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 "gstrtpelements.h"
30 #include "gstrtpmp4apay.h"
31 #include "gstrtputils.h"
33 GST_DEBUG_CATEGORY_STATIC (rtpmp4apay_debug);
34 #define GST_CAT_DEFAULT (rtpmp4apay_debug)
36 /* FIXME: add framed=(boolean)true once our encoders have this field set
37 * on their output caps */
38 static GstStaticPadTemplate gst_rtp_mp4a_pay_sink_template =
39 GST_STATIC_PAD_TEMPLATE ("sink",
42 GST_STATIC_CAPS ("audio/mpeg, mpegversion=(int)4, "
43 "stream-format=(string)raw")
46 static GstStaticPadTemplate gst_rtp_mp4a_pay_src_template =
47 GST_STATIC_PAD_TEMPLATE ("src",
50 GST_STATIC_CAPS ("application/x-rtp, "
51 "media = (string) \"audio\", "
52 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
53 "clock-rate = (int) [1, MAX ], "
54 "encoding-name = (string) \"MP4A-LATM\""
55 /* All optional parameters
57 * "cpresent = (string) \"0\""
63 static void gst_rtp_mp4a_pay_finalize (GObject * object);
65 static gboolean gst_rtp_mp4a_pay_setcaps (GstRTPBasePayload * payload,
67 static GstFlowReturn gst_rtp_mp4a_pay_handle_buffer (GstRTPBasePayload *
68 payload, GstBuffer * buffer);
70 #define gst_rtp_mp4a_pay_parent_class parent_class
71 G_DEFINE_TYPE (GstRtpMP4APay, gst_rtp_mp4a_pay, GST_TYPE_RTP_BASE_PAYLOAD);
72 GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpmp4apay, "rtpmp4apay",
73 GST_RANK_SECONDARY, GST_TYPE_RTP_MP4A_PAY, rtp_element_init (plugin));
76 gst_rtp_mp4a_pay_class_init (GstRtpMP4APayClass * klass)
78 GObjectClass *gobject_class;
79 GstElementClass *gstelement_class;
80 GstRTPBasePayloadClass *gstrtpbasepayload_class;
82 gobject_class = (GObjectClass *) klass;
83 gstelement_class = (GstElementClass *) klass;
84 gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
86 gobject_class->finalize = gst_rtp_mp4a_pay_finalize;
88 gstrtpbasepayload_class->set_caps = gst_rtp_mp4a_pay_setcaps;
89 gstrtpbasepayload_class->handle_buffer = gst_rtp_mp4a_pay_handle_buffer;
91 gst_element_class_add_static_pad_template (gstelement_class,
92 &gst_rtp_mp4a_pay_src_template);
93 gst_element_class_add_static_pad_template (gstelement_class,
94 &gst_rtp_mp4a_pay_sink_template);
96 gst_element_class_set_static_metadata (gstelement_class,
97 "RTP MPEG4 audio payloader", "Codec/Payloader/Network/RTP",
98 "Payload MPEG4 audio as RTP packets (RFC 3016)",
99 "Wim Taymans <wim.taymans@gmail.com>");
101 GST_DEBUG_CATEGORY_INIT (rtpmp4apay_debug, "rtpmp4apay", 0,
102 "MP4A-LATM RTP Payloader");
106 gst_rtp_mp4a_pay_init (GstRtpMP4APay * rtpmp4apay)
108 rtpmp4apay->rate = 90000;
109 rtpmp4apay->profile = g_strdup ("1");
113 gst_rtp_mp4a_pay_finalize (GObject * object)
115 GstRtpMP4APay *rtpmp4apay;
117 rtpmp4apay = GST_RTP_MP4A_PAY (object);
119 g_free (rtpmp4apay->params);
120 rtpmp4apay->params = NULL;
122 if (rtpmp4apay->config)
123 gst_buffer_unref (rtpmp4apay->config);
124 rtpmp4apay->config = NULL;
126 g_free (rtpmp4apay->profile);
127 rtpmp4apay->profile = NULL;
129 G_OBJECT_CLASS (parent_class)->finalize (object);
132 static const unsigned int sampling_table[16] = {
133 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
134 16000, 12000, 11025, 8000, 7350, 0, 0, 0
138 gst_rtp_mp4a_pay_parse_audio_config (GstRtpMP4APay * rtpmp4apay,
148 gst_buffer_map (buffer, &map, GST_MAP_READ);
155 /* any object type is fine, we need to copy it to the profile-level-id field. */
156 objectType = (data[0] & 0xf8) >> 3;
160 samplingIdx = ((data[0] & 0x07) << 1) | ((data[1] & 0x80) >> 7);
161 /* only fixed values for now */
162 if (samplingIdx > 12 && samplingIdx != 15)
165 channelCfg = ((data[1] & 0x78) >> 3);
169 /* rtp rate depends on sampling rate of the audio */
170 if (samplingIdx == 15) {
174 /* index of 15 means we get the rate in the next 24 bits */
175 rtpmp4apay->rate = ((data[1] & 0x7f) << 17) |
176 ((data[2]) << 9) | ((data[3]) << 1) | ((data[4] & 0x80) >> 7);
178 /* else use the rate from the table */
179 rtpmp4apay->rate = sampling_table[samplingIdx];
181 /* extra rtp params contain the number of channels */
182 g_free (rtpmp4apay->params);
183 rtpmp4apay->params = g_strdup_printf ("%d", channelCfg);
184 /* audio stream type */
185 rtpmp4apay->streamtype = "5";
187 g_free (rtpmp4apay->profile);
188 rtpmp4apay->profile = g_strdup_printf ("%d", objectType);
190 GST_DEBUG_OBJECT (rtpmp4apay,
191 "objectType: %d, samplingIdx: %d (%d), channelCfg: %d", objectType,
192 samplingIdx, rtpmp4apay->rate, channelCfg);
194 gst_buffer_unmap (buffer, &map);
201 GST_ELEMENT_ERROR (rtpmp4apay, STREAM, FORMAT,
203 ("config string too short, expected 2 bytes, got %" G_GSIZE_FORMAT,
205 gst_buffer_unmap (buffer, &map);
210 GST_ELEMENT_ERROR (rtpmp4apay, STREAM, FORMAT,
211 (NULL), ("invalid object type 0"));
212 gst_buffer_unmap (buffer, &map);
217 GST_ELEMENT_ERROR (rtpmp4apay, STREAM, NOT_IMPLEMENTED,
218 (NULL), ("unsupported frequency index %d", samplingIdx));
219 gst_buffer_unmap (buffer, &map);
224 GST_ELEMENT_ERROR (rtpmp4apay, STREAM, NOT_IMPLEMENTED,
225 (NULL), ("unsupported number of channels %d, must < 8", channelCfg));
226 gst_buffer_unmap (buffer, &map);
232 gst_rtp_mp4a_pay_new_caps (GstRtpMP4APay * rtpmp4apay)
238 g_value_init (&v, GST_TYPE_BUFFER);
239 gst_value_set_buffer (&v, rtpmp4apay->config);
240 config = gst_value_serialize (&v);
242 res = gst_rtp_base_payload_set_outcaps (GST_RTP_BASE_PAYLOAD (rtpmp4apay),
243 "cpresent", G_TYPE_STRING, "0", "config", G_TYPE_STRING, config, NULL);
252 gst_rtp_mp4a_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
254 GstRtpMP4APay *rtpmp4apay;
255 GstStructure *structure;
256 const GValue *codec_data;
257 gboolean res, framed = TRUE;
258 const gchar *stream_format;
260 rtpmp4apay = GST_RTP_MP4A_PAY (payload);
262 structure = gst_caps_get_structure (caps, 0);
264 /* this is already handled by the template caps, but it is better
265 * to leave here to have meaningful warning messages when linking
267 stream_format = gst_structure_get_string (structure, "stream-format");
269 if (strcmp (stream_format, "raw") != 0) {
270 GST_WARNING_OBJECT (rtpmp4apay, "AAC's stream-format must be 'raw', "
271 "%s is not supported", stream_format);
275 GST_WARNING_OBJECT (rtpmp4apay, "AAC's stream-format not specified, "
279 codec_data = gst_structure_get_value (structure, "codec_data");
281 GST_LOG_OBJECT (rtpmp4apay, "got codec_data");
282 if (G_VALUE_TYPE (codec_data) == GST_TYPE_BUFFER) {
283 GstBuffer *buffer, *cbuffer;
288 buffer = gst_value_get_buffer (codec_data);
289 GST_LOG_OBJECT (rtpmp4apay, "configuring codec_data");
292 res = gst_rtp_mp4a_pay_parse_audio_config (rtpmp4apay, buffer);
297 gst_buffer_map (buffer, &map, GST_MAP_READ);
299 /* make the StreamMuxConfig, we need 15 bits for the header */
300 cbuffer = gst_buffer_new_and_alloc (map.size + 2);
301 gst_buffer_map (cbuffer, &cmap, GST_MAP_WRITE);
303 memset (cmap.data, 0, map.size + 2);
305 /* Create StreamMuxConfig according to ISO/IEC 14496-3:
307 * audioMuxVersion == 0 (1 bit)
308 * allStreamsSameTimeFraming == 1 (1 bit)
309 * numSubFrames == numSubFrames (6 bits)
310 * numProgram == 0 (4 bits)
311 * numLayer == 0 (3 bits)
316 /* append the config bits, shifting them 1 bit left */
317 for (i = 0; i < map.size; i++) {
318 cmap.data[i + 1] |= ((map.data[i] & 0x80) >> 7);
319 cmap.data[i + 2] |= ((map.data[i] & 0x7f) << 1);
322 gst_buffer_unmap (cbuffer, &cmap);
323 gst_buffer_unmap (buffer, &map);
325 /* now we can configure the buffer */
326 if (rtpmp4apay->config)
327 gst_buffer_unref (rtpmp4apay->config);
328 rtpmp4apay->config = cbuffer;
332 if (gst_structure_get_boolean (structure, "framed", &framed) && !framed) {
333 GST_WARNING_OBJECT (payload, "Need framed AAC data as input!");
336 gst_rtp_base_payload_set_options (payload, "audio", TRUE, "MP4A-LATM",
339 res = gst_rtp_mp4a_pay_new_caps (rtpmp4apay);
346 GST_DEBUG_OBJECT (rtpmp4apay, "failed to parse config");
351 #define RTP_HEADER_LEN 12
353 /* we expect buffers as exactly one complete AU
356 gst_rtp_mp4a_pay_handle_buffer (GstRTPBasePayload * basepayload,
359 GstRtpMP4APay *rtpmp4apay;
366 GstClockTime timestamp;
370 rtpmp4apay = GST_RTP_MP4A_PAY (basepayload);
373 size = gst_buffer_get_size (buffer);
375 timestamp = GST_BUFFER_PTS (buffer);
378 mtu = GST_RTP_BASE_PAYLOAD_MTU (rtpmp4apay);
380 list = gst_buffer_list_new_sized (size / (mtu - RTP_HEADER_LEN) + 1);
389 GstRTPBuffer rtp = { NULL };
394 /* first packet calculate space for the packet including the header */
396 while (count >= 0xff) {
403 packet_len = gst_rtp_buffer_calc_packet_len (header_len + size, 0, 0);
404 towrite = MIN (packet_len, mtu);
405 payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
406 payload_len -= header_len;
408 GST_DEBUG_OBJECT (rtpmp4apay,
409 "avail %" G_GSIZE_FORMAT
410 ", header_len %d, packet_len %d, payload_len %d", size, header_len,
411 packet_len, payload_len);
413 /* create buffer to hold the payload. */
414 outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload,
418 gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
421 guint8 *payload = gst_rtp_buffer_get_payload (&rtp);
424 /* first packet write the header */
426 while (count >= 0xff) {
433 /* marker only if the packet is complete */
434 gst_rtp_buffer_set_marker (&rtp, size == payload_len);
435 if (size == payload_len)
436 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_MARKER);
438 gst_rtp_buffer_unmap (&rtp);
440 /* create a new buf to hold the payload */
441 paybuf = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_ALL,
442 offset, payload_len);
444 /* join memory parts */
445 gst_rtp_copy_audio_meta (rtpmp4apay, outbuf, paybuf);
446 outbuf = gst_buffer_append (outbuf, paybuf);
447 gst_buffer_list_add (list, outbuf);
448 offset += payload_len;
451 /* copy incoming timestamp (if any) to outgoing buffers */
452 GST_BUFFER_PTS (outbuf) = timestamp;
458 gst_rtp_base_payload_push_list (GST_RTP_BASE_PAYLOAD (rtpmp4apay), list);
460 gst_buffer_unref (buffer);