Mercurial > libavformat.hg
annotate rtp.c @ 456:c13fd248e222 libavformat
match latest spec
seeking to startcode-less frames
author | michael |
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date | Sun, 02 May 2004 01:03:55 +0000 |
parents | 845f9de2c883 |
children | 60f897e8dd2d |
rev | line source |
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0 | 1 /* |
2 * RTP input/output format | |
3 * Copyright (c) 2002 Fabrice Bellard. | |
4 * | |
5 * This library is free software; you can redistribute it and/or | |
6 * modify it under the terms of the GNU Lesser General Public | |
7 * License as published by the Free Software Foundation; either | |
8 * version 2 of the License, or (at your option) any later version. | |
9 * | |
10 * This library is distributed in the hope that it will be useful, | |
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of | |
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
13 * Lesser General Public License for more details. | |
14 * | |
15 * You should have received a copy of the GNU Lesser General Public | |
16 * License along with this library; if not, write to the Free Software | |
17 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA | |
18 */ | |
19 #include "avformat.h" | |
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20 #include "mpegts.h" |
0 | 21 |
22 #include <unistd.h> | |
23 #include <sys/types.h> | |
24 #include <sys/socket.h> | |
25 #include <netinet/in.h> | |
26 #ifndef __BEOS__ | |
27 # include <arpa/inet.h> | |
28 #else | |
29 # include "barpainet.h" | |
30 #endif | |
31 #include <netdb.h> | |
32 | |
33 //#define DEBUG | |
34 | |
35 | |
36 /* TODO: - add RTCP statistics reporting (should be optional). | |
37 | |
38 - add support for h263/mpeg4 packetized output : IDEA: send a | |
39 buffer to 'rtp_write_packet' contains all the packets for ONE | |
40 frame. Each packet should have a four byte header containing | |
41 the length in big endian format (same trick as | |
42 'url_open_dyn_packet_buf') | |
43 */ | |
44 | |
45 #define RTP_VERSION 2 | |
46 | |
47 #define RTP_MAX_SDES 256 /* maximum text length for SDES */ | |
48 | |
49 /* RTCP paquets use 0.5 % of the bandwidth */ | |
50 #define RTCP_TX_RATIO_NUM 5 | |
51 #define RTCP_TX_RATIO_DEN 1000 | |
52 | |
53 typedef enum { | |
54 RTCP_SR = 200, | |
55 RTCP_RR = 201, | |
56 RTCP_SDES = 202, | |
57 RTCP_BYE = 203, | |
58 RTCP_APP = 204 | |
59 } rtcp_type_t; | |
60 | |
61 typedef enum { | |
62 RTCP_SDES_END = 0, | |
63 RTCP_SDES_CNAME = 1, | |
64 RTCP_SDES_NAME = 2, | |
65 RTCP_SDES_EMAIL = 3, | |
66 RTCP_SDES_PHONE = 4, | |
67 RTCP_SDES_LOC = 5, | |
68 RTCP_SDES_TOOL = 6, | |
69 RTCP_SDES_NOTE = 7, | |
70 RTCP_SDES_PRIV = 8, | |
71 RTCP_SDES_IMG = 9, | |
72 RTCP_SDES_DOOR = 10, | |
73 RTCP_SDES_SOURCE = 11 | |
74 } rtcp_sdes_type_t; | |
75 | |
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76 struct RTPDemuxContext { |
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77 AVFormatContext *ic; |
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78 AVStream *st; |
0 | 79 int payload_type; |
65 | 80 uint32_t ssrc; |
81 uint16_t seq; | |
82 uint32_t timestamp; | |
83 uint32_t base_timestamp; | |
84 uint32_t cur_timestamp; | |
0 | 85 int max_payload_size; |
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86 MpegTSContext *ts; /* only used for RTP_PT_MPEG2TS payloads */ |
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87 int read_buf_index; |
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88 int read_buf_size; |
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89 |
0 | 90 /* rtcp sender statistics receive */ |
65 | 91 int64_t last_rtcp_ntp_time; |
173 | 92 int64_t first_rtcp_ntp_time; |
65 | 93 uint32_t last_rtcp_timestamp; |
0 | 94 /* rtcp sender statistics */ |
95 unsigned int packet_count; | |
96 unsigned int octet_count; | |
97 unsigned int last_octet_count; | |
98 int first_packet; | |
99 /* buffer for output */ | |
65 | 100 uint8_t buf[RTP_MAX_PACKET_LENGTH]; |
101 uint8_t *buf_ptr; | |
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102 }; |
0 | 103 |
104 int rtp_get_codec_info(AVCodecContext *codec, int payload_type) | |
105 { | |
106 switch(payload_type) { | |
107 case RTP_PT_ULAW: | |
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108 codec->codec_type = CODEC_TYPE_AUDIO; |
0 | 109 codec->codec_id = CODEC_ID_PCM_MULAW; |
110 codec->channels = 1; | |
111 codec->sample_rate = 8000; | |
112 break; | |
113 case RTP_PT_ALAW: | |
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114 codec->codec_type = CODEC_TYPE_AUDIO; |
0 | 115 codec->codec_id = CODEC_ID_PCM_ALAW; |
116 codec->channels = 1; | |
117 codec->sample_rate = 8000; | |
118 break; | |
119 case RTP_PT_S16BE_STEREO: | |
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120 codec->codec_type = CODEC_TYPE_AUDIO; |
0 | 121 codec->codec_id = CODEC_ID_PCM_S16BE; |
122 codec->channels = 2; | |
123 codec->sample_rate = 44100; | |
124 break; | |
125 case RTP_PT_S16BE_MONO: | |
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126 codec->codec_type = CODEC_TYPE_AUDIO; |
0 | 127 codec->codec_id = CODEC_ID_PCM_S16BE; |
128 codec->channels = 1; | |
129 codec->sample_rate = 44100; | |
130 break; | |
131 case RTP_PT_MPEGAUDIO: | |
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132 codec->codec_type = CODEC_TYPE_AUDIO; |
0 | 133 codec->codec_id = CODEC_ID_MP2; |
134 break; | |
135 case RTP_PT_JPEG: | |
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136 codec->codec_type = CODEC_TYPE_VIDEO; |
0 | 137 codec->codec_id = CODEC_ID_MJPEG; |
138 break; | |
139 case RTP_PT_MPEGVIDEO: | |
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140 codec->codec_type = CODEC_TYPE_VIDEO; |
0 | 141 codec->codec_id = CODEC_ID_MPEG1VIDEO; |
142 break; | |
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143 case RTP_PT_MPEG2TS: |
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144 codec->codec_type = CODEC_TYPE_DATA; |
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145 codec->codec_id = CODEC_ID_MPEG2TS; |
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146 break; |
0 | 147 default: |
148 return -1; | |
149 } | |
150 return 0; | |
151 } | |
152 | |
153 /* return < 0 if unknown payload type */ | |
154 int rtp_get_payload_type(AVCodecContext *codec) | |
155 { | |
156 int payload_type; | |
157 | |
158 /* compute the payload type */ | |
159 payload_type = -1; | |
160 switch(codec->codec_id) { | |
161 case CODEC_ID_PCM_MULAW: | |
162 payload_type = RTP_PT_ULAW; | |
163 break; | |
164 case CODEC_ID_PCM_ALAW: | |
165 payload_type = RTP_PT_ALAW; | |
166 break; | |
167 case CODEC_ID_PCM_S16BE: | |
168 if (codec->channels == 1) { | |
169 payload_type = RTP_PT_S16BE_MONO; | |
170 } else if (codec->channels == 2) { | |
171 payload_type = RTP_PT_S16BE_STEREO; | |
172 } | |
173 break; | |
174 case CODEC_ID_MP2: | |
232 | 175 case CODEC_ID_MP3: |
0 | 176 payload_type = RTP_PT_MPEGAUDIO; |
177 break; | |
178 case CODEC_ID_MJPEG: | |
179 payload_type = RTP_PT_JPEG; | |
180 break; | |
181 case CODEC_ID_MPEG1VIDEO: | |
182 payload_type = RTP_PT_MPEGVIDEO; | |
183 break; | |
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184 case CODEC_ID_MPEG2TS: |
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185 payload_type = RTP_PT_MPEG2TS; |
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186 break; |
0 | 187 default: |
188 break; | |
189 } | |
190 return payload_type; | |
191 } | |
192 | |
65 | 193 static inline uint32_t decode_be32(const uint8_t *p) |
0 | 194 { |
195 return (p[0] << 24) | (p[1] << 16) | (p[2] << 8) | p[3]; | |
196 } | |
197 | |
173 | 198 static inline uint64_t decode_be64(const uint8_t *p) |
0 | 199 { |
65 | 200 return ((uint64_t)decode_be32(p) << 32) | decode_be32(p + 4); |
0 | 201 } |
202 | |
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203 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len) |
0 | 204 { |
205 if (buf[1] != 200) | |
206 return -1; | |
207 s->last_rtcp_ntp_time = decode_be64(buf + 8); | |
173 | 208 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) |
209 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time; | |
0 | 210 s->last_rtcp_timestamp = decode_be32(buf + 16); |
211 return 0; | |
212 } | |
213 | |
214 /** | |
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215 * open a new RTP parse context for stream 'st'. 'st' can be NULL for |
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216 * MPEG2TS streams to indicate that they should be demuxed inside the |
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217 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned) |
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218 */ |
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219 RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type) |
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220 { |
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221 RTPDemuxContext *s; |
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222 |
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223 s = av_mallocz(sizeof(RTPDemuxContext)); |
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224 if (!s) |
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225 return NULL; |
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226 s->payload_type = payload_type; |
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227 s->last_rtcp_ntp_time = AV_NOPTS_VALUE; |
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228 s->first_rtcp_ntp_time = AV_NOPTS_VALUE; |
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229 s->ic = s1; |
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230 s->st = st; |
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231 if (payload_type == RTP_PT_MPEG2TS) { |
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232 s->ts = mpegts_parse_open(s->ic); |
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233 if (s->ts == NULL) { |
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234 av_free(s); |
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235 return NULL; |
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236 } |
305 | 237 } else { |
238 switch(st->codec.codec_id) { | |
239 case CODEC_ID_MPEG1VIDEO: | |
240 case CODEC_ID_MPEG2VIDEO: | |
241 case CODEC_ID_MP2: | |
242 case CODEC_ID_MP3: | |
243 case CODEC_ID_MPEG4: | |
244 st->need_parsing = 1; | |
245 break; | |
246 default: | |
247 break; | |
248 } | |
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249 } |
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250 return s; |
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251 } |
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252 |
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253 /** |
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254 * Parse an RTP or RTCP packet directly sent as a buffer. |
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255 * @param s RTP parse context. |
0 | 256 * @param pkt returned packet |
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257 * @param buf input buffer or NULL to read the next packets |
0 | 258 * @param len buffer len |
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259 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow |
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260 * (use buf as NULL to read the next). -1 if no packet (error or no more packet). |
0 | 261 */ |
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262 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, |
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263 const uint8_t *buf, int len) |
0 | 264 { |
265 unsigned int ssrc, h; | |
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266 int payload_type, seq, delta_timestamp, ret; |
0 | 267 AVStream *st; |
65 | 268 uint32_t timestamp; |
0 | 269 |
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270 if (!buf) { |
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271 /* return the next packets, if any */ |
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272 if (s->read_buf_index >= s->read_buf_size) |
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273 return -1; |
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274 ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index, |
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275 s->read_buf_size - s->read_buf_index); |
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276 if (ret < 0) |
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277 return -1; |
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278 s->read_buf_index += ret; |
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279 if (s->read_buf_index < s->read_buf_size) |
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280 return 1; |
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281 else |
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282 return 0; |
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283 } |
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284 |
0 | 285 if (len < 12) |
286 return -1; | |
287 | |
288 if ((buf[0] & 0xc0) != (RTP_VERSION << 6)) | |
289 return -1; | |
290 if (buf[1] >= 200 && buf[1] <= 204) { | |
294
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291 rtcp_parse_packet(s, buf, len); |
0 | 292 return -1; |
293 } | |
294 payload_type = buf[1] & 0x7f; | |
295 seq = (buf[2] << 8) | buf[3]; | |
296 timestamp = decode_be32(buf + 4); | |
297 ssrc = decode_be32(buf + 8); | |
298 | |
299 /* NOTE: we can handle only one payload type */ | |
300 if (s->payload_type != payload_type) | |
301 return -1; | |
302 #if defined(DEBUG) || 1 | |
303 if (seq != ((s->seq + 1) & 0xffff)) { | |
370
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304 av_log(&s->st->codec, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n", |
0 | 305 payload_type, seq, ((s->seq + 1) & 0xffff)); |
306 } | |
307 s->seq = seq; | |
308 #endif | |
309 len -= 12; | |
310 buf += 12; | |
294
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311 |
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312 st = s->st; |
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313 if (!st) { |
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314 /* specific MPEG2TS demux support */ |
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315 ret = mpegts_parse_packet(s->ts, pkt, buf, len); |
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316 if (ret < 0) |
0 | 317 return -1; |
294
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318 if (ret < len) { |
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319 s->read_buf_size = len - ret; |
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320 memcpy(s->buf, buf + ret, s->read_buf_size); |
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321 s->read_buf_index = 0; |
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322 return 1; |
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323 } |
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324 } else { |
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325 switch(st->codec.codec_id) { |
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326 case CODEC_ID_MP2: |
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327 /* better than nothing: skip mpeg audio RTP header */ |
0 | 328 if (len <= 4) |
329 return -1; | |
294
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330 h = decode_be32(buf); |
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331 len -= 4; |
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332 buf += 4; |
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333 av_new_packet(pkt, len); |
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334 memcpy(pkt->data, buf, len); |
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335 break; |
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336 case CODEC_ID_MPEG1VIDEO: |
305 | 337 /* better than nothing: skip mpeg video RTP header */ |
294
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338 if (len <= 4) |
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339 return -1; |
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340 h = decode_be32(buf); |
0 | 341 buf += 4; |
342 len -= 4; | |
294
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343 if (h & (1 << 26)) { |
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344 /* mpeg2 */ |
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345 if (len <= 4) |
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346 return -1; |
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347 buf += 4; |
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348 len -= 4; |
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349 } |
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350 av_new_packet(pkt, len); |
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351 memcpy(pkt->data, buf, len); |
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352 break; |
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353 default: |
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354 av_new_packet(pkt, len); |
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355 memcpy(pkt->data, buf, len); |
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356 break; |
0 | 357 } |
294
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358 |
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359 switch(st->codec.codec_id) { |
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360 case CODEC_ID_MP2: |
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361 case CODEC_ID_MPEG1VIDEO: |
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362 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) { |
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363 int64_t addend; |
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364 /* XXX: is it really necessary to unify the timestamp base ? */ |
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365 /* compute pts from timestamp with received ntp_time */ |
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366 delta_timestamp = timestamp - s->last_rtcp_timestamp; |
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367 /* convert to 90 kHz without overflow */ |
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368 addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14; |
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369 addend = (addend * 5625) >> 14; |
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370 pkt->pts = addend + delta_timestamp; |
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371 } |
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372 break; |
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373 default: |
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374 /* no timestamp info yet */ |
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375 break; |
173 | 376 } |
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377 pkt->stream_index = s->st->index; |
0 | 378 } |
379 return 0; | |
380 } | |
381 | |
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382 void rtp_parse_close(RTPDemuxContext *s) |
0 | 383 { |
294
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384 if (s->payload_type == RTP_PT_MPEG2TS) { |
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385 mpegts_parse_close(s->ts); |
0 | 386 } |
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387 av_free(s); |
0 | 388 } |
389 | |
390 /* rtp output */ | |
391 | |
392 static int rtp_write_header(AVFormatContext *s1) | |
393 { | |
294
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394 RTPDemuxContext *s = s1->priv_data; |
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395 int payload_type, max_packet_size, n; |
0 | 396 AVStream *st; |
397 | |
398 if (s1->nb_streams != 1) | |
399 return -1; | |
400 st = s1->streams[0]; | |
401 | |
402 payload_type = rtp_get_payload_type(&st->codec); | |
403 if (payload_type < 0) | |
404 payload_type = RTP_PT_PRIVATE; /* private payload type */ | |
405 s->payload_type = payload_type; | |
406 | |
407 s->base_timestamp = random(); | |
408 s->timestamp = s->base_timestamp; | |
409 s->ssrc = random(); | |
410 s->first_packet = 1; | |
411 | |
412 max_packet_size = url_fget_max_packet_size(&s1->pb); | |
413 if (max_packet_size <= 12) | |
414 return AVERROR_IO; | |
415 s->max_payload_size = max_packet_size - 12; | |
416 | |
417 switch(st->codec.codec_id) { | |
418 case CODEC_ID_MP2: | |
232 | 419 case CODEC_ID_MP3: |
0 | 420 s->buf_ptr = s->buf + 4; |
421 s->cur_timestamp = 0; | |
422 break; | |
423 case CODEC_ID_MPEG1VIDEO: | |
424 s->cur_timestamp = 0; | |
425 break; | |
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426 case CODEC_ID_MPEG2TS: |
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427 n = s->max_payload_size / TS_PACKET_SIZE; |
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428 if (n < 1) |
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429 n = 1; |
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430 s->max_payload_size = n * TS_PACKET_SIZE; |
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431 s->buf_ptr = s->buf; |
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432 break; |
0 | 433 default: |
434 s->buf_ptr = s->buf; | |
435 break; | |
436 } | |
437 | |
438 return 0; | |
439 } | |
440 | |
441 /* send an rtcp sender report packet */ | |
65 | 442 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time) |
0 | 443 { |
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444 RTPDemuxContext *s = s1->priv_data; |
0 | 445 #if defined(DEBUG) |
446 printf("RTCP: %02x %Lx %x\n", s->payload_type, ntp_time, s->timestamp); | |
447 #endif | |
448 put_byte(&s1->pb, (RTP_VERSION << 6)); | |
449 put_byte(&s1->pb, 200); | |
450 put_be16(&s1->pb, 6); /* length in words - 1 */ | |
451 put_be32(&s1->pb, s->ssrc); | |
452 put_be64(&s1->pb, ntp_time); | |
453 put_be32(&s1->pb, s->timestamp); | |
454 put_be32(&s1->pb, s->packet_count); | |
455 put_be32(&s1->pb, s->octet_count); | |
456 put_flush_packet(&s1->pb); | |
457 } | |
458 | |
459 /* send an rtp packet. sequence number is incremented, but the caller | |
460 must update the timestamp itself */ | |
241 | 461 static void rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len) |
0 | 462 { |
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463 RTPDemuxContext *s = s1->priv_data; |
0 | 464 |
465 #ifdef DEBUG | |
466 printf("rtp_send_data size=%d\n", len); | |
467 #endif | |
468 | |
469 /* build the RTP header */ | |
470 put_byte(&s1->pb, (RTP_VERSION << 6)); | |
471 put_byte(&s1->pb, s->payload_type & 0x7f); | |
472 put_be16(&s1->pb, s->seq); | |
473 put_be32(&s1->pb, s->timestamp); | |
474 put_be32(&s1->pb, s->ssrc); | |
475 | |
476 put_buffer(&s1->pb, buf1, len); | |
477 put_flush_packet(&s1->pb); | |
478 | |
479 s->seq++; | |
480 s->octet_count += len; | |
481 s->packet_count++; | |
482 } | |
483 | |
484 /* send an integer number of samples and compute time stamp and fill | |
485 the rtp send buffer before sending. */ | |
486 static void rtp_send_samples(AVFormatContext *s1, | |
241 | 487 const uint8_t *buf1, int size, int sample_size) |
0 | 488 { |
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489 RTPDemuxContext *s = s1->priv_data; |
0 | 490 int len, max_packet_size, n; |
491 | |
492 max_packet_size = (s->max_payload_size / sample_size) * sample_size; | |
493 /* not needed, but who nows */ | |
494 if ((size % sample_size) != 0) | |
495 av_abort(); | |
496 while (size > 0) { | |
497 len = (max_packet_size - (s->buf_ptr - s->buf)); | |
498 if (len > size) | |
499 len = size; | |
500 | |
501 /* copy data */ | |
502 memcpy(s->buf_ptr, buf1, len); | |
503 s->buf_ptr += len; | |
504 buf1 += len; | |
505 size -= len; | |
506 n = (s->buf_ptr - s->buf); | |
507 /* if buffer full, then send it */ | |
508 if (n >= max_packet_size) { | |
509 rtp_send_data(s1, s->buf, n); | |
510 s->buf_ptr = s->buf; | |
511 /* update timestamp */ | |
512 s->timestamp += n / sample_size; | |
513 } | |
514 } | |
515 } | |
516 | |
517 /* NOTE: we suppose that exactly one frame is given as argument here */ | |
518 /* XXX: test it */ | |
519 static void rtp_send_mpegaudio(AVFormatContext *s1, | |
241 | 520 const uint8_t *buf1, int size) |
0 | 521 { |
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522 RTPDemuxContext *s = s1->priv_data; |
0 | 523 AVStream *st = s1->streams[0]; |
524 int len, count, max_packet_size; | |
525 | |
526 max_packet_size = s->max_payload_size; | |
527 | |
528 /* test if we must flush because not enough space */ | |
529 len = (s->buf_ptr - s->buf); | |
530 if ((len + size) > max_packet_size) { | |
531 if (len > 4) { | |
532 rtp_send_data(s1, s->buf, s->buf_ptr - s->buf); | |
533 s->buf_ptr = s->buf + 4; | |
534 /* 90 KHz time stamp */ | |
535 s->timestamp = s->base_timestamp + | |
536 (s->cur_timestamp * 90000LL) / st->codec.sample_rate; | |
537 } | |
538 } | |
539 | |
540 /* add the packet */ | |
541 if (size > max_packet_size) { | |
542 /* big packet: fragment */ | |
543 count = 0; | |
544 while (size > 0) { | |
545 len = max_packet_size - 4; | |
546 if (len > size) | |
547 len = size; | |
548 /* build fragmented packet */ | |
549 s->buf[0] = 0; | |
550 s->buf[1] = 0; | |
551 s->buf[2] = count >> 8; | |
552 s->buf[3] = count; | |
553 memcpy(s->buf + 4, buf1, len); | |
554 rtp_send_data(s1, s->buf, len + 4); | |
555 size -= len; | |
556 buf1 += len; | |
557 count += len; | |
558 } | |
559 } else { | |
560 if (s->buf_ptr == s->buf + 4) { | |
561 /* no fragmentation possible */ | |
562 s->buf[0] = 0; | |
563 s->buf[1] = 0; | |
564 s->buf[2] = 0; | |
565 s->buf[3] = 0; | |
566 } | |
567 memcpy(s->buf_ptr, buf1, size); | |
568 s->buf_ptr += size; | |
569 } | |
570 s->cur_timestamp += st->codec.frame_size; | |
571 } | |
572 | |
573 /* NOTE: a single frame must be passed with sequence header if | |
574 needed. XXX: use slices. */ | |
575 static void rtp_send_mpegvideo(AVFormatContext *s1, | |
241 | 576 const uint8_t *buf1, int size) |
0 | 577 { |
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578 RTPDemuxContext *s = s1->priv_data; |
0 | 579 AVStream *st = s1->streams[0]; |
580 int len, h, max_packet_size; | |
65 | 581 uint8_t *q; |
0 | 582 |
583 max_packet_size = s->max_payload_size; | |
584 | |
585 while (size > 0) { | |
586 /* XXX: more correct headers */ | |
587 h = 0; | |
588 if (st->codec.sub_id == 2) | |
589 h |= 1 << 26; /* mpeg 2 indicator */ | |
590 q = s->buf; | |
591 *q++ = h >> 24; | |
592 *q++ = h >> 16; | |
593 *q++ = h >> 8; | |
594 *q++ = h; | |
595 | |
596 if (st->codec.sub_id == 2) { | |
597 h = 0; | |
598 *q++ = h >> 24; | |
599 *q++ = h >> 16; | |
600 *q++ = h >> 8; | |
601 *q++ = h; | |
602 } | |
603 | |
604 len = max_packet_size - (q - s->buf); | |
605 if (len > size) | |
606 len = size; | |
607 | |
608 memcpy(q, buf1, len); | |
609 q += len; | |
610 | |
611 /* 90 KHz time stamp */ | |
612 s->timestamp = s->base_timestamp + | |
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613 av_rescale((int64_t)s->cur_timestamp * st->codec.frame_rate_base, 90000, st->codec.frame_rate); |
0 | 614 rtp_send_data(s1, s->buf, q - s->buf); |
615 | |
616 buf1 += len; | |
617 size -= len; | |
618 } | |
619 s->cur_timestamp++; | |
620 } | |
621 | |
622 static void rtp_send_raw(AVFormatContext *s1, | |
241 | 623 const uint8_t *buf1, int size) |
0 | 624 { |
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625 RTPDemuxContext *s = s1->priv_data; |
0 | 626 AVStream *st = s1->streams[0]; |
627 int len, max_packet_size; | |
628 | |
629 max_packet_size = s->max_payload_size; | |
630 | |
631 while (size > 0) { | |
632 len = max_packet_size; | |
633 if (len > size) | |
634 len = size; | |
635 | |
636 /* 90 KHz time stamp */ | |
637 s->timestamp = s->base_timestamp + | |
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638 av_rescale((int64_t)s->cur_timestamp * st->codec.frame_rate_base, 90000, st->codec.frame_rate); |
0 | 639 rtp_send_data(s1, buf1, len); |
640 | |
641 buf1 += len; | |
642 size -= len; | |
643 } | |
644 s->cur_timestamp++; | |
645 } | |
646 | |
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647 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */ |
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648 static void rtp_send_mpegts_raw(AVFormatContext *s1, |
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649 const uint8_t *buf1, int size) |
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650 { |
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651 RTPDemuxContext *s = s1->priv_data; |
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652 int len, out_len; |
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653 |
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654 while (size >= TS_PACKET_SIZE) { |
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655 len = s->max_payload_size - (s->buf_ptr - s->buf); |
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656 if (len > size) |
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657 len = size; |
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658 memcpy(s->buf_ptr, buf1, len); |
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659 buf1 += len; |
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660 size -= len; |
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661 s->buf_ptr += len; |
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662 |
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663 out_len = s->buf_ptr - s->buf; |
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664 if (out_len >= s->max_payload_size) { |
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665 rtp_send_data(s1, s->buf, out_len); |
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666 s->buf_ptr = s->buf; |
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667 } |
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668 } |
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669 } |
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670 |
0 | 671 /* write an RTP packet. 'buf1' must contain a single specific frame. */ |
672 static int rtp_write_packet(AVFormatContext *s1, int stream_index, | |
241 | 673 const uint8_t *buf1, int size, int64_t pts) |
0 | 674 { |
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675 RTPDemuxContext *s = s1->priv_data; |
0 | 676 AVStream *st = s1->streams[0]; |
677 int rtcp_bytes; | |
65 | 678 int64_t ntp_time; |
0 | 679 |
680 #ifdef DEBUG | |
681 printf("%d: write len=%d\n", stream_index, size); | |
682 #endif | |
683 | |
684 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */ | |
685 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / | |
686 RTCP_TX_RATIO_DEN; | |
687 if (s->first_packet || rtcp_bytes >= 28) { | |
688 /* compute NTP time */ | |
173 | 689 /* XXX: 90 kHz timestamp hardcoded */ |
241 | 690 ntp_time = (pts << 28) / 5625; |
0 | 691 rtcp_send_sr(s1, ntp_time); |
692 s->last_octet_count = s->octet_count; | |
693 s->first_packet = 0; | |
694 } | |
695 | |
696 switch(st->codec.codec_id) { | |
697 case CODEC_ID_PCM_MULAW: | |
698 case CODEC_ID_PCM_ALAW: | |
699 case CODEC_ID_PCM_U8: | |
700 case CODEC_ID_PCM_S8: | |
701 rtp_send_samples(s1, buf1, size, 1 * st->codec.channels); | |
702 break; | |
703 case CODEC_ID_PCM_U16BE: | |
704 case CODEC_ID_PCM_U16LE: | |
705 case CODEC_ID_PCM_S16BE: | |
706 case CODEC_ID_PCM_S16LE: | |
707 rtp_send_samples(s1, buf1, size, 2 * st->codec.channels); | |
708 break; | |
709 case CODEC_ID_MP2: | |
232 | 710 case CODEC_ID_MP3: |
0 | 711 rtp_send_mpegaudio(s1, buf1, size); |
712 break; | |
713 case CODEC_ID_MPEG1VIDEO: | |
714 rtp_send_mpegvideo(s1, buf1, size); | |
715 break; | |
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716 case CODEC_ID_MPEG2TS: |
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717 rtp_send_mpegts_raw(s1, buf1, size); |
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718 break; |
0 | 719 default: |
720 /* better than nothing : send the codec raw data */ | |
721 rtp_send_raw(s1, buf1, size); | |
722 break; | |
723 } | |
724 return 0; | |
725 } | |
726 | |
727 static int rtp_write_trailer(AVFormatContext *s1) | |
728 { | |
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729 // RTPDemuxContext *s = s1->priv_data; |
0 | 730 return 0; |
731 } | |
732 | |
733 AVOutputFormat rtp_mux = { | |
734 "rtp", | |
735 "RTP output format", | |
736 NULL, | |
737 NULL, | |
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738 sizeof(RTPDemuxContext), |
0 | 739 CODEC_ID_PCM_MULAW, |
740 CODEC_ID_NONE, | |
741 rtp_write_header, | |
742 rtp_write_packet, | |
743 rtp_write_trailer, | |
744 }; | |
745 | |
746 int rtp_init(void) | |
747 { | |
748 av_register_output_format(&rtp_mux); | |
749 return 0; | |
750 } |