Mercurial > libavformat.hg
annotate rtp.c @ 1740:5ebac013d792 libavformat
stealing a few field names from mplayers asf demuxer
author | michael |
---|---|
date | Sat, 27 Jan 2007 12:07:16 +0000 |
parents | a782462e2497 |
children | 1f7a6dc01100 |
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> | |
1670 | 29 #include <arpa/inet.h> |
0 | 30 #include <netdb.h> |
31 | |
1419 | 32 #include "rtp_internal.h" |
1460 | 33 #include "rtp_h264.h" |
1419 | 34 |
0 | 35 //#define DEBUG |
36 | |
37 | |
38 /* TODO: - add RTCP statistics reporting (should be optional). | |
39 | |
40 - add support for h263/mpeg4 packetized output : IDEA: send a | |
41 buffer to 'rtp_write_packet' contains all the packets for ONE | |
42 frame. Each packet should have a four byte header containing | |
43 the length in big endian format (same trick as | |
885 | 44 'url_open_dyn_packet_buf') |
0 | 45 */ |
46 | |
774 | 47 /* from http://www.iana.org/assignments/rtp-parameters last updated 05 January 2005 */ |
48 AVRtpPayloadType_t AVRtpPayloadTypes[]= | |
49 { | |
50 {0, "PCMU", CODEC_TYPE_AUDIO, CODEC_ID_PCM_MULAW, 8000, 1}, | |
51 {1, "Reserved", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
52 {2, "Reserved", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
53 {3, "GSM", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
54 {4, "G723", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
55 {5, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
56 {6, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 16000, 1}, | |
57 {7, "LPC", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
58 {8, "PCMA", CODEC_TYPE_AUDIO, CODEC_ID_PCM_ALAW, 8000, 1}, | |
59 {9, "G722", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
60 {10, "L16", CODEC_TYPE_AUDIO, CODEC_ID_PCM_S16BE, 44100, 2}, | |
61 {11, "L16", CODEC_TYPE_AUDIO, CODEC_ID_PCM_S16BE, 44100, 1}, | |
1259 | 62 {12, "QCELP", CODEC_TYPE_AUDIO, CODEC_ID_QCELP, 8000, 1}, |
774 | 63 {13, "CN", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, |
64 {14, "MPA", CODEC_TYPE_AUDIO, CODEC_ID_MP2, 90000, -1}, | |
65 {15, "G728", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
66 {16, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 11025, 1}, | |
67 {17, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 22050, 1}, | |
68 {18, "G729", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
69 {19, "reserved", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, | |
70 {20, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, | |
71 {21, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, | |
72 {22, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, | |
73 {23, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, | |
74 {24, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1}, | |
75 {25, "CelB", CODEC_TYPE_VIDEO, CODEC_ID_NONE, 90000, -1}, | |
76 {26, "JPEG", CODEC_TYPE_VIDEO, CODEC_ID_MJPEG, 90000, -1}, | |
77 {27, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1}, | |
78 {28, "nv", CODEC_TYPE_VIDEO, CODEC_ID_NONE, 90000, -1}, | |
79 {29, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1}, | |
80 {30, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1}, | |
81 {31, "H261", CODEC_TYPE_VIDEO, CODEC_ID_H261, 90000, -1}, | |
82 {32, "MPV", CODEC_TYPE_VIDEO, CODEC_ID_MPEG1VIDEO, 90000, -1}, | |
83 {33, "MP2T", CODEC_TYPE_DATA, CODEC_ID_MPEG2TS, 90000, -1}, | |
84 {34, "H263", CODEC_TYPE_VIDEO, CODEC_ID_H263, 90000, -1}, | |
85 {35, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
86 {36, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
87 {37, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
88 {38, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
89 {39, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
90 {40, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
91 {41, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
92 {42, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
93 {43, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
94 {44, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
95 {45, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
96 {46, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
97 {47, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
98 {48, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
99 {49, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
100 {50, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
101 {51, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
102 {52, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
103 {53, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
104 {54, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
105 {55, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
106 {56, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
107 {57, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
108 {58, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
109 {59, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
110 {60, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
111 {61, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
112 {62, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
113 {63, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
114 {64, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
115 {65, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
116 {66, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
117 {67, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
118 {68, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
119 {69, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
120 {70, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
121 {71, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
122 {72, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
123 {73, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
124 {74, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
125 {75, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
126 {76, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
127 {77, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
128 {78, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
129 {79, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
130 {80, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
131 {81, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
132 {82, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
133 {83, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
134 {84, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
135 {85, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
136 {86, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
137 {87, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
138 {88, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
139 {89, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
140 {90, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
141 {91, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
142 {92, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
143 {93, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
144 {94, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
145 {95, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
146 {96, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
147 {97, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
148 {98, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
149 {99, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
150 {100, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
151 {101, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
152 {102, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
153 {103, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
154 {104, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
155 {105, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
156 {106, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
157 {107, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
158 {108, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
159 {109, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
160 {110, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
161 {111, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
162 {112, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
163 {113, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
164 {114, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
165 {115, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
166 {116, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
167 {117, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
168 {118, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
169 {119, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
170 {120, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
171 {121, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
172 {122, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
173 {123, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
174 {124, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
175 {125, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
176 {126, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
177 {127, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
178 {-1, "", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1} | |
179 }; | |
0 | 180 |
1419 | 181 /* statistics functions */ |
182 RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL; | |
183 | |
184 static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4}; | |
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185 static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC}; |
0 | 186 |
1419 | 187 static void register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler) |
188 { | |
189 handler->next= RTPFirstDynamicPayloadHandler; | |
190 RTPFirstDynamicPayloadHandler= handler; | |
191 } | |
885 | 192 |
1419 | 193 void av_register_rtp_dynamic_payload_handlers() |
194 { | |
195 register_dynamic_payload_handler(&mp4v_es_handler); | |
196 register_dynamic_payload_handler(&mpeg4_generic_handler); | |
197 register_dynamic_payload_handler(&ff_h264_dynamic_handler); | |
198 } | |
0 | 199 |
200 int rtp_get_codec_info(AVCodecContext *codec, int payload_type) | |
201 { | |
774 | 202 if (AVRtpPayloadTypes[payload_type].codec_id != CODEC_ID_NONE) { |
203 codec->codec_type = AVRtpPayloadTypes[payload_type].codec_type; | |
1116 | 204 codec->codec_id = AVRtpPayloadTypes[payload_type].codec_id; |
774 | 205 if (AVRtpPayloadTypes[payload_type].audio_channels > 0) |
206 codec->channels = AVRtpPayloadTypes[payload_type].audio_channels; | |
207 if (AVRtpPayloadTypes[payload_type].clock_rate > 0) | |
208 codec->sample_rate = AVRtpPayloadTypes[payload_type].clock_rate; | |
209 return 0; | |
0 | 210 } |
774 | 211 return -1; |
0 | 212 } |
213 | |
214 /* return < 0 if unknown payload type */ | |
215 int rtp_get_payload_type(AVCodecContext *codec) | |
216 { | |
774 | 217 int i, payload_type; |
0 | 218 |
219 /* compute the payload type */ | |
774 | 220 for (payload_type = -1, i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i) |
221 if (AVRtpPayloadTypes[i].codec_id == codec->codec_id) { | |
222 if (codec->codec_id == CODEC_ID_PCM_S16BE) | |
223 if (codec->channels != AVRtpPayloadTypes[i].audio_channels) | |
224 continue; | |
225 payload_type = AVRtpPayloadTypes[i].pt; | |
0 | 226 } |
227 return payload_type; | |
228 } | |
229 | |
65 | 230 static inline uint32_t decode_be32(const uint8_t *p) |
0 | 231 { |
232 return (p[0] << 24) | (p[1] << 16) | (p[2] << 8) | p[3]; | |
233 } | |
234 | |
173 | 235 static inline uint64_t decode_be64(const uint8_t *p) |
0 | 236 { |
65 | 237 return ((uint64_t)decode_be32(p) << 32) | decode_be32(p + 4); |
0 | 238 } |
239 | |
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240 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len) |
0 | 241 { |
242 if (buf[1] != 200) | |
243 return -1; | |
244 s->last_rtcp_ntp_time = decode_be64(buf + 8); | |
173 | 245 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) |
246 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time; | |
0 | 247 s->last_rtcp_timestamp = decode_be32(buf + 16); |
248 return 0; | |
249 } | |
250 | |
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251 #define RTP_SEQ_MOD (1<<16) |
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252 |
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253 /** |
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254 * called on parse open packet |
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255 */ |
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256 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet. |
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257 { |
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258 memset(s, 0, sizeof(RTPStatistics)); |
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259 s->max_seq= base_sequence; |
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260 s->probation= 1; |
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261 } |
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262 |
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263 /** |
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264 * called whenever there is a large jump in sequence numbers, or when they get out of probation... |
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265 */ |
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266 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq) |
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267 { |
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268 s->max_seq= seq; |
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269 s->cycles= 0; |
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270 s->base_seq= seq -1; |
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271 s->bad_seq= RTP_SEQ_MOD + 1; |
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272 s->received= 0; |
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273 s->expected_prior= 0; |
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274 s->received_prior= 0; |
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275 s->jitter= 0; |
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276 s->transit= 0; |
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277 } |
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278 |
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279 /** |
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280 * returns 1 if we should handle this packet. |
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281 */ |
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282 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq) |
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283 { |
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284 uint16_t udelta= seq - s->max_seq; |
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285 const int MAX_DROPOUT= 3000; |
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286 const int MAX_MISORDER = 100; |
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287 const int MIN_SEQUENTIAL = 2; |
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288 |
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289 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */ |
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290 if(s->probation) |
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291 { |
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292 if(seq==s->max_seq + 1) { |
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293 s->probation--; |
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294 s->max_seq= seq; |
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295 if(s->probation==0) { |
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296 rtp_init_sequence(s, seq); |
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297 s->received++; |
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298 return 1; |
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299 } |
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300 } else { |
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301 s->probation= MIN_SEQUENTIAL - 1; |
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302 s->max_seq = seq; |
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303 } |
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304 } else if (udelta < MAX_DROPOUT) { |
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305 // in order, with permissible gap |
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306 if(seq < s->max_seq) { |
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307 //sequence number wrapped; count antother 64k cycles |
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308 s->cycles += RTP_SEQ_MOD; |
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309 } |
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310 s->max_seq= seq; |
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311 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) { |
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312 // sequence made a large jump... |
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313 if(seq==s->bad_seq) { |
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314 // two sequential packets-- assume that the other side restarted without telling us; just resync. |
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315 rtp_init_sequence(s, seq); |
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316 } else { |
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317 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1); |
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318 return 0; |
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319 } |
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320 } else { |
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321 // duplicate or reordered packet... |
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322 } |
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323 s->received++; |
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324 return 1; |
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325 } |
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326 |
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327 #if 0 |
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328 /** |
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329 * This function is currently unused; without a valid local ntp time, I don't see how we could calculate the |
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330 * difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values |
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331 * never change. I left this in in case someone else can see a way. (rdm) |
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332 */ |
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333 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp) |
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334 { |
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335 uint32_t transit= arrival_timestamp - sent_timestamp; |
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336 int d; |
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337 s->transit= transit; |
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338 d= FFABS(transit - s->transit); |
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339 s->jitter += d - ((s->jitter + 8)>>4); |
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340 } |
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341 #endif |
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342 |
0 | 343 /** |
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344 * some rtp servers assume client is dead if they don't hear from them... |
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345 * so we send a Receiver Report to the provided ByteIO context |
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346 * (we don't have access to the rtcp handle from here) |
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347 */ |
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348 int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count) |
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349 { |
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350 ByteIOContext pb; |
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351 uint8_t *buf; |
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352 int len; |
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353 int rtcp_bytes; |
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354 RTPStatistics *stats= &s->statistics; |
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355 uint32_t lost; |
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356 uint32_t extended_max; |
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357 uint32_t expected_interval; |
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358 uint32_t received_interval; |
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359 uint32_t lost_interval; |
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360 uint32_t expected; |
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361 uint32_t fraction; |
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362 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time? |
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363 |
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364 if (!s->rtp_ctx || (count < 1)) |
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365 return -1; |
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366 |
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367 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */ |
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368 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */ |
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369 s->octet_count += count; |
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370 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / |
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371 RTCP_TX_RATIO_DEN; |
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372 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !? |
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373 if (rtcp_bytes < 28) |
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374 return -1; |
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375 s->last_octet_count = s->octet_count; |
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376 |
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377 if (url_open_dyn_buf(&pb) < 0) |
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378 return -1; |
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379 |
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380 // Receiver Report |
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381 put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */ |
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382 put_byte(&pb, 201); |
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383 put_be16(&pb, 7); /* length in words - 1 */ |
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384 put_be32(&pb, s->ssrc); // our own SSRC |
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385 put_be32(&pb, s->ssrc); // XXX: should be the server's here! |
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386 // some placeholders we should really fill... |
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387 // RFC 1889/p64 |
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388 extended_max= stats->cycles + stats->max_seq; |
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389 expected= extended_max - stats->base_seq + 1; |
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390 lost= expected - stats->received; |
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391 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits... |
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392 expected_interval= expected - stats->expected_prior; |
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393 stats->expected_prior= expected; |
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394 received_interval= stats->received - stats->received_prior; |
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395 stats->received_prior= stats->received; |
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396 lost_interval= expected_interval - received_interval; |
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397 if (expected_interval==0 || lost_interval<=0) fraction= 0; |
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398 else fraction = (lost_interval<<8)/expected_interval; |
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399 |
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400 fraction= (fraction<<24) | lost; |
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401 |
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402 put_be32(&pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */ |
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403 put_be32(&pb, extended_max); /* max sequence received */ |
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404 put_be32(&pb, stats->jitter>>4); /* jitter */ |
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405 |
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406 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE) |
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407 { |
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408 put_be32(&pb, 0); /* last SR timestamp */ |
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409 put_be32(&pb, 0); /* delay since last SR */ |
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410 } else { |
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411 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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412 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time; |
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413 |
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414 put_be32(&pb, middle_32_bits); /* last SR timestamp */ |
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415 put_be32(&pb, delay_since_last); /* delay since last SR */ |
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416 } |
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417 |
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418 // CNAME |
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419 put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */ |
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420 put_byte(&pb, 202); |
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421 len = strlen(s->hostname); |
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422 put_be16(&pb, (6 + len + 3) / 4); /* length in words - 1 */ |
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423 put_be32(&pb, s->ssrc); |
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424 put_byte(&pb, 0x01); |
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425 put_byte(&pb, len); |
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426 put_buffer(&pb, s->hostname, len); |
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427 // padding |
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428 for (len = (6 + len) % 4; len % 4; len++) { |
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429 put_byte(&pb, 0); |
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430 } |
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431 |
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432 put_flush_packet(&pb); |
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433 len = url_close_dyn_buf(&pb, &buf); |
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434 if ((len > 0) && buf) { |
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435 int result; |
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436 #if defined(DEBUG) |
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437 printf("sending %d bytes of RR\n", len); |
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438 #endif |
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439 result= url_write(s->rtp_ctx, buf, len); |
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440 #if defined(DEBUG) |
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441 printf("result from url_write: %d\n", result); |
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442 #endif |
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443 av_free(buf); |
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444 } |
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445 return 0; |
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446 } |
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447 |
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448 /** |
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449 * open a new RTP parse context for stream 'st'. 'st' can be NULL for |
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450 * MPEG2TS streams to indicate that they should be demuxed inside the |
885 | 451 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned) |
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452 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system. |
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453 */ |
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454 RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_t *rtp_payload_data) |
294
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455 { |
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456 RTPDemuxContext *s; |
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457 |
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458 s = av_mallocz(sizeof(RTPDemuxContext)); |
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459 if (!s) |
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460 return NULL; |
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461 s->payload_type = payload_type; |
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462 s->last_rtcp_ntp_time = AV_NOPTS_VALUE; |
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463 s->first_rtcp_ntp_time = AV_NOPTS_VALUE; |
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464 s->ic = s1; |
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465 s->st = st; |
774 | 466 s->rtp_payload_data = rtp_payload_data; |
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467 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp? |
774 | 468 if (!strcmp(AVRtpPayloadTypes[payload_type].enc_name, "MP2T")) { |
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469 s->ts = mpegts_parse_open(s->ic); |
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470 if (s->ts == NULL) { |
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471 av_free(s); |
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472 return NULL; |
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473 } |
305 | 474 } else { |
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475 switch(st->codec->codec_id) { |
305 | 476 case CODEC_ID_MPEG1VIDEO: |
477 case CODEC_ID_MPEG2VIDEO: | |
478 case CODEC_ID_MP2: | |
479 case CODEC_ID_MP3: | |
480 case CODEC_ID_MPEG4: | |
1419 | 481 case CODEC_ID_H264: |
305 | 482 st->need_parsing = 1; |
483 break; | |
484 default: | |
485 break; | |
486 } | |
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487 } |
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488 // needed to send back RTCP RR in RTSP sessions |
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489 s->rtp_ctx = rtpc; |
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490 gethostname(s->hostname, sizeof(s->hostname)); |
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491 return s; |
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492 } |
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493 |
774 | 494 static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf) |
495 { | |
496 int au_headers_length, au_header_size, i; | |
497 GetBitContext getbitcontext; | |
498 rtp_payload_data_t *infos; | |
499 | |
500 infos = s->rtp_payload_data; | |
501 | |
502 if (infos == NULL) | |
503 return -1; | |
504 | |
505 /* decode the first 2 bytes where are stored the AUHeader sections | |
506 length in bits */ | |
1673 | 507 au_headers_length = AV_RB16(buf); |
774 | 508 |
509 if (au_headers_length > RTP_MAX_PACKET_LENGTH) | |
510 return -1; | |
511 | |
512 infos->au_headers_length_bytes = (au_headers_length + 7) / 8; | |
513 | |
514 /* skip AU headers length section (2 bytes) */ | |
515 buf += 2; | |
516 | |
517 init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8); | |
518 | |
519 /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */ | |
520 au_header_size = infos->sizelength + infos->indexlength; | |
521 if (au_header_size <= 0 || (au_headers_length % au_header_size != 0)) | |
522 return -1; | |
523 | |
524 infos->nb_au_headers = au_headers_length / au_header_size; | |
525 infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers); | |
526 | |
527 /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving) | |
528 In my test, the faad decoder doesnt behave correctly when sending each AU one by one | |
529 but does when sending the whole as one big packet... */ | |
530 infos->au_headers[0].size = 0; | |
531 infos->au_headers[0].index = 0; | |
532 for (i = 0; i < infos->nb_au_headers; ++i) { | |
533 infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength); | |
534 infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength); | |
535 } | |
536 | |
537 infos->nb_au_headers = 1; | |
538 | |
539 return 0; | |
540 } | |
541 | |
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542 /** |
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543 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc. |
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544 */ |
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545 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp) |
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546 { |
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547 switch(s->st->codec->codec_id) { |
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548 case CODEC_ID_MP2: |
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549 case CODEC_ID_MPEG1VIDEO: |
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550 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) { |
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551 int64_t addend; |
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552 |
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553 int delta_timestamp; |
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554 /* XXX: is it really necessary to unify the timestamp base ? */ |
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555 /* compute pts from timestamp with received ntp_time */ |
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556 delta_timestamp = timestamp - s->last_rtcp_timestamp; |
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557 /* convert to 90 kHz without overflow */ |
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558 addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14; |
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559 addend = (addend * 5625) >> 14; |
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560 pkt->pts = addend + delta_timestamp; |
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561 } |
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562 break; |
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563 case CODEC_ID_AAC: |
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564 case CODEC_ID_H264: |
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565 case CODEC_ID_MPEG4: |
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566 pkt->pts = timestamp; |
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567 break; |
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568 default: |
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569 /* no timestamp info yet */ |
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570 break; |
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571 } |
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572 pkt->stream_index = s->st->index; |
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573 } |
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574 |
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575 /** |
885 | 576 * Parse an RTP or RTCP packet directly sent as a buffer. |
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577 * @param s RTP parse context. |
0 | 578 * @param pkt returned packet |
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579 * @param buf input buffer or NULL to read the next packets |
0 | 580 * @param len buffer len |
885 | 581 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow |
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582 * (use buf as NULL to read the next). -1 if no packet (error or no more packet). |
0 | 583 */ |
885 | 584 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, |
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585 const uint8_t *buf, int len) |
0 | 586 { |
587 unsigned int ssrc, h; | |
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588 int payload_type, seq, ret; |
0 | 589 AVStream *st; |
65 | 590 uint32_t timestamp; |
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591 int rv= 0; |
885 | 592 |
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593 if (!buf) { |
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594 /* return the next packets, if any */ |
1419 | 595 if(s->st && s->parse_packet) { |
1431
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|
596 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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597 rv= s->parse_packet(s, pkt, ×tamp, NULL, 0); |
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598 finalize_packet(s, pkt, timestamp); |
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599 return rv; |
1419 | 600 } else { |
1431
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|
601 // TODO: Move to a dynamic packet handler (like above) |
1424
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
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1419
diff
changeset
|
602 if (s->read_buf_index >= s->read_buf_size) |
1c39ce5c6a5d
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gpoirier
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|
603 return -1; |
1c39ce5c6a5d
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gpoirier
parents:
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|
604 ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index, |
1c39ce5c6a5d
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gpoirier
parents:
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changeset
|
605 s->read_buf_size - s->read_buf_index); |
1c39ce5c6a5d
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gpoirier
parents:
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|
606 if (ret < 0) |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
607 return -1; |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
608 s->read_buf_index += ret; |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
609 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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|
610 return 1; |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
611 else |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
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1419
diff
changeset
|
612 return 0; |
1419 | 613 } |
294
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614 } |
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|
615 |
0 | 616 if (len < 12) |
617 return -1; | |
618 | |
619 if ((buf[0] & 0xc0) != (RTP_VERSION << 6)) | |
620 return -1; | |
621 if (buf[1] >= 200 && buf[1] <= 204) { | |
294
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622 rtcp_parse_packet(s, buf, len); |
0 | 623 return -1; |
624 } | |
625 payload_type = buf[1] & 0x7f; | |
626 seq = (buf[2] << 8) | buf[3]; | |
627 timestamp = decode_be32(buf + 4); | |
628 ssrc = decode_be32(buf + 8); | |
1425
00d9393a126f
make ffmpeg able to send back a RTCP receiver report.
gpoirier
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changeset
|
629 /* store the ssrc in the RTPDemuxContext */ |
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gpoirier
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|
630 s->ssrc = ssrc; |
885 | 631 |
0 | 632 /* NOTE: we can handle only one payload type */ |
633 if (s->payload_type != payload_type) | |
634 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
|
635 |
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
|
636 st = s->st; |
1445
db97355877b1
add valid statistics for the RTCP receiver report.
gpoirier
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changeset
|
637 // 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
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changeset
|
638 if(!rtp_valid_packet_in_sequence(&s->statistics, seq)) |
db97355877b1
add valid statistics for the RTCP receiver report.
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changeset
|
639 { |
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
|
640 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n", |
0 | 641 payload_type, seq, ((s->seq + 1) & 0xffff)); |
1445
db97355877b1
add valid statistics for the RTCP receiver report.
gpoirier
parents:
1443
diff
changeset
|
642 return -1; |
0 | 643 } |
1445
db97355877b1
add valid statistics for the RTCP receiver report.
gpoirier
parents:
1443
diff
changeset
|
644 |
0 | 645 s->seq = seq; |
646 len -= 12; | |
647 buf += 12; | |
294
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648 |
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649 if (!st) { |
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|
650 /* specific MPEG2TS demux support */ |
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651 ret = mpegts_parse_packet(s->ts, pkt, buf, len); |
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652 if (ret < 0) |
0 | 653 return -1; |
294
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654 if (ret < len) { |
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655 s->read_buf_size = len - ret; |
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656 memcpy(s->buf, buf + ret, s->read_buf_size); |
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|
657 s->read_buf_index = 0; |
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|
658 return 1; |
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659 } |
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|
660 } else { |
1419 | 661 // 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
|
662 switch(st->codec->codec_id) { |
294
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663 case CODEC_ID_MP2: |
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changeset
|
664 /* better than nothing: skip mpeg audio RTP header */ |
0 | 665 if (len <= 4) |
666 return -1; | |
294
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667 h = decode_be32(buf); |
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changeset
|
668 len -= 4; |
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diff
changeset
|
669 buf += 4; |
6091b76cfc2a
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diff
changeset
|
670 av_new_packet(pkt, len); |
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241
diff
changeset
|
671 memcpy(pkt->data, buf, len); |
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diff
changeset
|
672 break; |
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diff
changeset
|
673 case CODEC_ID_MPEG1VIDEO: |
305 | 674 /* better than nothing: skip mpeg video RTP header */ |
294
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bellard
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241
diff
changeset
|
675 if (len <= 4) |
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bellard
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241
diff
changeset
|
676 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
changeset
|
677 h = decode_be32(buf); |
0 | 678 buf += 4; |
679 len -= 4; | |
294
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parents:
241
diff
changeset
|
680 if (h & (1 << 26)) { |
6091b76cfc2a
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diff
changeset
|
681 /* mpeg2 */ |
6091b76cfc2a
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diff
changeset
|
682 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
changeset
|
683 return -1; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
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diff
changeset
|
684 buf += 4; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
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parents:
241
diff
changeset
|
685 len -= 4; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
686 } |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
687 av_new_packet(pkt, len); |
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diff
changeset
|
688 memcpy(pkt->data, buf, len); |
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diff
changeset
|
689 break; |
1427
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
690 // 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
|
691 // timestamps. |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
692 // 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
|
693 case CODEC_ID_AAC: |
1427
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
694 if (rtp_parse_mp4_au(s, buf)) |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
695 return -1; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
696 { |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
697 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
|
698 if (infos == NULL) |
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 buf += infos->au_headers_length_bytes + 2; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
701 len -= infos->au_headers_length_bytes + 2; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
702 |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
703 /* 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
|
704 one au_header */ |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
705 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
|
706 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
|
707 buf += infos->au_headers[0].size; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
708 len -= infos->au_headers[0].size; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
709 } |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
710 s->read_buf_size = len; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
711 s->buf_ptr = buf; |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
712 rv= 0; |
1427
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
713 break; |
294
6091b76cfc2a
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bellard
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241
diff
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|
714 default: |
1419 | 715 if(s->parse_packet) { |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
716 rv= s->parse_packet(s, pkt, ×tamp, buf, len); |
1419 | 717 } else { |
1424
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
718 av_new_packet(pkt, len); |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
719 memcpy(pkt->data, buf, len); |
1419 | 720 } |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
721 break; |
0 | 722 } |
885 | 723 |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
724 // now perform timestamp things.... |
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
725 finalize_packet(s, pkt, timestamp); |
0 | 726 } |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
727 return rv; |
0 | 728 } |
729 | |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
730 void rtp_parse_close(RTPDemuxContext *s) |
0 | 731 { |
1419 | 732 // TODO: fold this into the protocol specific data fields. |
774 | 733 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
|
734 mpegts_parse_close(s->ts); |
0 | 735 } |
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
|
736 av_free(s); |
0 | 737 } |
738 | |
739 /* rtp output */ | |
740 | |
741 static int rtp_write_header(AVFormatContext *s1) | |
742 { | |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
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241
diff
changeset
|
743 RTPDemuxContext *s = s1->priv_data; |
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diff
changeset
|
744 int payload_type, max_packet_size, n; |
0 | 745 AVStream *st; |
746 | |
747 if (s1->nb_streams != 1) | |
748 return -1; | |
749 st = s1->streams[0]; | |
750 | |
820
feca73904e67
changing AVCodecContext codec -> *codec in AVStream so additions to AVCodecContext dont randomize AVStream and break binary compatibility
michael
parents:
777
diff
changeset
|
751 payload_type = rtp_get_payload_type(st->codec); |
0 | 752 if (payload_type < 0) |
753 payload_type = RTP_PT_PRIVATE; /* private payload type */ | |
754 s->payload_type = payload_type; | |
755 | |
1099 | 756 // 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
|
757 s->base_timestamp = 0; /* FIXME: was random(), what should this be? */ |
0 | 758 s->timestamp = s->base_timestamp; |
1045
7f1b7f811f01
fix constraint violation: libavformat is not allowed to modify state of caller, including rng state
rfelker
parents:
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759 s->ssrc = 0; /* FIXME: was random(), what should this be? */ |
0 | 760 s->first_packet = 1; |
761 | |
762 max_packet_size = url_fget_max_packet_size(&s1->pb); | |
763 if (max_packet_size <= 12) | |
764 return AVERROR_IO; | |
765 s->max_payload_size = max_packet_size - 12; | |
766 | |
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changing AVCodecContext codec -> *codec in AVStream so additions to AVCodecContext dont randomize AVStream and break binary compatibility
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767 switch(st->codec->codec_id) { |
0 | 768 case CODEC_ID_MP2: |
232 | 769 case CODEC_ID_MP3: |
0 | 770 s->buf_ptr = s->buf + 4; |
771 s->cur_timestamp = 0; | |
772 break; | |
773 case CODEC_ID_MPEG1VIDEO: | |
774 s->cur_timestamp = 0; | |
775 break; | |
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776 case CODEC_ID_MPEG2TS: |
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777 n = s->max_payload_size / TS_PACKET_SIZE; |
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778 if (n < 1) |
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779 n = 1; |
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780 s->max_payload_size = n * TS_PACKET_SIZE; |
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781 s->buf_ptr = s->buf; |
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782 break; |
0 | 783 default: |
784 s->buf_ptr = s->buf; | |
785 break; | |
786 } | |
787 | |
788 return 0; | |
789 } | |
790 | |
791 /* send an rtcp sender report packet */ | |
65 | 792 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time) |
0 | 793 { |
294
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794 RTPDemuxContext *s = s1->priv_data; |
0 | 795 #if defined(DEBUG) |
1443
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796 printf("RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp); |
0 | 797 #endif |
798 put_byte(&s1->pb, (RTP_VERSION << 6)); | |
799 put_byte(&s1->pb, 200); | |
800 put_be16(&s1->pb, 6); /* length in words - 1 */ | |
801 put_be32(&s1->pb, s->ssrc); | |
802 put_be64(&s1->pb, ntp_time); | |
803 put_be32(&s1->pb, s->timestamp); | |
804 put_be32(&s1->pb, s->packet_count); | |
805 put_be32(&s1->pb, s->octet_count); | |
806 put_flush_packet(&s1->pb); | |
807 } | |
808 | |
809 /* send an rtp packet. sequence number is incremented, but the caller | |
810 must update the timestamp itself */ | |
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811 static void rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m) |
0 | 812 { |
294
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813 RTPDemuxContext *s = s1->priv_data; |
0 | 814 |
815 #ifdef DEBUG | |
816 printf("rtp_send_data size=%d\n", len); | |
817 #endif | |
818 | |
819 /* build the RTP header */ | |
820 put_byte(&s1->pb, (RTP_VERSION << 6)); | |
826
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821 put_byte(&s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7)); |
0 | 822 put_be16(&s1->pb, s->seq); |
823 put_be32(&s1->pb, s->timestamp); | |
824 put_be32(&s1->pb, s->ssrc); | |
885 | 825 |
0 | 826 put_buffer(&s1->pb, buf1, len); |
827 put_flush_packet(&s1->pb); | |
885 | 828 |
0 | 829 s->seq++; |
830 s->octet_count += len; | |
831 s->packet_count++; | |
832 } | |
833 | |
834 /* send an integer number of samples and compute time stamp and fill | |
835 the rtp send buffer before sending. */ | |
836 static void rtp_send_samples(AVFormatContext *s1, | |
241 | 837 const uint8_t *buf1, int size, int sample_size) |
0 | 838 { |
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839 RTPDemuxContext *s = s1->priv_data; |
0 | 840 int len, max_packet_size, n; |
841 | |
842 max_packet_size = (s->max_payload_size / sample_size) * sample_size; | |
843 /* not needed, but who nows */ | |
844 if ((size % sample_size) != 0) | |
845 av_abort(); | |
846 while (size > 0) { | |
847 len = (max_packet_size - (s->buf_ptr - s->buf)); | |
848 if (len > size) | |
849 len = size; | |
850 | |
851 /* copy data */ | |
852 memcpy(s->buf_ptr, buf1, len); | |
853 s->buf_ptr += len; | |
854 buf1 += len; | |
855 size -= len; | |
856 n = (s->buf_ptr - s->buf); | |
857 /* if buffer full, then send it */ | |
858 if (n >= max_packet_size) { | |
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859 rtp_send_data(s1, s->buf, n, 0); |
0 | 860 s->buf_ptr = s->buf; |
861 /* update timestamp */ | |
862 s->timestamp += n / sample_size; | |
863 } | |
864 } | |
885 | 865 } |
0 | 866 |
867 /* NOTE: we suppose that exactly one frame is given as argument here */ | |
868 /* XXX: test it */ | |
869 static void rtp_send_mpegaudio(AVFormatContext *s1, | |
241 | 870 const uint8_t *buf1, int size) |
0 | 871 { |
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872 RTPDemuxContext *s = s1->priv_data; |
0 | 873 AVStream *st = s1->streams[0]; |
874 int len, count, max_packet_size; | |
875 | |
876 max_packet_size = s->max_payload_size; | |
877 | |
878 /* test if we must flush because not enough space */ | |
879 len = (s->buf_ptr - s->buf); | |
880 if ((len + size) > max_packet_size) { | |
881 if (len > 4) { | |
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882 rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); |
0 | 883 s->buf_ptr = s->buf + 4; |
884 /* 90 KHz time stamp */ | |
885 | 885 s->timestamp = s->base_timestamp + |
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886 (s->cur_timestamp * 90000LL) / st->codec->sample_rate; |
0 | 887 } |
888 } | |
889 | |
890 /* add the packet */ | |
891 if (size > max_packet_size) { | |
892 /* big packet: fragment */ | |
893 count = 0; | |
894 while (size > 0) { | |
895 len = max_packet_size - 4; | |
896 if (len > size) | |
897 len = size; | |
898 /* build fragmented packet */ | |
899 s->buf[0] = 0; | |
900 s->buf[1] = 0; | |
901 s->buf[2] = count >> 8; | |
902 s->buf[3] = count; | |
903 memcpy(s->buf + 4, buf1, len); | |
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904 rtp_send_data(s1, s->buf, len + 4, 0); |
0 | 905 size -= len; |
906 buf1 += len; | |
907 count += len; | |
908 } | |
909 } else { | |
910 if (s->buf_ptr == s->buf + 4) { | |
911 /* no fragmentation possible */ | |
912 s->buf[0] = 0; | |
913 s->buf[1] = 0; | |
914 s->buf[2] = 0; | |
915 s->buf[3] = 0; | |
916 } | |
917 memcpy(s->buf_ptr, buf1, size); | |
918 s->buf_ptr += size; | |
919 } | |
820
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920 s->cur_timestamp += st->codec->frame_size; |
0 | 921 } |
922 | |
923 /* NOTE: a single frame must be passed with sequence header if | |
924 needed. XXX: use slices. */ | |
925 static void rtp_send_mpegvideo(AVFormatContext *s1, | |
241 | 926 const uint8_t *buf1, int size) |
0 | 927 { |
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928 RTPDemuxContext *s = s1->priv_data; |
0 | 929 AVStream *st = s1->streams[0]; |
930 int len, h, max_packet_size; | |
65 | 931 uint8_t *q; |
0 | 932 |
933 max_packet_size = s->max_payload_size; | |
934 | |
935 while (size > 0) { | |
936 /* XXX: more correct headers */ | |
937 h = 0; | |
820
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michael
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938 if (st->codec->sub_id == 2) |
0 | 939 h |= 1 << 26; /* mpeg 2 indicator */ |
940 q = s->buf; | |
941 *q++ = h >> 24; | |
942 *q++ = h >> 16; | |
943 *q++ = h >> 8; | |
944 *q++ = h; | |
945 | |
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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777
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946 if (st->codec->sub_id == 2) { |
0 | 947 h = 0; |
948 *q++ = h >> 24; | |
949 *q++ = h >> 16; | |
950 *q++ = h >> 8; | |
951 *q++ = h; | |
952 } | |
885 | 953 |
0 | 954 len = max_packet_size - (q - s->buf); |
955 if (len > size) | |
956 len = size; | |
957 | |
958 memcpy(q, buf1, len); | |
959 q += len; | |
960 | |
961 /* 90 KHz time stamp */ | |
885 | 962 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
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963 av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps |
827
d064b7d0899d
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michael
parents:
826
diff
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964 rtp_send_data(s1, s->buf, q - s->buf, (len == size)); |
0 | 965 |
966 buf1 += len; | |
967 size -= len; | |
968 } | |
969 s->cur_timestamp++; | |
970 } | |
971 | |
972 static void rtp_send_raw(AVFormatContext *s1, | |
241 | 973 const uint8_t *buf1, int size) |
0 | 974 { |
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975 RTPDemuxContext *s = s1->priv_data; |
0 | 976 AVStream *st = s1->streams[0]; |
977 int len, max_packet_size; | |
978 | |
979 max_packet_size = s->max_payload_size; | |
980 | |
981 while (size > 0) { | |
982 len = max_packet_size; | |
983 if (len > size) | |
984 len = size; | |
985 | |
986 /* 90 KHz time stamp */ | |
885 | 987 s->timestamp = s->base_timestamp + |
820
feca73904e67
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michael
parents:
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988 av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps |
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989 rtp_send_data(s1, buf1, len, (len == size)); |
0 | 990 |
991 buf1 += len; | |
992 size -= len; | |
993 } | |
994 s->cur_timestamp++; | |
995 } | |
996 | |
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997 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */ |
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998 static void rtp_send_mpegts_raw(AVFormatContext *s1, |
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999 const uint8_t *buf1, int size) |
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1000 { |
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1001 RTPDemuxContext *s = s1->priv_data; |
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1002 int len, out_len; |
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1003 |
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1004 while (size >= TS_PACKET_SIZE) { |
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1005 len = s->max_payload_size - (s->buf_ptr - s->buf); |
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1006 if (len > size) |
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1007 len = size; |
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1008 memcpy(s->buf_ptr, buf1, len); |
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1009 buf1 += len; |
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1010 size -= len; |
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1011 s->buf_ptr += len; |
885 | 1012 |
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1013 out_len = s->buf_ptr - s->buf; |
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1014 if (out_len >= s->max_payload_size) { |
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1015 rtp_send_data(s1, s->buf, out_len, 0); |
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1016 s->buf_ptr = s->buf; |
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1017 } |
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1018 } |
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1019 } |
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1020 |
0 | 1021 /* write an RTP packet. 'buf1' must contain a single specific frame. */ |
468 | 1022 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) |
0 | 1023 { |
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1024 RTPDemuxContext *s = s1->priv_data; |
0 | 1025 AVStream *st = s1->streams[0]; |
1026 int rtcp_bytes; | |
65 | 1027 int64_t ntp_time; |
468 | 1028 int size= pkt->size; |
1029 uint8_t *buf1= pkt->data; | |
885 | 1030 |
0 | 1031 #ifdef DEBUG |
468 | 1032 printf("%d: write len=%d\n", pkt->stream_index, size); |
0 | 1033 #endif |
1034 | |
1035 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */ | |
885 | 1036 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / |
0 | 1037 RTCP_TX_RATIO_DEN; |
1038 if (s->first_packet || rtcp_bytes >= 28) { | |
1039 /* compute NTP time */ | |
173 | 1040 /* XXX: 90 kHz timestamp hardcoded */ |
468 | 1041 ntp_time = (pkt->pts << 28) / 5625; |
885 | 1042 rtcp_send_sr(s1, ntp_time); |
0 | 1043 s->last_octet_count = s->octet_count; |
1044 s->first_packet = 0; | |
1045 } | |
1046 | |
820
feca73904e67
changing AVCodecContext codec -> *codec in AVStream so additions to AVCodecContext dont randomize AVStream and break binary compatibility
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1047 switch(st->codec->codec_id) { |
0 | 1048 case CODEC_ID_PCM_MULAW: |
1049 case CODEC_ID_PCM_ALAW: | |
1050 case CODEC_ID_PCM_U8: | |
1051 case CODEC_ID_PCM_S8: | |
820
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1052 rtp_send_samples(s1, buf1, size, 1 * st->codec->channels); |
0 | 1053 break; |
1054 case CODEC_ID_PCM_U16BE: | |
1055 case CODEC_ID_PCM_U16LE: | |
1056 case CODEC_ID_PCM_S16BE: | |
1057 case CODEC_ID_PCM_S16LE: | |
820
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1058 rtp_send_samples(s1, buf1, size, 2 * st->codec->channels); |
0 | 1059 break; |
1060 case CODEC_ID_MP2: | |
232 | 1061 case CODEC_ID_MP3: |
0 | 1062 rtp_send_mpegaudio(s1, buf1, size); |
1063 break; | |
1064 case CODEC_ID_MPEG1VIDEO: | |
1065 rtp_send_mpegvideo(s1, buf1, size); | |
1066 break; | |
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1067 case CODEC_ID_MPEG2TS: |
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1068 rtp_send_mpegts_raw(s1, buf1, size); |
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1069 break; |
0 | 1070 default: |
1071 /* better than nothing : send the codec raw data */ | |
1072 rtp_send_raw(s1, buf1, size); | |
1073 break; | |
1074 } | |
1075 return 0; | |
1076 } | |
1077 | |
1078 static int rtp_write_trailer(AVFormatContext *s1) | |
1079 { | |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
1080 // RTPDemuxContext *s = s1->priv_data; |
0 | 1081 return 0; |
1082 } | |
1083 | |
1167 | 1084 AVOutputFormat rtp_muxer = { |
0 | 1085 "rtp", |
1086 "RTP output format", | |
1087 NULL, | |
1088 NULL, | |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
1089 sizeof(RTPDemuxContext), |
0 | 1090 CODEC_ID_PCM_MULAW, |
1091 CODEC_ID_NONE, | |
1092 rtp_write_header, | |
1093 rtp_write_packet, | |
1094 rtp_write_trailer, | |
1095 }; |