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) { false, 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 ();
112 rtpmp4adepay->framed = FALSE;
116 gst_rtp_mp4a_depay_finalize (GObject * object)
118 GstRtpMP4ADepay *rtpmp4adepay;
120 rtpmp4adepay = GST_RTP_MP4A_DEPAY (object);
122 g_object_unref (rtpmp4adepay->adapter);
123 rtpmp4adepay->adapter = NULL;
125 G_OBJECT_CLASS (parent_class)->finalize (object);
128 static const guint aac_sample_rates[] = { 96000, 88200, 64000, 48000,
129 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000, 7350
133 gst_rtp_mp4a_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
135 GstStructure *structure;
136 GstRtpMP4ADepay *rtpmp4adepay;
141 gint channels = 2; /* default */
144 rtpmp4adepay = GST_RTP_MP4A_DEPAY (depayload);
146 rtpmp4adepay->framed = FALSE;
148 structure = gst_caps_get_structure (caps, 0);
150 if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
151 clock_rate = 90000; /* default */
152 depayload->clock_rate = clock_rate;
154 if (!gst_structure_get_int (structure, "object", &object_type))
155 object_type = 2; /* AAC LC default */
157 srccaps = gst_caps_new_simple ("audio/mpeg",
158 "mpegversion", G_TYPE_INT, 4,
159 "framed", G_TYPE_BOOLEAN, FALSE, "channels", G_TYPE_INT, channels,
160 "stream-format", G_TYPE_STRING, "raw", NULL);
162 if ((str = gst_structure_get_string (structure, "config"))) {
165 g_value_init (&v, GST_TYPE_BUFFER);
166 if (gst_value_deserialize (&v, str)) {
173 guint8 obj_type = 0, sr_idx = 0, channels = 0;
176 buffer = gst_value_get_buffer (&v);
177 gst_buffer_ref (buffer);
180 gst_buffer_map (buffer, &map, GST_MAP_READ);
185 GST_WARNING_OBJECT (depayload, "config too short (%d < 2)",
190 /* Parse StreamMuxConfig according to ISO/IEC 14496-3:
192 * audioMuxVersion == 0 (1 bit)
193 * allStreamsSameTimeFraming == 1 (1 bit)
194 * numSubFrames == rtpmp4adepay->numSubFrames (6 bits)
195 * numProgram == 0 (4 bits)
196 * numLayer == 0 (3 bits)
198 * We only require audioMuxVersion == 0;
200 * The remaining bit of the second byte and the rest of the bits are used
201 * for audioSpecificConfig which we need to set in codec_info.
203 if ((data[0] & 0x80) != 0x00) {
204 GST_WARNING_OBJECT (depayload, "unknown audioMuxVersion 1");
208 rtpmp4adepay->numSubFrames = (data[0] & 0x3F);
210 GST_LOG_OBJECT (rtpmp4adepay, "numSubFrames %d",
211 rtpmp4adepay->numSubFrames);
213 /* shift rest of string 15 bits down */
215 for (i = 0; i < size; i++) {
216 data[i] = ((data[i + 1] & 1) << 7) | ((data[i + 2] & 0xfe) >> 1);
219 gst_bit_reader_init (&br, data, size);
221 /* any object type is fine, we need to copy it to the profile-level-id field. */
222 if (!gst_bit_reader_get_bits_uint8 (&br, &obj_type, 5))
225 GST_WARNING_OBJECT (depayload, "invalid object type 0");
229 if (!gst_bit_reader_get_bits_uint8 (&br, &sr_idx, 4))
231 if (sr_idx >= G_N_ELEMENTS (aac_sample_rates) && sr_idx != 15) {
232 GST_WARNING_OBJECT (depayload, "invalid sample rate index %d", sr_idx);
235 GST_LOG_OBJECT (rtpmp4adepay, "sample rate index %u", sr_idx);
237 if (!gst_bit_reader_get_bits_uint8 (&br, &channels, 4))
240 GST_WARNING_OBJECT (depayload, "invalid channels %u", (guint) channels);
244 /* rtp rate depends on sampling rate of the audio */
246 /* index of 15 means we get the rate in the next 24 bits */
247 if (!gst_bit_reader_get_bits_uint32 (&br, &rate, 24))
249 } else if (sr_idx >= G_N_ELEMENTS (aac_sample_rates)) {
252 /* else use the rate from the table */
253 rate = aac_sample_rates[sr_idx];
256 rtpmp4adepay->frame_len = 1024;
266 guint8 frameLenFlag = 0;
268 if (gst_bit_reader_get_bits_uint8 (&br, &frameLenFlag, 1))
270 rtpmp4adepay->frame_len = 960;
277 /* ignore remaining bit, we're only interested in full bytes */
278 gst_buffer_resize (buffer, 0, size);
279 gst_buffer_unmap (buffer, &map);
282 gst_caps_set_simple (srccaps,
283 "channels", G_TYPE_INT, (gint) channels,
284 "rate", G_TYPE_INT, (gint) rate,
285 "codec_data", GST_TYPE_BUFFER, buffer, NULL);
288 gst_buffer_unmap (buffer, &map);
289 gst_buffer_unref (buffer);
291 g_warning ("cannot convert config to buffer");
294 res = gst_pad_set_caps (depayload->srcpad, srccaps);
295 gst_caps_unref (srccaps);
301 gst_rtp_mp4a_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
303 GstRtpMP4ADepay *rtpmp4adepay;
307 rtpmp4adepay = GST_RTP_MP4A_DEPAY (depayload);
309 /* flush remaining data on discont */
310 if (GST_BUFFER_IS_DISCONT (rtp->buffer)) {
311 gst_adapter_clear (rtpmp4adepay->adapter);
314 outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
316 if (!rtpmp4adepay->framed) {
317 if (gst_rtp_buffer_get_marker (rtp)) {
320 rtpmp4adepay->framed = TRUE;
322 gst_rtp_base_depayload_push (depayload, outbuf);
324 caps = gst_pad_get_current_caps (depayload->srcpad);
325 caps = gst_caps_make_writable (caps);
326 gst_caps_set_simple (caps, "framed", G_TYPE_BOOLEAN, TRUE, NULL);
327 gst_pad_set_caps (depayload->srcpad, caps);
328 gst_caps_unref (caps);
335 outbuf = gst_buffer_make_writable (outbuf);
336 GST_BUFFER_PTS (outbuf) = GST_BUFFER_PTS (rtp->buffer);
337 gst_adapter_push (rtpmp4adepay->adapter, outbuf);
339 /* RTP marker bit indicates the last packet of the AudioMuxElement => create
340 * and push a buffer */
341 if (gst_rtp_buffer_get_marker (rtp)) {
346 GstClockTime timestamp;
348 avail = gst_adapter_available (rtpmp4adepay->adapter);
349 timestamp = gst_adapter_prev_pts (rtpmp4adepay->adapter, NULL);
351 GST_LOG_OBJECT (rtpmp4adepay, "have marker and %u available", avail);
353 outbuf = gst_adapter_take_buffer (rtpmp4adepay->adapter, avail);
354 gst_buffer_map (outbuf, &map, GST_MAP_READ);
356 /* position in data we are at */
359 /* looping through the number of sub-frames in the audio payload */
360 for (i = 0; i <= rtpmp4adepay->numSubFrames; i++) {
361 /* determine payload length and set buffer data pointer accordingly */
364 GstBuffer *tmp = NULL;
366 /* each subframe starts with a variable length encoding */
368 for (skip = 0; skip < avail; skip++) {
369 data_len += data[skip];
370 if (data[skip] != 0xff)
375 /* this can not be possible, we have not enough data or the length
376 * decoding failed because we ran out of data. */
377 if (skip + data_len > avail)
380 GST_LOG_OBJECT (rtpmp4adepay,
381 "subframe %u, header len %u, data len %u, left %u", i, skip, data_len,
384 /* take data out, skip the header */
386 tmp = gst_buffer_copy_region (outbuf, GST_BUFFER_COPY_ALL, pos, data_len);
392 /* update our pointers with what we consumed */
396 GST_BUFFER_PTS (tmp) = timestamp;
397 gst_rtp_drop_non_audio_meta (depayload, tmp);
398 gst_rtp_base_depayload_push (depayload, tmp);
400 /* shift ts for next buffers */
401 if (rtpmp4adepay->frame_len && timestamp != -1
402 && depayload->clock_rate != 0) {
404 gst_util_uint64_scale_int (rtpmp4adepay->frame_len, GST_SECOND,
405 depayload->clock_rate);
409 /* just a check that lengths match */
411 GST_ELEMENT_WARNING (depayload, STREAM, DECODE,
412 ("Packet invalid"), ("Not all payload consumed: "
413 "possible wrongly encoded packet."));
416 gst_buffer_unmap (outbuf, &map);
417 gst_buffer_unref (outbuf);
424 GST_ELEMENT_WARNING (rtpmp4adepay, STREAM, DECODE,
425 ("Packet did not validate"), ("wrong packet size"));
426 gst_buffer_unmap (outbuf, &map);
427 gst_buffer_unref (outbuf);
432 static GstStateChangeReturn
433 gst_rtp_mp4a_depay_change_state (GstElement * element,
434 GstStateChange transition)
436 GstRtpMP4ADepay *rtpmp4adepay;
437 GstStateChangeReturn ret;
439 rtpmp4adepay = GST_RTP_MP4A_DEPAY (element);
441 switch (transition) {
442 case GST_STATE_CHANGE_READY_TO_PAUSED:
443 gst_adapter_clear (rtpmp4adepay->adapter);
444 rtpmp4adepay->frame_len = 0;
445 rtpmp4adepay->numSubFrames = 0;
446 rtpmp4adepay->framed = FALSE;
452 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
454 switch (transition) {
462 gst_rtp_mp4a_depay_plugin_init (GstPlugin * plugin)
464 return gst_element_register (plugin, "rtpmp4adepay",
465 GST_RANK_SECONDARY, GST_TYPE_RTP_MP4A_DEPAY);