2 * Copyright (C) <2008> Wim Taymans <wim.taymans@gmail.com>
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.
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.
26 #include <gst/rtp/gstrtpbuffer.h>
27 #include <gst/audio/audio.h>
29 #include "gstrtpmp4apay.h"
30 #include "gstrtputils.h"
32 GST_DEBUG_CATEGORY_STATIC (rtpmp4apay_debug);
33 #define GST_CAT_DEFAULT (rtpmp4apay_debug)
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",
41 GST_STATIC_CAPS ("audio/mpeg, mpegversion=(int)4, "
42 "stream-format=(string)raw")
45 static GstStaticPadTemplate gst_rtp_mp4a_pay_src_template =
46 GST_STATIC_PAD_TEMPLATE ("src",
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
56 * "cpresent = (string) \"0\""
62 static void gst_rtp_mp4a_pay_finalize (GObject * object);
64 static gboolean gst_rtp_mp4a_pay_setcaps (GstRTPBasePayload * payload,
66 static GstFlowReturn gst_rtp_mp4a_pay_handle_buffer (GstRTPBasePayload *
67 payload, GstBuffer * buffer);
69 #define gst_rtp_mp4a_pay_parent_class parent_class
70 G_DEFINE_TYPE (GstRtpMP4APay, gst_rtp_mp4a_pay, GST_TYPE_RTP_BASE_PAYLOAD)
72 static void gst_rtp_mp4a_pay_class_init (GstRtpMP4APayClass * klass)
74 GObjectClass *gobject_class;
75 GstElementClass *gstelement_class;
76 GstRTPBasePayloadClass *gstrtpbasepayload_class;
78 gobject_class = (GObjectClass *) klass;
79 gstelement_class = (GstElementClass *) klass;
80 gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
82 gobject_class->finalize = gst_rtp_mp4a_pay_finalize;
84 gstrtpbasepayload_class->set_caps = gst_rtp_mp4a_pay_setcaps;
85 gstrtpbasepayload_class->handle_buffer = gst_rtp_mp4a_pay_handle_buffer;
87 gst_element_class_add_static_pad_template (gstelement_class,
88 &gst_rtp_mp4a_pay_src_template);
89 gst_element_class_add_static_pad_template (gstelement_class,
90 &gst_rtp_mp4a_pay_sink_template);
92 gst_element_class_set_static_metadata (gstelement_class,
93 "RTP MPEG4 audio payloader", "Codec/Payloader/Network/RTP",
94 "Payload MPEG4 audio as RTP packets (RFC 3016)",
95 "Wim Taymans <wim.taymans@gmail.com>");
97 GST_DEBUG_CATEGORY_INIT (rtpmp4apay_debug, "rtpmp4apay", 0,
98 "MP4A-LATM RTP Payloader");
102 gst_rtp_mp4a_pay_init (GstRtpMP4APay * rtpmp4apay)
104 rtpmp4apay->rate = 90000;
105 rtpmp4apay->profile = g_strdup ("1");
109 gst_rtp_mp4a_pay_finalize (GObject * object)
111 GstRtpMP4APay *rtpmp4apay;
113 rtpmp4apay = GST_RTP_MP4A_PAY (object);
115 g_free (rtpmp4apay->params);
116 rtpmp4apay->params = NULL;
118 if (rtpmp4apay->config)
119 gst_buffer_unref (rtpmp4apay->config);
120 rtpmp4apay->config = NULL;
122 g_free (rtpmp4apay->profile);
123 rtpmp4apay->profile = NULL;
125 G_OBJECT_CLASS (parent_class)->finalize (object);
128 static const unsigned int sampling_table[16] = {
129 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
130 16000, 12000, 11025, 8000, 7350, 0, 0, 0
134 gst_rtp_mp4a_pay_parse_audio_config (GstRtpMP4APay * rtpmp4apay,
144 gst_buffer_map (buffer, &map, GST_MAP_READ);
151 /* any object type is fine, we need to copy it to the profile-level-id field. */
152 objectType = (data[0] & 0xf8) >> 3;
156 samplingIdx = ((data[0] & 0x07) << 1) | ((data[1] & 0x80) >> 7);
157 /* only fixed values for now */
158 if (samplingIdx > 12 && samplingIdx != 15)
161 channelCfg = ((data[1] & 0x78) >> 3);
165 /* rtp rate depends on sampling rate of the audio */
166 if (samplingIdx == 15) {
170 /* index of 15 means we get the rate in the next 24 bits */
171 rtpmp4apay->rate = ((data[1] & 0x7f) << 17) |
172 ((data[2]) << 9) | ((data[3]) << 1) | ((data[4] & 0x80) >> 7);
174 /* else use the rate from the table */
175 rtpmp4apay->rate = sampling_table[samplingIdx];
177 /* extra rtp params contain the number of channels */
178 g_free (rtpmp4apay->params);
179 rtpmp4apay->params = g_strdup_printf ("%d", channelCfg);
180 /* audio stream type */
181 rtpmp4apay->streamtype = "5";
183 g_free (rtpmp4apay->profile);
184 rtpmp4apay->profile = g_strdup_printf ("%d", objectType);
186 GST_DEBUG_OBJECT (rtpmp4apay,
187 "objectType: %d, samplingIdx: %d (%d), channelCfg: %d", objectType,
188 samplingIdx, rtpmp4apay->rate, channelCfg);
190 gst_buffer_unmap (buffer, &map);
197 GST_ELEMENT_ERROR (rtpmp4apay, STREAM, FORMAT,
199 ("config string too short, expected 2 bytes, got %" G_GSIZE_FORMAT,
201 gst_buffer_unmap (buffer, &map);
206 GST_ELEMENT_ERROR (rtpmp4apay, STREAM, FORMAT,
207 (NULL), ("invalid object type 0"));
208 gst_buffer_unmap (buffer, &map);
213 GST_ELEMENT_ERROR (rtpmp4apay, STREAM, NOT_IMPLEMENTED,
214 (NULL), ("unsupported frequency index %d", samplingIdx));
215 gst_buffer_unmap (buffer, &map);
220 GST_ELEMENT_ERROR (rtpmp4apay, STREAM, NOT_IMPLEMENTED,
221 (NULL), ("unsupported number of channels %d, must < 8", channelCfg));
222 gst_buffer_unmap (buffer, &map);
228 gst_rtp_mp4a_pay_new_caps (GstRtpMP4APay * rtpmp4apay)
234 g_value_init (&v, GST_TYPE_BUFFER);
235 gst_value_set_buffer (&v, rtpmp4apay->config);
236 config = gst_value_serialize (&v);
238 res = gst_rtp_base_payload_set_outcaps (GST_RTP_BASE_PAYLOAD (rtpmp4apay),
239 "cpresent", G_TYPE_STRING, "0", "config", G_TYPE_STRING, config, NULL);
248 gst_rtp_mp4a_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
250 GstRtpMP4APay *rtpmp4apay;
251 GstStructure *structure;
252 const GValue *codec_data;
253 gboolean res, framed = TRUE;
254 const gchar *stream_format;
256 rtpmp4apay = GST_RTP_MP4A_PAY (payload);
258 structure = gst_caps_get_structure (caps, 0);
260 /* this is already handled by the template caps, but it is better
261 * to leave here to have meaningful warning messages when linking
263 stream_format = gst_structure_get_string (structure, "stream-format");
265 if (strcmp (stream_format, "raw") != 0) {
266 GST_WARNING_OBJECT (rtpmp4apay, "AAC's stream-format must be 'raw', "
267 "%s is not supported", stream_format);
271 GST_WARNING_OBJECT (rtpmp4apay, "AAC's stream-format not specified, "
275 codec_data = gst_structure_get_value (structure, "codec_data");
277 GST_LOG_OBJECT (rtpmp4apay, "got codec_data");
278 if (G_VALUE_TYPE (codec_data) == GST_TYPE_BUFFER) {
279 GstBuffer *buffer, *cbuffer;
284 buffer = gst_value_get_buffer (codec_data);
285 GST_LOG_OBJECT (rtpmp4apay, "configuring codec_data");
288 res = gst_rtp_mp4a_pay_parse_audio_config (rtpmp4apay, buffer);
293 gst_buffer_map (buffer, &map, GST_MAP_READ);
295 /* make the StreamMuxConfig, we need 15 bits for the header */
296 cbuffer = gst_buffer_new_and_alloc (map.size + 2);
297 gst_buffer_map (cbuffer, &cmap, GST_MAP_WRITE);
299 memset (cmap.data, 0, map.size + 2);
301 /* Create StreamMuxConfig according to ISO/IEC 14496-3:
303 * audioMuxVersion == 0 (1 bit)
304 * allStreamsSameTimeFraming == 1 (1 bit)
305 * numSubFrames == numSubFrames (6 bits)
306 * numProgram == 0 (4 bits)
307 * numLayer == 0 (3 bits)
312 /* append the config bits, shifting them 1 bit left */
313 for (i = 0; i < map.size; i++) {
314 cmap.data[i + 1] |= ((map.data[i] & 0x80) >> 7);
315 cmap.data[i + 2] |= ((map.data[i] & 0x7f) << 1);
318 gst_buffer_unmap (cbuffer, &cmap);
319 gst_buffer_unmap (buffer, &map);
321 /* now we can configure the buffer */
322 if (rtpmp4apay->config)
323 gst_buffer_unref (rtpmp4apay->config);
324 rtpmp4apay->config = cbuffer;
328 if (gst_structure_get_boolean (structure, "framed", &framed) && !framed) {
329 GST_WARNING_OBJECT (payload, "Need framed AAC data as input!");
332 gst_rtp_base_payload_set_options (payload, "audio", TRUE, "MP4A-LATM",
335 res = gst_rtp_mp4a_pay_new_caps (rtpmp4apay);
342 GST_DEBUG_OBJECT (rtpmp4apay, "failed to parse config");
347 #define RTP_HEADER_LEN 12
349 /* we expect buffers as exactly one complete AU
352 gst_rtp_mp4a_pay_handle_buffer (GstRTPBasePayload * basepayload,
355 GstRtpMP4APay *rtpmp4apay;
362 GstClockTime timestamp;
366 rtpmp4apay = GST_RTP_MP4A_PAY (basepayload);
369 size = gst_buffer_get_size (buffer);
371 timestamp = GST_BUFFER_PTS (buffer);
374 mtu = GST_RTP_BASE_PAYLOAD_MTU (rtpmp4apay);
376 list = gst_buffer_list_new_sized (size / (mtu - RTP_HEADER_LEN) + 1);
385 GstRTPBuffer rtp = { NULL };
390 /* first packet calculate space for the packet including the header */
392 while (count >= 0xff) {
399 packet_len = gst_rtp_buffer_calc_packet_len (header_len + size, 0, 0);
400 towrite = MIN (packet_len, mtu);
401 payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
402 payload_len -= header_len;
404 GST_DEBUG_OBJECT (rtpmp4apay,
405 "avail %" G_GSIZE_FORMAT
406 ", header_len %d, packet_len %d, payload_len %d", size, header_len,
407 packet_len, payload_len);
409 /* create buffer to hold the payload. */
410 outbuf = gst_rtp_buffer_new_allocate (header_len, 0, 0);
413 gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
416 guint8 *payload = gst_rtp_buffer_get_payload (&rtp);
419 /* first packet write the header */
421 while (count >= 0xff) {
428 /* marker only if the packet is complete */
429 gst_rtp_buffer_set_marker (&rtp, size == payload_len);
431 gst_rtp_buffer_unmap (&rtp);
433 /* create a new buf to hold the payload */
434 paybuf = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_ALL,
435 offset, payload_len);
437 /* join memory parts */
438 gst_rtp_copy_meta (GST_ELEMENT_CAST (rtpmp4apay), outbuf, paybuf,
439 g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
440 outbuf = gst_buffer_append (outbuf, paybuf);
441 gst_buffer_list_add (list, outbuf);
442 offset += payload_len;
445 /* copy incomming timestamp (if any) to outgoing buffers */
446 GST_BUFFER_PTS (outbuf) = timestamp;
452 gst_rtp_base_payload_push_list (GST_RTP_BASE_PAYLOAD (rtpmp4apay), list);
454 gst_buffer_unref (buffer);
460 gst_rtp_mp4a_pay_plugin_init (GstPlugin * plugin)
462 return gst_element_register (plugin, "rtpmp4apay",
463 GST_RANK_SECONDARY, GST_TYPE_RTP_MP4A_PAY);