Mercurial > libavformat.hg
annotate rtp.c @ 1534:4bf9dc98e127 libavformat
add support for tta in matroska
author | aurel |
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date | Mon, 20 Nov 2006 20:16:32 +0000 |
parents | 49d5a5ca2987 |
children | 92afee454599 |
rev | line source |
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0 | 1 /* |
2 * RTP input/output format | |
3 * Copyright (c) 2002 Fabrice Bellard. | |
4 * | |
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5 * This file is part of FFmpeg. |
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6 * |
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7 * FFmpeg is free software; you can redistribute it and/or |
0 | 8 * modify it under the terms of the GNU Lesser General Public |
9 * License as published by the Free Software Foundation; either | |
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10 * version 2.1 of the License, or (at your option) any later version. |
0 | 11 * |
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12 * FFmpeg is distributed in the hope that it will be useful, |
0 | 13 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
15 * Lesser General Public License for more details. | |
16 * | |
17 * You should have received a copy of the GNU Lesser General Public | |
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18 * License along with FFmpeg; if not, write to the Free Software |
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19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
0 | 20 */ |
21 #include "avformat.h" | |
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22 #include "mpegts.h" |
774 | 23 #include "bitstream.h" |
0 | 24 |
25 #include <unistd.h> | |
26 #include <sys/types.h> | |
27 #include <sys/socket.h> | |
28 #include <netinet/in.h> | |
29 #ifndef __BEOS__ | |
30 # include <arpa/inet.h> | |
31 #else | |
32 # include "barpainet.h" | |
33 #endif | |
34 #include <netdb.h> | |
35 | |
1419 | 36 #include "rtp_internal.h" |
1460 | 37 #include "rtp_h264.h" |
1419 | 38 |
0 | 39 //#define DEBUG |
40 | |
41 | |
42 /* TODO: - add RTCP statistics reporting (should be optional). | |
43 | |
44 - add support for h263/mpeg4 packetized output : IDEA: send a | |
45 buffer to 'rtp_write_packet' contains all the packets for ONE | |
46 frame. Each packet should have a four byte header containing | |
47 the length in big endian format (same trick as | |
885 | 48 'url_open_dyn_packet_buf') |
0 | 49 */ |
50 | |
774 | 51 /* from http://www.iana.org/assignments/rtp-parameters last updated 05 January 2005 */ |
52 AVRtpPayloadType_t AVRtpPayloadTypes[]= | |
53 { | |
54 {0, "PCMU", CODEC_TYPE_AUDIO, CODEC_ID_PCM_MULAW, 8000, 1}, | |
55 {1, "Reserved", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
56 {2, "Reserved", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
57 {3, "GSM", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
58 {4, "G723", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
59 {5, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
60 {6, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 16000, 1}, | |
61 {7, "LPC", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
62 {8, "PCMA", CODEC_TYPE_AUDIO, CODEC_ID_PCM_ALAW, 8000, 1}, | |
63 {9, "G722", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
64 {10, "L16", CODEC_TYPE_AUDIO, CODEC_ID_PCM_S16BE, 44100, 2}, | |
65 {11, "L16", CODEC_TYPE_AUDIO, CODEC_ID_PCM_S16BE, 44100, 1}, | |
1259 | 66 {12, "QCELP", CODEC_TYPE_AUDIO, CODEC_ID_QCELP, 8000, 1}, |
774 | 67 {13, "CN", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, |
68 {14, "MPA", CODEC_TYPE_AUDIO, CODEC_ID_MP2, 90000, -1}, | |
69 {15, "G728", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
70 {16, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 11025, 1}, | |
71 {17, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 22050, 1}, | |
72 {18, "G729", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
73 {19, "reserved", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, | |
74 {20, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, | |
75 {21, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, | |
76 {22, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, | |
77 {23, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, | |
78 {24, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1}, | |
79 {25, "CelB", CODEC_TYPE_VIDEO, CODEC_ID_NONE, 90000, -1}, | |
80 {26, "JPEG", CODEC_TYPE_VIDEO, CODEC_ID_MJPEG, 90000, -1}, | |
81 {27, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1}, | |
82 {28, "nv", CODEC_TYPE_VIDEO, CODEC_ID_NONE, 90000, -1}, | |
83 {29, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1}, | |
84 {30, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1}, | |
85 {31, "H261", CODEC_TYPE_VIDEO, CODEC_ID_H261, 90000, -1}, | |
86 {32, "MPV", CODEC_TYPE_VIDEO, CODEC_ID_MPEG1VIDEO, 90000, -1}, | |
87 {33, "MP2T", CODEC_TYPE_DATA, CODEC_ID_MPEG2TS, 90000, -1}, | |
88 {34, "H263", CODEC_TYPE_VIDEO, CODEC_ID_H263, 90000, -1}, | |
89 {35, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
90 {36, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
91 {37, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
92 {38, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
93 {39, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
94 {40, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
95 {41, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
96 {42, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
97 {43, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
98 {44, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
99 {45, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
100 {46, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
101 {47, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
102 {48, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
103 {49, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
104 {50, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
105 {51, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
106 {52, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
107 {53, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
108 {54, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
109 {55, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
110 {56, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
111 {57, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
112 {58, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
113 {59, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
114 {60, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
115 {61, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
116 {62, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
117 {63, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
118 {64, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
119 {65, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
120 {66, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
121 {67, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
122 {68, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
123 {69, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
124 {70, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
125 {71, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
126 {72, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
127 {73, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
128 {74, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
129 {75, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
130 {76, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
131 {77, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
132 {78, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
133 {79, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
134 {80, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
135 {81, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
136 {82, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
137 {83, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
138 {84, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
139 {85, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
140 {86, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
141 {87, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
142 {88, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
143 {89, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
144 {90, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
145 {91, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
146 {92, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
147 {93, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
148 {94, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
149 {95, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
150 {96, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
151 {97, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
152 {98, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
153 {99, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
154 {100, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
155 {101, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
156 {102, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
157 {103, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
158 {104, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
159 {105, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
160 {106, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
161 {107, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
162 {108, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
163 {109, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
164 {110, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
165 {111, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
166 {112, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
167 {113, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
168 {114, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
169 {115, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
170 {116, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
171 {117, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
172 {118, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
173 {119, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
174 {120, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
175 {121, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
176 {122, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
177 {123, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
178 {124, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
179 {125, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
180 {126, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
181 {127, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
182 {-1, "", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1} | |
183 }; | |
0 | 184 |
1419 | 185 /* statistics functions */ |
186 RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL; | |
187 | |
188 static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4}; | |
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189 static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC}; |
0 | 190 |
1419 | 191 static void register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler) |
192 { | |
193 handler->next= RTPFirstDynamicPayloadHandler; | |
194 RTPFirstDynamicPayloadHandler= handler; | |
195 } | |
885 | 196 |
1419 | 197 void av_register_rtp_dynamic_payload_handlers() |
198 { | |
199 register_dynamic_payload_handler(&mp4v_es_handler); | |
200 register_dynamic_payload_handler(&mpeg4_generic_handler); | |
201 register_dynamic_payload_handler(&ff_h264_dynamic_handler); | |
202 } | |
0 | 203 |
204 int rtp_get_codec_info(AVCodecContext *codec, int payload_type) | |
205 { | |
774 | 206 if (AVRtpPayloadTypes[payload_type].codec_id != CODEC_ID_NONE) { |
207 codec->codec_type = AVRtpPayloadTypes[payload_type].codec_type; | |
1116 | 208 codec->codec_id = AVRtpPayloadTypes[payload_type].codec_id; |
774 | 209 if (AVRtpPayloadTypes[payload_type].audio_channels > 0) |
210 codec->channels = AVRtpPayloadTypes[payload_type].audio_channels; | |
211 if (AVRtpPayloadTypes[payload_type].clock_rate > 0) | |
212 codec->sample_rate = AVRtpPayloadTypes[payload_type].clock_rate; | |
213 return 0; | |
0 | 214 } |
774 | 215 return -1; |
0 | 216 } |
217 | |
218 /* return < 0 if unknown payload type */ | |
219 int rtp_get_payload_type(AVCodecContext *codec) | |
220 { | |
774 | 221 int i, payload_type; |
0 | 222 |
223 /* compute the payload type */ | |
774 | 224 for (payload_type = -1, i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i) |
225 if (AVRtpPayloadTypes[i].codec_id == codec->codec_id) { | |
226 if (codec->codec_id == CODEC_ID_PCM_S16BE) | |
227 if (codec->channels != AVRtpPayloadTypes[i].audio_channels) | |
228 continue; | |
229 payload_type = AVRtpPayloadTypes[i].pt; | |
0 | 230 } |
231 return payload_type; | |
232 } | |
233 | |
65 | 234 static inline uint32_t decode_be32(const uint8_t *p) |
0 | 235 { |
236 return (p[0] << 24) | (p[1] << 16) | (p[2] << 8) | p[3]; | |
237 } | |
238 | |
173 | 239 static inline uint64_t decode_be64(const uint8_t *p) |
0 | 240 { |
65 | 241 return ((uint64_t)decode_be32(p) << 32) | decode_be32(p + 4); |
0 | 242 } |
243 | |
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244 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len) |
0 | 245 { |
246 if (buf[1] != 200) | |
247 return -1; | |
248 s->last_rtcp_ntp_time = decode_be64(buf + 8); | |
173 | 249 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) |
250 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time; | |
0 | 251 s->last_rtcp_timestamp = decode_be32(buf + 16); |
252 return 0; | |
253 } | |
254 | |
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255 #define RTP_SEQ_MOD (1<<16) |
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256 |
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257 /** |
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258 * called on parse open packet |
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259 */ |
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260 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet. |
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261 { |
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262 memset(s, 0, sizeof(RTPStatistics)); |
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263 s->max_seq= base_sequence; |
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264 s->probation= 1; |
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265 } |
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266 |
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267 /** |
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268 * called whenever there is a large jump in sequence numbers, or when they get out of probation... |
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269 */ |
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270 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq) |
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271 { |
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272 s->max_seq= seq; |
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273 s->cycles= 0; |
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274 s->base_seq= seq -1; |
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275 s->bad_seq= RTP_SEQ_MOD + 1; |
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276 s->received= 0; |
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277 s->expected_prior= 0; |
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278 s->received_prior= 0; |
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279 s->jitter= 0; |
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280 s->transit= 0; |
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281 } |
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282 |
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283 /** |
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284 * returns 1 if we should handle this packet. |
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285 */ |
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286 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq) |
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287 { |
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288 uint16_t udelta= seq - s->max_seq; |
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289 const int MAX_DROPOUT= 3000; |
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290 const int MAX_MISORDER = 100; |
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291 const int MIN_SEQUENTIAL = 2; |
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292 |
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293 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */ |
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294 if(s->probation) |
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295 { |
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296 if(seq==s->max_seq + 1) { |
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297 s->probation--; |
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298 s->max_seq= seq; |
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299 if(s->probation==0) { |
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300 rtp_init_sequence(s, seq); |
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301 s->received++; |
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302 return 1; |
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303 } |
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304 } else { |
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305 s->probation= MIN_SEQUENTIAL - 1; |
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306 s->max_seq = seq; |
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307 } |
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308 } else if (udelta < MAX_DROPOUT) { |
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309 // in order, with permissible gap |
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310 if(seq < s->max_seq) { |
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311 //sequence number wrapped; count antother 64k cycles |
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312 s->cycles += RTP_SEQ_MOD; |
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313 } |
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314 s->max_seq= seq; |
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315 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) { |
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316 // sequence made a large jump... |
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317 if(seq==s->bad_seq) { |
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318 // two sequential packets-- assume that the other side restarted without telling us; just resync. |
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319 rtp_init_sequence(s, seq); |
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320 } else { |
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321 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1); |
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322 return 0; |
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323 } |
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324 } else { |
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325 // duplicate or reordered packet... |
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326 } |
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327 s->received++; |
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328 return 1; |
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329 } |
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330 |
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331 #if 0 |
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332 /** |
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333 * This function is currently unused; without a valid local ntp time, I don't see how we could calculate the |
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334 * difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values |
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335 * never change. I left this in in case someone else can see a way. (rdm) |
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336 */ |
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337 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp) |
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338 { |
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339 uint32_t transit= arrival_timestamp - sent_timestamp; |
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340 int d; |
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341 s->transit= transit; |
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342 d= FFABS(transit - s->transit); |
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343 s->jitter += d - ((s->jitter + 8)>>4); |
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344 } |
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345 #endif |
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346 |
0 | 347 /** |
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348 * some rtp servers assume client is dead if they don't hear from them... |
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349 * so we send a Receiver Report to the provided ByteIO context |
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350 * (we don't have access to the rtcp handle from here) |
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351 */ |
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352 int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count) |
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353 { |
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354 ByteIOContext pb; |
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355 uint8_t *buf; |
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356 int len; |
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357 int rtcp_bytes; |
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358 RTPStatistics *stats= &s->statistics; |
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359 uint32_t lost; |
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360 uint32_t extended_max; |
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361 uint32_t expected_interval; |
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362 uint32_t received_interval; |
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363 uint32_t lost_interval; |
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364 uint32_t expected; |
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365 uint32_t fraction; |
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366 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time? |
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367 |
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368 if (!s->rtp_ctx || (count < 1)) |
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369 return -1; |
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370 |
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371 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */ |
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372 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */ |
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373 s->octet_count += count; |
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374 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / |
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375 RTCP_TX_RATIO_DEN; |
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376 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !? |
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377 if (rtcp_bytes < 28) |
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378 return -1; |
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379 s->last_octet_count = s->octet_count; |
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380 |
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381 if (url_open_dyn_buf(&pb) < 0) |
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382 return -1; |
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383 |
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384 // Receiver Report |
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385 put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */ |
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386 put_byte(&pb, 201); |
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387 put_be16(&pb, 7); /* length in words - 1 */ |
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388 put_be32(&pb, s->ssrc); // our own SSRC |
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389 put_be32(&pb, s->ssrc); // XXX: should be the server's here! |
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390 // some placeholders we should really fill... |
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391 // RFC 1889/p64 |
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392 extended_max= stats->cycles + stats->max_seq; |
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393 expected= extended_max - stats->base_seq + 1; |
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394 lost= expected - stats->received; |
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395 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits... |
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396 expected_interval= expected - stats->expected_prior; |
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397 stats->expected_prior= expected; |
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398 received_interval= stats->received - stats->received_prior; |
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399 stats->received_prior= stats->received; |
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400 lost_interval= expected_interval - received_interval; |
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401 if (expected_interval==0 || lost_interval<=0) fraction= 0; |
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402 else fraction = (lost_interval<<8)/expected_interval; |
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403 |
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404 fraction= (fraction<<24) | lost; |
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405 |
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406 put_be32(&pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */ |
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407 put_be32(&pb, extended_max); /* max sequence received */ |
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408 put_be32(&pb, stats->jitter>>4); /* jitter */ |
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409 |
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410 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE) |
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411 { |
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412 put_be32(&pb, 0); /* last SR timestamp */ |
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413 put_be32(&pb, 0); /* delay since last SR */ |
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414 } else { |
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415 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special? |
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416 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time; |
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417 |
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418 put_be32(&pb, middle_32_bits); /* last SR timestamp */ |
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419 put_be32(&pb, delay_since_last); /* delay since last SR */ |
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420 } |
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421 |
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422 // CNAME |
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423 put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */ |
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424 put_byte(&pb, 202); |
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425 len = strlen(s->hostname); |
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426 put_be16(&pb, (6 + len + 3) / 4); /* length in words - 1 */ |
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427 put_be32(&pb, s->ssrc); |
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428 put_byte(&pb, 0x01); |
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429 put_byte(&pb, len); |
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430 put_buffer(&pb, s->hostname, len); |
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431 // padding |
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432 for (len = (6 + len) % 4; len % 4; len++) { |
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433 put_byte(&pb, 0); |
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434 } |
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435 |
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436 put_flush_packet(&pb); |
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437 len = url_close_dyn_buf(&pb, &buf); |
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438 if ((len > 0) && buf) { |
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439 int result; |
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440 #if defined(DEBUG) |
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441 printf("sending %d bytes of RR\n", len); |
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442 #endif |
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443 result= url_write(s->rtp_ctx, buf, len); |
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444 #if defined(DEBUG) |
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445 printf("result from url_write: %d\n", result); |
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446 #endif |
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447 av_free(buf); |
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448 } |
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449 return 0; |
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450 } |
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451 |
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452 /** |
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453 * open a new RTP parse context for stream 'st'. 'st' can be NULL for |
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454 * MPEG2TS streams to indicate that they should be demuxed inside the |
885 | 455 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned) |
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456 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system. |
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457 */ |
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458 RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_t *rtp_payload_data) |
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459 { |
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460 RTPDemuxContext *s; |
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461 |
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462 s = av_mallocz(sizeof(RTPDemuxContext)); |
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463 if (!s) |
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464 return NULL; |
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465 s->payload_type = payload_type; |
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466 s->last_rtcp_ntp_time = AV_NOPTS_VALUE; |
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467 s->first_rtcp_ntp_time = AV_NOPTS_VALUE; |
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468 s->ic = s1; |
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469 s->st = st; |
774 | 470 s->rtp_payload_data = rtp_payload_data; |
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471 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp? |
774 | 472 if (!strcmp(AVRtpPayloadTypes[payload_type].enc_name, "MP2T")) { |
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473 s->ts = mpegts_parse_open(s->ic); |
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474 if (s->ts == NULL) { |
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475 av_free(s); |
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476 return NULL; |
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477 } |
305 | 478 } else { |
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479 switch(st->codec->codec_id) { |
305 | 480 case CODEC_ID_MPEG1VIDEO: |
481 case CODEC_ID_MPEG2VIDEO: | |
482 case CODEC_ID_MP2: | |
483 case CODEC_ID_MP3: | |
484 case CODEC_ID_MPEG4: | |
1419 | 485 case CODEC_ID_H264: |
305 | 486 st->need_parsing = 1; |
487 break; | |
488 default: | |
489 break; | |
490 } | |
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491 } |
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492 // needed to send back RTCP RR in RTSP sessions |
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493 s->rtp_ctx = rtpc; |
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494 gethostname(s->hostname, sizeof(s->hostname)); |
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495 return s; |
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496 } |
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497 |
774 | 498 static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf) |
499 { | |
500 int au_headers_length, au_header_size, i; | |
501 GetBitContext getbitcontext; | |
502 rtp_payload_data_t *infos; | |
503 | |
504 infos = s->rtp_payload_data; | |
505 | |
506 if (infos == NULL) | |
507 return -1; | |
508 | |
509 /* decode the first 2 bytes where are stored the AUHeader sections | |
510 length in bits */ | |
511 au_headers_length = BE_16(buf); | |
512 | |
513 if (au_headers_length > RTP_MAX_PACKET_LENGTH) | |
514 return -1; | |
515 | |
516 infos->au_headers_length_bytes = (au_headers_length + 7) / 8; | |
517 | |
518 /* skip AU headers length section (2 bytes) */ | |
519 buf += 2; | |
520 | |
521 init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8); | |
522 | |
523 /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */ | |
524 au_header_size = infos->sizelength + infos->indexlength; | |
525 if (au_header_size <= 0 || (au_headers_length % au_header_size != 0)) | |
526 return -1; | |
527 | |
528 infos->nb_au_headers = au_headers_length / au_header_size; | |
529 infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers); | |
530 | |
531 /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving) | |
532 In my test, the faad decoder doesnt behave correctly when sending each AU one by one | |
533 but does when sending the whole as one big packet... */ | |
534 infos->au_headers[0].size = 0; | |
535 infos->au_headers[0].index = 0; | |
536 for (i = 0; i < infos->nb_au_headers; ++i) { | |
537 infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength); | |
538 infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength); | |
539 } | |
540 | |
541 infos->nb_au_headers = 1; | |
542 | |
543 return 0; | |
544 } | |
545 | |
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546 /** |
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547 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc. |
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548 */ |
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549 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp) |
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550 { |
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551 switch(s->st->codec->codec_id) { |
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552 case CODEC_ID_MP2: |
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553 case CODEC_ID_MPEG1VIDEO: |
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554 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) { |
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555 int64_t addend; |
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556 |
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557 int delta_timestamp; |
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558 /* XXX: is it really necessary to unify the timestamp base ? */ |
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559 /* compute pts from timestamp with received ntp_time */ |
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560 delta_timestamp = timestamp - s->last_rtcp_timestamp; |
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561 /* convert to 90 kHz without overflow */ |
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562 addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14; |
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563 addend = (addend * 5625) >> 14; |
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564 pkt->pts = addend + delta_timestamp; |
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565 } |
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566 break; |
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567 case CODEC_ID_AAC: |
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568 case CODEC_ID_H264: |
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569 case CODEC_ID_MPEG4: |
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570 pkt->pts = timestamp; |
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571 break; |
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572 default: |
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573 /* no timestamp info yet */ |
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574 break; |
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575 } |
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576 pkt->stream_index = s->st->index; |
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577 } |
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578 |
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579 /** |
885 | 580 * Parse an RTP or RTCP packet directly sent as a buffer. |
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581 * @param s RTP parse context. |
0 | 582 * @param pkt returned packet |
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583 * @param buf input buffer or NULL to read the next packets |
0 | 584 * @param len buffer len |
885 | 585 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow |
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586 * (use buf as NULL to read the next). -1 if no packet (error or no more packet). |
0 | 587 */ |
885 | 588 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, |
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589 const uint8_t *buf, int len) |
0 | 590 { |
591 unsigned int ssrc, h; | |
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592 int payload_type, seq, ret; |
0 | 593 AVStream *st; |
65 | 594 uint32_t timestamp; |
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595 int rv= 0; |
885 | 596 |
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597 if (!buf) { |
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598 /* return the next packets, if any */ |
1419 | 599 if(s->st && s->parse_packet) { |
1431
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gpoirier
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|
600 timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned.... |
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|
601 rv= s->parse_packet(s, pkt, ×tamp, NULL, 0); |
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602 finalize_packet(s, pkt, timestamp); |
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|
603 return rv; |
1419 | 604 } else { |
1431
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gpoirier
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diff
changeset
|
605 // TODO: Move to a dynamic packet handler (like above) |
1424
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
606 if (s->read_buf_index >= s->read_buf_size) |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
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|
607 return -1; |
1c39ce5c6a5d
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gpoirier
parents:
1419
diff
changeset
|
608 ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index, |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
609 s->read_buf_size - s->read_buf_index); |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
610 if (ret < 0) |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
611 return -1; |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
612 s->read_buf_index += ret; |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
613 if (s->read_buf_index < s->read_buf_size) |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
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|
614 return 1; |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
615 else |
1c39ce5c6a5d
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gpoirier
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1419
diff
changeset
|
616 return 0; |
1419 | 617 } |
294
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|
618 } |
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|
619 |
0 | 620 if (len < 12) |
621 return -1; | |
622 | |
623 if ((buf[0] & 0xc0) != (RTP_VERSION << 6)) | |
624 return -1; | |
625 if (buf[1] >= 200 && buf[1] <= 204) { | |
294
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|
626 rtcp_parse_packet(s, buf, len); |
0 | 627 return -1; |
628 } | |
629 payload_type = buf[1] & 0x7f; | |
630 seq = (buf[2] << 8) | buf[3]; | |
631 timestamp = decode_be32(buf + 4); | |
632 ssrc = decode_be32(buf + 8); | |
1425
00d9393a126f
make ffmpeg able to send back a RTCP receiver report.
gpoirier
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1424
diff
changeset
|
633 /* store the ssrc in the RTPDemuxContext */ |
00d9393a126f
make ffmpeg able to send back a RTCP receiver report.
gpoirier
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1424
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|
634 s->ssrc = ssrc; |
885 | 635 |
0 | 636 /* NOTE: we can handle only one payload type */ |
637 if (s->payload_type != payload_type) | |
638 return -1; | |
986
f5194ed612b5
fix debug code: do not try to take st->codec if st is NULL. This makes ffplay give me pics from FreeBoxTV \o/
mmu_man
parents:
896
diff
changeset
|
639 |
f5194ed612b5
fix debug code: do not try to take st->codec if st is NULL. This makes ffplay give me pics from FreeBoxTV \o/
mmu_man
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896
diff
changeset
|
640 st = s->st; |
1445
db97355877b1
add valid statistics for the RTCP receiver report.
gpoirier
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changeset
|
641 // only do something with this if all the rtp checks pass... |
db97355877b1
add valid statistics for the RTCP receiver report.
gpoirier
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1443
diff
changeset
|
642 if(!rtp_valid_packet_in_sequence(&s->statistics, seq)) |
db97355877b1
add valid statistics for the RTCP receiver report.
gpoirier
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|
643 { |
986
f5194ed612b5
fix debug code: do not try to take st->codec if st is NULL. This makes ffplay give me pics from FreeBoxTV \o/
mmu_man
parents:
896
diff
changeset
|
644 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n", |
0 | 645 payload_type, seq, ((s->seq + 1) & 0xffff)); |
1445
db97355877b1
add valid statistics for the RTCP receiver report.
gpoirier
parents:
1443
diff
changeset
|
646 return -1; |
0 | 647 } |
1445
db97355877b1
add valid statistics for the RTCP receiver report.
gpoirier
parents:
1443
diff
changeset
|
648 |
0 | 649 s->seq = seq; |
650 len -= 12; | |
651 buf += 12; | |
294
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652 |
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653 if (!st) { |
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654 /* specific MPEG2TS demux support */ |
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655 ret = mpegts_parse_packet(s->ts, pkt, buf, len); |
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656 if (ret < 0) |
0 | 657 return -1; |
294
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658 if (ret < len) { |
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659 s->read_buf_size = len - ret; |
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660 memcpy(s->buf, buf + ret, s->read_buf_size); |
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661 s->read_buf_index = 0; |
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662 return 1; |
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|
663 } |
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|
664 } else { |
1419 | 665 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise. |
820
feca73904e67
changing AVCodecContext codec -> *codec in AVStream so additions to AVCodecContext dont randomize AVStream and break binary compatibility
michael
parents:
777
diff
changeset
|
666 switch(st->codec->codec_id) { |
294
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667 case CODEC_ID_MP2: |
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|
668 /* better than nothing: skip mpeg audio RTP header */ |
0 | 669 if (len <= 4) |
670 return -1; | |
294
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671 h = decode_be32(buf); |
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|
672 len -= 4; |
6091b76cfc2a
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diff
changeset
|
673 buf += 4; |
6091b76cfc2a
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diff
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|
674 av_new_packet(pkt, len); |
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diff
changeset
|
675 memcpy(pkt->data, buf, len); |
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diff
changeset
|
676 break; |
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diff
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677 case CODEC_ID_MPEG1VIDEO: |
305 | 678 /* better than nothing: skip mpeg video RTP header */ |
294
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bellard
parents:
241
diff
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|
679 if (len <= 4) |
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bellard
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241
diff
changeset
|
680 return -1; |
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added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
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diff
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|
681 h = decode_be32(buf); |
0 | 682 buf += 4; |
683 len -= 4; | |
294
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parents:
241
diff
changeset
|
684 if (h & (1 << 26)) { |
6091b76cfc2a
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diff
changeset
|
685 /* mpeg2 */ |
6091b76cfc2a
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diff
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|
686 if (len <= 4) |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
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diff
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|
687 return -1; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
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diff
changeset
|
688 buf += 4; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
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diff
changeset
|
689 len -= 4; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
690 } |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
691 av_new_packet(pkt, len); |
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bellard
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diff
changeset
|
692 memcpy(pkt->data, buf, len); |
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241
diff
changeset
|
693 break; |
1427
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
694 // moved from below, verbatim. this is because this section handles packets, and the lower switch handles |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
695 // timestamps. |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
696 // TODO: Put this into a dynamic packet handler... |
1472
49d5a5ca2987
get rid of CODEC_ID_MPEG4AAC after next version bump, and change it to CODEC_ID_AAC where used
bcoudurier
parents:
1460
diff
changeset
|
697 case CODEC_ID_AAC: |
1427
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
698 if (rtp_parse_mp4_au(s, buf)) |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
699 return -1; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
700 { |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
701 rtp_payload_data_t *infos = s->rtp_payload_data; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
702 if (infos == NULL) |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
703 return -1; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
704 buf += infos->au_headers_length_bytes + 2; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
705 len -= infos->au_headers_length_bytes + 2; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
706 |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
707 /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
708 one au_header */ |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
709 av_new_packet(pkt, infos->au_headers[0].size); |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
710 memcpy(pkt->data, buf, infos->au_headers[0].size); |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
711 buf += infos->au_headers[0].size; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
712 len -= infos->au_headers[0].size; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
713 } |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
714 s->read_buf_size = len; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
715 s->buf_ptr = buf; |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
716 rv= 0; |
1427
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
717 break; |
294
6091b76cfc2a
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bellard
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241
diff
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|
718 default: |
1419 | 719 if(s->parse_packet) { |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
720 rv= s->parse_packet(s, pkt, ×tamp, buf, len); |
1419 | 721 } else { |
1424
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
722 av_new_packet(pkt, len); |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
723 memcpy(pkt->data, buf, len); |
1419 | 724 } |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
725 break; |
0 | 726 } |
885 | 727 |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
728 // now perform timestamp things.... |
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
729 finalize_packet(s, pkt, timestamp); |
0 | 730 } |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
731 return rv; |
0 | 732 } |
733 | |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
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241
diff
changeset
|
734 void rtp_parse_close(RTPDemuxContext *s) |
0 | 735 { |
1419 | 736 // TODO: fold this into the protocol specific data fields. |
774 | 737 if (!strcmp(AVRtpPayloadTypes[s->payload_type].enc_name, "MP2T")) { |
294
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added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
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241
diff
changeset
|
738 mpegts_parse_close(s->ts); |
0 | 739 } |
294
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241
diff
changeset
|
740 av_free(s); |
0 | 741 } |
742 | |
743 /* rtp output */ | |
744 | |
745 static int rtp_write_header(AVFormatContext *s1) | |
746 { | |
294
6091b76cfc2a
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241
diff
changeset
|
747 RTPDemuxContext *s = s1->priv_data; |
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diff
changeset
|
748 int payload_type, max_packet_size, n; |
0 | 749 AVStream *st; |
750 | |
751 if (s1->nb_streams != 1) | |
752 return -1; | |
753 st = s1->streams[0]; | |
754 | |
820
feca73904e67
changing AVCodecContext codec -> *codec in AVStream so additions to AVCodecContext dont randomize AVStream and break binary compatibility
michael
parents:
777
diff
changeset
|
755 payload_type = rtp_get_payload_type(st->codec); |
0 | 756 if (payload_type < 0) |
757 payload_type = RTP_PT_PRIVATE; /* private payload type */ | |
758 s->payload_type = payload_type; | |
759 | |
1099 | 760 // following 2 FIXMies could be set based on the current time, theres normaly no info leak, as rtp will likely be transmitted immedeatly |
1045
7f1b7f811f01
fix constraint violation: libavformat is not allowed to modify state of caller, including rng state
rfelker
parents:
986
diff
changeset
|
761 s->base_timestamp = 0; /* FIXME: was random(), what should this be? */ |
0 | 762 s->timestamp = s->base_timestamp; |
1045
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763 s->ssrc = 0; /* FIXME: was random(), what should this be? */ |
0 | 764 s->first_packet = 1; |
765 | |
766 max_packet_size = url_fget_max_packet_size(&s1->pb); | |
767 if (max_packet_size <= 12) | |
768 return AVERROR_IO; | |
769 s->max_payload_size = max_packet_size - 12; | |
770 | |
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changing AVCodecContext codec -> *codec in AVStream so additions to AVCodecContext dont randomize AVStream and break binary compatibility
michael
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777
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771 switch(st->codec->codec_id) { |
0 | 772 case CODEC_ID_MP2: |
232 | 773 case CODEC_ID_MP3: |
0 | 774 s->buf_ptr = s->buf + 4; |
775 s->cur_timestamp = 0; | |
776 break; | |
777 case CODEC_ID_MPEG1VIDEO: | |
778 s->cur_timestamp = 0; | |
779 break; | |
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780 case CODEC_ID_MPEG2TS: |
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781 n = s->max_payload_size / TS_PACKET_SIZE; |
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782 if (n < 1) |
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783 n = 1; |
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784 s->max_payload_size = n * TS_PACKET_SIZE; |
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785 s->buf_ptr = s->buf; |
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786 break; |
0 | 787 default: |
788 s->buf_ptr = s->buf; | |
789 break; | |
790 } | |
791 | |
792 return 0; | |
793 } | |
794 | |
795 /* send an rtcp sender report packet */ | |
65 | 796 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time) |
0 | 797 { |
294
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798 RTPDemuxContext *s = s1->priv_data; |
0 | 799 #if defined(DEBUG) |
1443
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800 printf("RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp); |
0 | 801 #endif |
802 put_byte(&s1->pb, (RTP_VERSION << 6)); | |
803 put_byte(&s1->pb, 200); | |
804 put_be16(&s1->pb, 6); /* length in words - 1 */ | |
805 put_be32(&s1->pb, s->ssrc); | |
806 put_be64(&s1->pb, ntp_time); | |
807 put_be32(&s1->pb, s->timestamp); | |
808 put_be32(&s1->pb, s->packet_count); | |
809 put_be32(&s1->pb, s->octet_count); | |
810 put_flush_packet(&s1->pb); | |
811 } | |
812 | |
813 /* send an rtp packet. sequence number is incremented, but the caller | |
814 must update the timestamp itself */ | |
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michael
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815 static void rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m) |
0 | 816 { |
294
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817 RTPDemuxContext *s = s1->priv_data; |
0 | 818 |
819 #ifdef DEBUG | |
820 printf("rtp_send_data size=%d\n", len); | |
821 #endif | |
822 | |
823 /* build the RTP header */ | |
824 put_byte(&s1->pb, (RTP_VERSION << 6)); | |
826
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michael
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825 put_byte(&s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7)); |
0 | 826 put_be16(&s1->pb, s->seq); |
827 put_be32(&s1->pb, s->timestamp); | |
828 put_be32(&s1->pb, s->ssrc); | |
885 | 829 |
0 | 830 put_buffer(&s1->pb, buf1, len); |
831 put_flush_packet(&s1->pb); | |
885 | 832 |
0 | 833 s->seq++; |
834 s->octet_count += len; | |
835 s->packet_count++; | |
836 } | |
837 | |
838 /* send an integer number of samples and compute time stamp and fill | |
839 the rtp send buffer before sending. */ | |
840 static void rtp_send_samples(AVFormatContext *s1, | |
241 | 841 const uint8_t *buf1, int size, int sample_size) |
0 | 842 { |
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843 RTPDemuxContext *s = s1->priv_data; |
0 | 844 int len, max_packet_size, n; |
845 | |
846 max_packet_size = (s->max_payload_size / sample_size) * sample_size; | |
847 /* not needed, but who nows */ | |
848 if ((size % sample_size) != 0) | |
849 av_abort(); | |
850 while (size > 0) { | |
851 len = (max_packet_size - (s->buf_ptr - s->buf)); | |
852 if (len > size) | |
853 len = size; | |
854 | |
855 /* copy data */ | |
856 memcpy(s->buf_ptr, buf1, len); | |
857 s->buf_ptr += len; | |
858 buf1 += len; | |
859 size -= len; | |
860 n = (s->buf_ptr - s->buf); | |
861 /* if buffer full, then send it */ | |
862 if (n >= max_packet_size) { | |
826
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michael
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863 rtp_send_data(s1, s->buf, n, 0); |
0 | 864 s->buf_ptr = s->buf; |
865 /* update timestamp */ | |
866 s->timestamp += n / sample_size; | |
867 } | |
868 } | |
885 | 869 } |
0 | 870 |
871 /* NOTE: we suppose that exactly one frame is given as argument here */ | |
872 /* XXX: test it */ | |
873 static void rtp_send_mpegaudio(AVFormatContext *s1, | |
241 | 874 const uint8_t *buf1, int size) |
0 | 875 { |
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876 RTPDemuxContext *s = s1->priv_data; |
0 | 877 AVStream *st = s1->streams[0]; |
878 int len, count, max_packet_size; | |
879 | |
880 max_packet_size = s->max_payload_size; | |
881 | |
882 /* test if we must flush because not enough space */ | |
883 len = (s->buf_ptr - s->buf); | |
884 if ((len + size) > max_packet_size) { | |
885 if (len > 4) { | |
826
ee7ba4017c8d
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michael
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820
diff
changeset
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886 rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); |
0 | 887 s->buf_ptr = s->buf + 4; |
888 /* 90 KHz time stamp */ | |
885 | 889 s->timestamp = s->base_timestamp + |
820
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890 (s->cur_timestamp * 90000LL) / st->codec->sample_rate; |
0 | 891 } |
892 } | |
893 | |
894 /* add the packet */ | |
895 if (size > max_packet_size) { | |
896 /* big packet: fragment */ | |
897 count = 0; | |
898 while (size > 0) { | |
899 len = max_packet_size - 4; | |
900 if (len > size) | |
901 len = size; | |
902 /* build fragmented packet */ | |
903 s->buf[0] = 0; | |
904 s->buf[1] = 0; | |
905 s->buf[2] = count >> 8; | |
906 s->buf[3] = count; | |
907 memcpy(s->buf + 4, buf1, len); | |
826
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michael
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820
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908 rtp_send_data(s1, s->buf, len + 4, 0); |
0 | 909 size -= len; |
910 buf1 += len; | |
911 count += len; | |
912 } | |
913 } else { | |
914 if (s->buf_ptr == s->buf + 4) { | |
915 /* no fragmentation possible */ | |
916 s->buf[0] = 0; | |
917 s->buf[1] = 0; | |
918 s->buf[2] = 0; | |
919 s->buf[3] = 0; | |
920 } | |
921 memcpy(s->buf_ptr, buf1, size); | |
922 s->buf_ptr += size; | |
923 } | |
820
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changing AVCodecContext codec -> *codec in AVStream so additions to AVCodecContext dont randomize AVStream and break binary compatibility
michael
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924 s->cur_timestamp += st->codec->frame_size; |
0 | 925 } |
926 | |
927 /* NOTE: a single frame must be passed with sequence header if | |
928 needed. XXX: use slices. */ | |
929 static void rtp_send_mpegvideo(AVFormatContext *s1, | |
241 | 930 const uint8_t *buf1, int size) |
0 | 931 { |
294
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932 RTPDemuxContext *s = s1->priv_data; |
0 | 933 AVStream *st = s1->streams[0]; |
934 int len, h, max_packet_size; | |
65 | 935 uint8_t *q; |
0 | 936 |
937 max_packet_size = s->max_payload_size; | |
938 | |
939 while (size > 0) { | |
940 /* XXX: more correct headers */ | |
941 h = 0; | |
820
feca73904e67
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michael
parents:
777
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|
942 if (st->codec->sub_id == 2) |
0 | 943 h |= 1 << 26; /* mpeg 2 indicator */ |
944 q = s->buf; | |
945 *q++ = h >> 24; | |
946 *q++ = h >> 16; | |
947 *q++ = h >> 8; | |
948 *q++ = h; | |
949 | |
820
feca73904e67
changing AVCodecContext codec -> *codec in AVStream so additions to AVCodecContext dont randomize AVStream and break binary compatibility
michael
parents:
777
diff
changeset
|
950 if (st->codec->sub_id == 2) { |
0 | 951 h = 0; |
952 *q++ = h >> 24; | |
953 *q++ = h >> 16; | |
954 *q++ = h >> 8; | |
955 *q++ = h; | |
956 } | |
885 | 957 |
0 | 958 len = max_packet_size - (q - s->buf); |
959 if (len > size) | |
960 len = size; | |
961 | |
962 memcpy(q, buf1, len); | |
963 q += len; | |
964 | |
965 /* 90 KHz time stamp */ | |
885 | 966 s->timestamp = s->base_timestamp + |
820
feca73904e67
changing AVCodecContext codec -> *codec in AVStream so additions to AVCodecContext dont randomize AVStream and break binary compatibility
michael
parents:
777
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changeset
|
967 av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps |
827
d064b7d0899d
mpeg1video m bit fix by (Luca Abeni: lucabe72, email it)
michael
parents:
826
diff
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968 rtp_send_data(s1, s->buf, q - s->buf, (len == size)); |
0 | 969 |
970 buf1 += len; | |
971 size -= len; | |
972 } | |
973 s->cur_timestamp++; | |
974 } | |
975 | |
976 static void rtp_send_raw(AVFormatContext *s1, | |
241 | 977 const uint8_t *buf1, int size) |
0 | 978 { |
294
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979 RTPDemuxContext *s = s1->priv_data; |
0 | 980 AVStream *st = s1->streams[0]; |
981 int len, max_packet_size; | |
982 | |
983 max_packet_size = s->max_payload_size; | |
984 | |
985 while (size > 0) { | |
986 len = max_packet_size; | |
987 if (len > size) | |
988 len = size; | |
989 | |
990 /* 90 KHz time stamp */ | |
885 | 991 s->timestamp = s->base_timestamp + |
820
feca73904e67
changing AVCodecContext codec -> *codec in AVStream so additions to AVCodecContext dont randomize AVStream and break binary compatibility
michael
parents:
777
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|
992 av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps |
826
ee7ba4017c8d
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michael
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820
diff
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993 rtp_send_data(s1, buf1, len, (len == size)); |
0 | 994 |
995 buf1 += len; | |
996 size -= len; | |
997 } | |
998 s->cur_timestamp++; | |
999 } | |
1000 | |
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1001 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */ |
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1002 static void rtp_send_mpegts_raw(AVFormatContext *s1, |
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1003 const uint8_t *buf1, int size) |
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1004 { |
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1005 RTPDemuxContext *s = s1->priv_data; |
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1006 int len, out_len; |
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1007 |
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1008 while (size >= TS_PACKET_SIZE) { |
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1009 len = s->max_payload_size - (s->buf_ptr - s->buf); |
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1010 if (len > size) |
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1011 len = size; |
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1012 memcpy(s->buf_ptr, buf1, len); |
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1013 buf1 += len; |
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1014 size -= len; |
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1015 s->buf_ptr += len; |
885 | 1016 |
294
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1017 out_len = s->buf_ptr - s->buf; |
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1018 if (out_len >= s->max_payload_size) { |
826
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michael
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820
diff
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1019 rtp_send_data(s1, s->buf, out_len, 0); |
294
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1020 s->buf_ptr = s->buf; |
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1021 } |
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1022 } |
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1023 } |
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1024 |
0 | 1025 /* write an RTP packet. 'buf1' must contain a single specific frame. */ |
468 | 1026 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) |
0 | 1027 { |
294
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1028 RTPDemuxContext *s = s1->priv_data; |
0 | 1029 AVStream *st = s1->streams[0]; |
1030 int rtcp_bytes; | |
65 | 1031 int64_t ntp_time; |
468 | 1032 int size= pkt->size; |
1033 uint8_t *buf1= pkt->data; | |
885 | 1034 |
0 | 1035 #ifdef DEBUG |
468 | 1036 printf("%d: write len=%d\n", pkt->stream_index, size); |
0 | 1037 #endif |
1038 | |
1039 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */ | |
885 | 1040 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / |
0 | 1041 RTCP_TX_RATIO_DEN; |
1042 if (s->first_packet || rtcp_bytes >= 28) { | |
1043 /* compute NTP time */ | |
173 | 1044 /* XXX: 90 kHz timestamp hardcoded */ |
468 | 1045 ntp_time = (pkt->pts << 28) / 5625; |
885 | 1046 rtcp_send_sr(s1, ntp_time); |
0 | 1047 s->last_octet_count = s->octet_count; |
1048 s->first_packet = 0; | |
1049 } | |
1050 | |
820
feca73904e67
changing AVCodecContext codec -> *codec in AVStream so additions to AVCodecContext dont randomize AVStream and break binary compatibility
michael
parents:
777
diff
changeset
|
1051 switch(st->codec->codec_id) { |
0 | 1052 case CODEC_ID_PCM_MULAW: |
1053 case CODEC_ID_PCM_ALAW: | |
1054 case CODEC_ID_PCM_U8: | |
1055 case CODEC_ID_PCM_S8: | |
820
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changing AVCodecContext codec -> *codec in AVStream so additions to AVCodecContext dont randomize AVStream and break binary compatibility
michael
parents:
777
diff
changeset
|
1056 rtp_send_samples(s1, buf1, size, 1 * st->codec->channels); |
0 | 1057 break; |
1058 case CODEC_ID_PCM_U16BE: | |
1059 case CODEC_ID_PCM_U16LE: | |
1060 case CODEC_ID_PCM_S16BE: | |
1061 case CODEC_ID_PCM_S16LE: | |
820
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changing AVCodecContext codec -> *codec in AVStream so additions to AVCodecContext dont randomize AVStream and break binary compatibility
michael
parents:
777
diff
changeset
|
1062 rtp_send_samples(s1, buf1, size, 2 * st->codec->channels); |
0 | 1063 break; |
1064 case CODEC_ID_MP2: | |
232 | 1065 case CODEC_ID_MP3: |
0 | 1066 rtp_send_mpegaudio(s1, buf1, size); |
1067 break; | |
1068 case CODEC_ID_MPEG1VIDEO: | |
1069 rtp_send_mpegvideo(s1, buf1, size); | |
1070 break; | |
294
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1071 case CODEC_ID_MPEG2TS: |
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1072 rtp_send_mpegts_raw(s1, buf1, size); |
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1073 break; |
0 | 1074 default: |
1075 /* better than nothing : send the codec raw data */ | |
1076 rtp_send_raw(s1, buf1, size); | |
1077 break; | |
1078 } | |
1079 return 0; | |
1080 } | |
1081 | |
1082 static int rtp_write_trailer(AVFormatContext *s1) | |
1083 { | |
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added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
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parents:
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1084 // RTPDemuxContext *s = s1->priv_data; |
0 | 1085 return 0; |
1086 } | |
1087 | |
1167 | 1088 AVOutputFormat rtp_muxer = { |
0 | 1089 "rtp", |
1090 "RTP output format", | |
1091 NULL, | |
1092 NULL, | |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
1093 sizeof(RTPDemuxContext), |
0 | 1094 CODEC_ID_PCM_MULAW, |
1095 CODEC_ID_NONE, | |
1096 rtp_write_header, | |
1097 rtp_write_packet, | |
1098 rtp_write_trailer, | |
1099 }; |