE_STRING, "raw", NULL);
159 if ((str = gst_structure_get_string (structure, "config"))) {
162 g_value_init (&v, GST_TYPE_BUFFER);
163 if (gst_value_deserialize (&v, str)) {
170 guint8 obj_type = 0, sr_idx = 0, channels = 0;
173 buffer = gst_value_get_buffer (&v);
174 gst_buffer_ref (buffer);
177 gst_buffer_map (buffer, &map, GST_MAP_READ);
182 GST_WARNING_OBJECT (depayload, "config too short (%d < 2)",
187 /* Parse StreamMuxConfig according to ISO/IEC 14496-3:
189 * audioMuxVersion == 0 (1 bit)
190 * allStreamsSameTimeFraming == 1 (1 bit)
191 * numSubFrames == rtpmp4adepay->numSubFrames (6 bits)
192 * numProgram == 0 (4 bits)
193 * numLayer == 0 (3 bits)
195 * We only require audioMuxVersion == 0;
197 * The remaining bit of the second byte and the rest of the bits are used
198 * for audioSpecificConfig which we need to set in codec_info.
200 if ((data[0] & 0x80) != 0x00) {
201 GST_WARNING_OBJECT (depayload, "unknown audioMuxVersion 1");
205 rtpmp4adepay->numSubFrames = (data[0] & 0x3F);
207 GST_LOG_OBJECT (rtpmp4adepay, "numSubFrames %d",
208 rtpmp4adepay->numSubFrames);
210 /* shift rest of string 15 bits down */
212 for (i = 0; i < size; i++) {
213 data[i] = ((data[i + 1] & 1) << 7) | ((data[i + 2] & 0xfe) >> 1);
216 gst_bit_reader_init (&br, data, size);
218 /* any object type is fine, we need to copy it to the profile-level-id field. */
219 if (!gst_bit_reader_get_bits_uint8 (&br, &obj_type, 5))
222 GST_WARNING_OBJECT (depayload, "invalid object type 0");
226 if (!gst_bit_reader_get_bits_uint8 (&br, &sr_idx, 4))
228 if (sr_idx >= G_N_ELEMENTS (aac_sample_rates) && sr_idx != 15) {
229 GST_WARNING_OBJECT (depayload, "invalid sample rate index %d", sr_idx);
232 GST_LOG_OBJECT (rtpmp4adepay, "sample rate index %u", sr_idx);
234 if (!gst_bit_reader_get_bits_uint8 (&br, &channels, 4))
237 GST_WARNING_OBJECT (depayload, "invalid channels %u", (guint) channels);
241 /* rtp rate depends on sampling rate of the audio */
243 /* index of 15 means we get the rate in the next 24 bits */
244 if (!gst_bit_reader_get_bits_uint32 (&br, &rate, 24))
246 } else if (sr_idx >= G_N_ELEMENTS (aac_sample_rates)) {
249 /* else use the rate from the table */
250 rate = aac_sample_rates[sr_idx];
253 rtpmp4adepay->frame_len = 1024;
263 guint8 frameLenFlag = 0;
265 if (gst_bit_reader_get_bits_uint8 (&br, &frameLenFlag, 1))
267 rtpmp4adepay->frame_len = 960;
274 /* ignore remaining bit, we're only interested in full bytes */
275 gst_buffer_resize (buffer, 0, size);
276 gst_buffer_unmap (buffer, &map);
279 gst_caps_set_simple (srccaps,
280 "channels", G_TYPE_INT, (gint) channels,
281 "rate", G_TYPE_INT, (gint) rate,
282 "codec_data", GST_TYPE_BUFFER, buffer, NULL);
285 gst_buffer_unmap (buffer, &map);
286 gst_buffer_unref (buffer);
288 g_warning ("cannot convert config to buffer");
291 res = gst_pad_set_caps (depayload->srcpad, srccaps);
292 gst_caps_unref (srccaps);
298 gst_rtp_mp4a_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
300 GstRtpMP4ADepay *rtpmp4adepay;
304 rtpmp4adepay = GST_RTP_MP4A_DEPAY (depayload);
306 /* flush remaining data on discont */
307 if (GST_BUFFER_IS_DISCONT (rtp->buffer)) {
308 gst_adapter_clear (rtpmp4adepay->adapter);
311 outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
313 outbuf = gst_buffer_make_writable (outbuf);
314 GST_BUFFER_PTS (outbuf) = GST_BUFFER_PTS (rtp->buffer);
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_ALL, pos, data_len);
370 /* update our pointers whith what we consumed */
374 GST_BUFFER_PTS (tmp) = timestamp;
375 gst_rtp_drop_meta (GST_ELEMENT_CAST (depayload), tmp,
376 g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
377 gst_rtp_base_depayload_push (depayload, tmp);
379 /* shift ts for next buffers */
380 if (rtpmp4adepay->frame_len && timestamp != -1
381 && depayload->clock_rate != 0) {
383 gst_util_uint64_scale_int (rtpmp4adepay->frame_len, GST_SECOND,
384 depayload->clock_rate);
388 /* just a check that lengths match */
390 GST_ELEMENT_WARNING (depayload, STREAM, DECODE,
391 ("Packet invalid"), ("Not all payload consumed: "
392 "possible wrongly encoded packet."));
395 gst_buffer_unmap (outbuf, &map);
396 gst_buffer_unref (outbuf);
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);
411 static GstStateChangeReturn
412 gst_rtp_mp4a_depay_change_state (GstElement * element,
413 GstStateChange transition)
415 GstRtpMP4ADepay *rtpmp4adepay;
416 GstStateChangeReturn ret;
418 rtpmp4adepay = GST_RTP_MP4A_DEPAY (element);
420 switch (transition) {
421 case GST_STATE_CHANGE_READY_TO_PAUSED:
422 gst_adapter_clear (rtpmp4adepay->adapter);
423 rtpmp4adepay->frame_len = 0;
424 rtpmp4adepay->numSubFrames = 0;
430 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
432 switch (transition) {
440 gst_rtp_mp4a_depay_plugin_init (GstPlugin * plugin)
442 return gst_element_register (plugin, "rtpmp4adepay",
443 GST_RANK_SECONDARY, GST_TYPE_RTP_MP4A_DEPAY);