Mercurial > libavformat.hg
annotate rtp.c @ 1854:0432f6b969b0 libavformat
simplify swf muxer
author | bcoudurier |
---|---|
date | Sun, 04 Mar 2007 22:30:25 +0000 |
parents | 1f7a6dc01100 |
children | 99d50be70d9a |
rev | line source |
---|---|
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> | |
1754 | 26 #include "network.h" |
0 | 27 |
1419 | 28 #include "rtp_internal.h" |
1460 | 29 #include "rtp_h264.h" |
1419 | 30 |
0 | 31 //#define DEBUG |
32 | |
33 | |
34 /* TODO: - add RTCP statistics reporting (should be optional). | |
35 | |
36 - add support for h263/mpeg4 packetized output : IDEA: send a | |
37 buffer to 'rtp_write_packet' contains all the packets for ONE | |
38 frame. Each packet should have a four byte header containing | |
39 the length in big endian format (same trick as | |
885 | 40 'url_open_dyn_packet_buf') |
0 | 41 */ |
42 | |
774 | 43 /* from http://www.iana.org/assignments/rtp-parameters last updated 05 January 2005 */ |
44 AVRtpPayloadType_t AVRtpPayloadTypes[]= | |
45 { | |
46 {0, "PCMU", CODEC_TYPE_AUDIO, CODEC_ID_PCM_MULAW, 8000, 1}, | |
47 {1, "Reserved", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
48 {2, "Reserved", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
49 {3, "GSM", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
50 {4, "G723", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
51 {5, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
52 {6, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 16000, 1}, | |
53 {7, "LPC", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
54 {8, "PCMA", CODEC_TYPE_AUDIO, CODEC_ID_PCM_ALAW, 8000, 1}, | |
55 {9, "G722", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
56 {10, "L16", CODEC_TYPE_AUDIO, CODEC_ID_PCM_S16BE, 44100, 2}, | |
57 {11, "L16", CODEC_TYPE_AUDIO, CODEC_ID_PCM_S16BE, 44100, 1}, | |
1259 | 58 {12, "QCELP", CODEC_TYPE_AUDIO, CODEC_ID_QCELP, 8000, 1}, |
774 | 59 {13, "CN", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, |
60 {14, "MPA", CODEC_TYPE_AUDIO, CODEC_ID_MP2, 90000, -1}, | |
61 {15, "G728", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
62 {16, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 11025, 1}, | |
63 {17, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 22050, 1}, | |
64 {18, "G729", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
65 {19, "reserved", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, | |
66 {20, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, | |
67 {21, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, | |
68 {22, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, | |
69 {23, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, | |
70 {24, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1}, | |
71 {25, "CelB", CODEC_TYPE_VIDEO, CODEC_ID_NONE, 90000, -1}, | |
72 {26, "JPEG", CODEC_TYPE_VIDEO, CODEC_ID_MJPEG, 90000, -1}, | |
73 {27, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1}, | |
74 {28, "nv", CODEC_TYPE_VIDEO, CODEC_ID_NONE, 90000, -1}, | |
75 {29, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1}, | |
76 {30, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1}, | |
77 {31, "H261", CODEC_TYPE_VIDEO, CODEC_ID_H261, 90000, -1}, | |
78 {32, "MPV", CODEC_TYPE_VIDEO, CODEC_ID_MPEG1VIDEO, 90000, -1}, | |
79 {33, "MP2T", CODEC_TYPE_DATA, CODEC_ID_MPEG2TS, 90000, -1}, | |
80 {34, "H263", CODEC_TYPE_VIDEO, CODEC_ID_H263, 90000, -1}, | |
81 {35, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
82 {36, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
83 {37, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
84 {38, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
85 {39, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
86 {40, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
87 {41, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
88 {42, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
89 {43, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
90 {44, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
91 {45, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
92 {46, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
93 {47, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
94 {48, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
95 {49, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
96 {50, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
97 {51, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
98 {52, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
99 {53, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
100 {54, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
101 {55, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
102 {56, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
103 {57, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
104 {58, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
105 {59, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
106 {60, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
107 {61, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
108 {62, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
109 {63, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
110 {64, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
111 {65, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
112 {66, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
113 {67, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
114 {68, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
115 {69, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
116 {70, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
117 {71, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
118 {72, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
119 {73, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
120 {74, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
121 {75, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
122 {76, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
123 {77, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
124 {78, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
125 {79, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
126 {80, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
127 {81, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
128 {82, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
129 {83, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
130 {84, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
131 {85, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
132 {86, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
133 {87, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
134 {88, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
135 {89, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
136 {90, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
137 {91, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
138 {92, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
139 {93, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
140 {94, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
141 {95, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
142 {96, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
143 {97, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
144 {98, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
145 {99, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
146 {100, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
147 {101, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
148 {102, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
149 {103, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
150 {104, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
151 {105, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
152 {106, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
153 {107, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
154 {108, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
155 {109, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
156 {110, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
157 {111, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
158 {112, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
159 {113, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
160 {114, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
161 {115, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
162 {116, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
163 {117, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
164 {118, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
165 {119, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
166 {120, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
167 {121, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
168 {122, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
169 {123, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
170 {124, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
171 {125, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
172 {126, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
173 {127, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
174 {-1, "", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1} | |
175 }; | |
0 | 176 |
1419 | 177 /* statistics functions */ |
178 RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL; | |
179 | |
180 static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4}; | |
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181 static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC}; |
0 | 182 |
1419 | 183 static void register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler) |
184 { | |
185 handler->next= RTPFirstDynamicPayloadHandler; | |
186 RTPFirstDynamicPayloadHandler= handler; | |
187 } | |
885 | 188 |
1419 | 189 void av_register_rtp_dynamic_payload_handlers() |
190 { | |
191 register_dynamic_payload_handler(&mp4v_es_handler); | |
192 register_dynamic_payload_handler(&mpeg4_generic_handler); | |
193 register_dynamic_payload_handler(&ff_h264_dynamic_handler); | |
194 } | |
0 | 195 |
196 int rtp_get_codec_info(AVCodecContext *codec, int payload_type) | |
197 { | |
774 | 198 if (AVRtpPayloadTypes[payload_type].codec_id != CODEC_ID_NONE) { |
199 codec->codec_type = AVRtpPayloadTypes[payload_type].codec_type; | |
1116 | 200 codec->codec_id = AVRtpPayloadTypes[payload_type].codec_id; |
774 | 201 if (AVRtpPayloadTypes[payload_type].audio_channels > 0) |
202 codec->channels = AVRtpPayloadTypes[payload_type].audio_channels; | |
203 if (AVRtpPayloadTypes[payload_type].clock_rate > 0) | |
204 codec->sample_rate = AVRtpPayloadTypes[payload_type].clock_rate; | |
205 return 0; | |
0 | 206 } |
774 | 207 return -1; |
0 | 208 } |
209 | |
210 /* return < 0 if unknown payload type */ | |
211 int rtp_get_payload_type(AVCodecContext *codec) | |
212 { | |
774 | 213 int i, payload_type; |
0 | 214 |
215 /* compute the payload type */ | |
774 | 216 for (payload_type = -1, i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i) |
217 if (AVRtpPayloadTypes[i].codec_id == codec->codec_id) { | |
218 if (codec->codec_id == CODEC_ID_PCM_S16BE) | |
219 if (codec->channels != AVRtpPayloadTypes[i].audio_channels) | |
220 continue; | |
221 payload_type = AVRtpPayloadTypes[i].pt; | |
0 | 222 } |
223 return payload_type; | |
224 } | |
225 | |
65 | 226 static inline uint32_t decode_be32(const uint8_t *p) |
0 | 227 { |
228 return (p[0] << 24) | (p[1] << 16) | (p[2] << 8) | p[3]; | |
229 } | |
230 | |
173 | 231 static inline uint64_t decode_be64(const uint8_t *p) |
0 | 232 { |
65 | 233 return ((uint64_t)decode_be32(p) << 32) | decode_be32(p + 4); |
0 | 234 } |
235 | |
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236 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len) |
0 | 237 { |
238 if (buf[1] != 200) | |
239 return -1; | |
240 s->last_rtcp_ntp_time = decode_be64(buf + 8); | |
173 | 241 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) |
242 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time; | |
0 | 243 s->last_rtcp_timestamp = decode_be32(buf + 16); |
244 return 0; | |
245 } | |
246 | |
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247 #define RTP_SEQ_MOD (1<<16) |
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248 |
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249 /** |
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250 * called on parse open packet |
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251 */ |
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252 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet. |
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253 { |
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254 memset(s, 0, sizeof(RTPStatistics)); |
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255 s->max_seq= base_sequence; |
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256 s->probation= 1; |
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257 } |
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258 |
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259 /** |
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260 * called whenever there is a large jump in sequence numbers, or when they get out of probation... |
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261 */ |
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262 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq) |
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263 { |
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264 s->max_seq= seq; |
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265 s->cycles= 0; |
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266 s->base_seq= seq -1; |
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267 s->bad_seq= RTP_SEQ_MOD + 1; |
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268 s->received= 0; |
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269 s->expected_prior= 0; |
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270 s->received_prior= 0; |
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271 s->jitter= 0; |
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272 s->transit= 0; |
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273 } |
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274 |
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275 /** |
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276 * returns 1 if we should handle this packet. |
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277 */ |
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278 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq) |
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279 { |
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280 uint16_t udelta= seq - s->max_seq; |
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281 const int MAX_DROPOUT= 3000; |
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282 const int MAX_MISORDER = 100; |
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283 const int MIN_SEQUENTIAL = 2; |
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284 |
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285 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */ |
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286 if(s->probation) |
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287 { |
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288 if(seq==s->max_seq + 1) { |
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289 s->probation--; |
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290 s->max_seq= seq; |
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291 if(s->probation==0) { |
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292 rtp_init_sequence(s, seq); |
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293 s->received++; |
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294 return 1; |
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295 } |
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296 } else { |
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297 s->probation= MIN_SEQUENTIAL - 1; |
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298 s->max_seq = seq; |
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299 } |
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300 } else if (udelta < MAX_DROPOUT) { |
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301 // in order, with permissible gap |
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302 if(seq < s->max_seq) { |
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303 //sequence number wrapped; count antother 64k cycles |
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304 s->cycles += RTP_SEQ_MOD; |
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305 } |
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306 s->max_seq= seq; |
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307 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) { |
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308 // sequence made a large jump... |
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309 if(seq==s->bad_seq) { |
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310 // two sequential packets-- assume that the other side restarted without telling us; just resync. |
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311 rtp_init_sequence(s, seq); |
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312 } else { |
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313 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1); |
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314 return 0; |
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315 } |
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316 } else { |
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317 // duplicate or reordered packet... |
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318 } |
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319 s->received++; |
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320 return 1; |
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321 } |
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322 |
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323 #if 0 |
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324 /** |
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325 * This function is currently unused; without a valid local ntp time, I don't see how we could calculate the |
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326 * difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values |
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327 * never change. I left this in in case someone else can see a way. (rdm) |
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328 */ |
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329 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp) |
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330 { |
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331 uint32_t transit= arrival_timestamp - sent_timestamp; |
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332 int d; |
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333 s->transit= transit; |
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334 d= FFABS(transit - s->transit); |
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335 s->jitter += d - ((s->jitter + 8)>>4); |
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336 } |
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337 #endif |
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338 |
0 | 339 /** |
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340 * some rtp servers assume client is dead if they don't hear from them... |
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341 * so we send a Receiver Report to the provided ByteIO context |
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342 * (we don't have access to the rtcp handle from here) |
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343 */ |
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344 int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count) |
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345 { |
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346 ByteIOContext pb; |
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347 uint8_t *buf; |
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348 int len; |
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349 int rtcp_bytes; |
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350 RTPStatistics *stats= &s->statistics; |
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351 uint32_t lost; |
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352 uint32_t extended_max; |
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353 uint32_t expected_interval; |
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354 uint32_t received_interval; |
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355 uint32_t lost_interval; |
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356 uint32_t expected; |
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357 uint32_t fraction; |
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358 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time? |
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359 |
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360 if (!s->rtp_ctx || (count < 1)) |
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361 return -1; |
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362 |
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363 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */ |
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364 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */ |
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365 s->octet_count += count; |
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366 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / |
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367 RTCP_TX_RATIO_DEN; |
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368 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !? |
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369 if (rtcp_bytes < 28) |
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370 return -1; |
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371 s->last_octet_count = s->octet_count; |
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372 |
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373 if (url_open_dyn_buf(&pb) < 0) |
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374 return -1; |
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375 |
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376 // Receiver Report |
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377 put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */ |
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378 put_byte(&pb, 201); |
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379 put_be16(&pb, 7); /* length in words - 1 */ |
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380 put_be32(&pb, s->ssrc); // our own SSRC |
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381 put_be32(&pb, s->ssrc); // XXX: should be the server's here! |
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382 // some placeholders we should really fill... |
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383 // RFC 1889/p64 |
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384 extended_max= stats->cycles + stats->max_seq; |
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385 expected= extended_max - stats->base_seq + 1; |
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386 lost= expected - stats->received; |
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387 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits... |
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388 expected_interval= expected - stats->expected_prior; |
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389 stats->expected_prior= expected; |
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390 received_interval= stats->received - stats->received_prior; |
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391 stats->received_prior= stats->received; |
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392 lost_interval= expected_interval - received_interval; |
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393 if (expected_interval==0 || lost_interval<=0) fraction= 0; |
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394 else fraction = (lost_interval<<8)/expected_interval; |
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395 |
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396 fraction= (fraction<<24) | lost; |
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397 |
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398 put_be32(&pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */ |
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399 put_be32(&pb, extended_max); /* max sequence received */ |
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400 put_be32(&pb, stats->jitter>>4); /* jitter */ |
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401 |
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402 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE) |
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403 { |
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404 put_be32(&pb, 0); /* last SR timestamp */ |
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405 put_be32(&pb, 0); /* delay since last SR */ |
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406 } else { |
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407 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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408 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time; |
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409 |
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410 put_be32(&pb, middle_32_bits); /* last SR timestamp */ |
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411 put_be32(&pb, delay_since_last); /* delay since last SR */ |
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412 } |
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413 |
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414 // CNAME |
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415 put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */ |
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416 put_byte(&pb, 202); |
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417 len = strlen(s->hostname); |
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418 put_be16(&pb, (6 + len + 3) / 4); /* length in words - 1 */ |
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419 put_be32(&pb, s->ssrc); |
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420 put_byte(&pb, 0x01); |
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421 put_byte(&pb, len); |
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422 put_buffer(&pb, s->hostname, len); |
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423 // padding |
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424 for (len = (6 + len) % 4; len % 4; len++) { |
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425 put_byte(&pb, 0); |
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426 } |
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427 |
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428 put_flush_packet(&pb); |
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429 len = url_close_dyn_buf(&pb, &buf); |
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430 if ((len > 0) && buf) { |
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431 int result; |
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432 #if defined(DEBUG) |
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433 printf("sending %d bytes of RR\n", len); |
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434 #endif |
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435 result= url_write(s->rtp_ctx, buf, len); |
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436 #if defined(DEBUG) |
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437 printf("result from url_write: %d\n", result); |
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438 #endif |
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439 av_free(buf); |
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440 } |
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441 return 0; |
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442 } |
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443 |
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444 /** |
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445 * open a new RTP parse context for stream 'st'. 'st' can be NULL for |
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446 * MPEG2TS streams to indicate that they should be demuxed inside the |
885 | 447 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned) |
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448 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system. |
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449 */ |
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450 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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451 { |
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452 RTPDemuxContext *s; |
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453 |
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454 s = av_mallocz(sizeof(RTPDemuxContext)); |
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455 if (!s) |
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456 return NULL; |
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457 s->payload_type = payload_type; |
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458 s->last_rtcp_ntp_time = AV_NOPTS_VALUE; |
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459 s->first_rtcp_ntp_time = AV_NOPTS_VALUE; |
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460 s->ic = s1; |
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461 s->st = st; |
774 | 462 s->rtp_payload_data = rtp_payload_data; |
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463 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp? |
774 | 464 if (!strcmp(AVRtpPayloadTypes[payload_type].enc_name, "MP2T")) { |
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465 s->ts = mpegts_parse_open(s->ic); |
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466 if (s->ts == NULL) { |
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467 av_free(s); |
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468 return NULL; |
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469 } |
305 | 470 } else { |
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471 switch(st->codec->codec_id) { |
305 | 472 case CODEC_ID_MPEG1VIDEO: |
473 case CODEC_ID_MPEG2VIDEO: | |
474 case CODEC_ID_MP2: | |
475 case CODEC_ID_MP3: | |
476 case CODEC_ID_MPEG4: | |
1419 | 477 case CODEC_ID_H264: |
305 | 478 st->need_parsing = 1; |
479 break; | |
480 default: | |
481 break; | |
482 } | |
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483 } |
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484 // needed to send back RTCP RR in RTSP sessions |
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485 s->rtp_ctx = rtpc; |
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486 gethostname(s->hostname, sizeof(s->hostname)); |
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487 return s; |
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488 } |
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489 |
774 | 490 static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf) |
491 { | |
492 int au_headers_length, au_header_size, i; | |
493 GetBitContext getbitcontext; | |
494 rtp_payload_data_t *infos; | |
495 | |
496 infos = s->rtp_payload_data; | |
497 | |
498 if (infos == NULL) | |
499 return -1; | |
500 | |
501 /* decode the first 2 bytes where are stored the AUHeader sections | |
502 length in bits */ | |
1673 | 503 au_headers_length = AV_RB16(buf); |
774 | 504 |
505 if (au_headers_length > RTP_MAX_PACKET_LENGTH) | |
506 return -1; | |
507 | |
508 infos->au_headers_length_bytes = (au_headers_length + 7) / 8; | |
509 | |
510 /* skip AU headers length section (2 bytes) */ | |
511 buf += 2; | |
512 | |
513 init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8); | |
514 | |
515 /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */ | |
516 au_header_size = infos->sizelength + infos->indexlength; | |
517 if (au_header_size <= 0 || (au_headers_length % au_header_size != 0)) | |
518 return -1; | |
519 | |
520 infos->nb_au_headers = au_headers_length / au_header_size; | |
521 infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers); | |
522 | |
523 /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving) | |
524 In my test, the faad decoder doesnt behave correctly when sending each AU one by one | |
525 but does when sending the whole as one big packet... */ | |
526 infos->au_headers[0].size = 0; | |
527 infos->au_headers[0].index = 0; | |
528 for (i = 0; i < infos->nb_au_headers; ++i) { | |
529 infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength); | |
530 infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength); | |
531 } | |
532 | |
533 infos->nb_au_headers = 1; | |
534 | |
535 return 0; | |
536 } | |
537 | |
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538 /** |
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539 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc. |
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540 */ |
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541 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp) |
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542 { |
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543 switch(s->st->codec->codec_id) { |
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544 case CODEC_ID_MP2: |
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545 case CODEC_ID_MPEG1VIDEO: |
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546 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) { |
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547 int64_t addend; |
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548 |
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549 int delta_timestamp; |
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550 /* XXX: is it really necessary to unify the timestamp base ? */ |
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551 /* compute pts from timestamp with received ntp_time */ |
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552 delta_timestamp = timestamp - s->last_rtcp_timestamp; |
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553 /* convert to 90 kHz without overflow */ |
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554 addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14; |
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555 addend = (addend * 5625) >> 14; |
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556 pkt->pts = addend + delta_timestamp; |
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557 } |
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558 break; |
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559 case CODEC_ID_AAC: |
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560 case CODEC_ID_H264: |
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561 case CODEC_ID_MPEG4: |
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562 pkt->pts = timestamp; |
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563 break; |
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564 default: |
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565 /* no timestamp info yet */ |
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566 break; |
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567 } |
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568 pkt->stream_index = s->st->index; |
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569 } |
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570 |
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571 /** |
885 | 572 * Parse an RTP or RTCP packet directly sent as a buffer. |
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573 * @param s RTP parse context. |
0 | 574 * @param pkt returned packet |
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575 * @param buf input buffer or NULL to read the next packets |
0 | 576 * @param len buffer len |
885 | 577 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow |
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578 * (use buf as NULL to read the next). -1 if no packet (error or no more packet). |
0 | 579 */ |
885 | 580 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, |
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581 const uint8_t *buf, int len) |
0 | 582 { |
583 unsigned int ssrc, h; | |
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584 int payload_type, seq, ret; |
0 | 585 AVStream *st; |
65 | 586 uint32_t timestamp; |
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587 int rv= 0; |
885 | 588 |
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589 if (!buf) { |
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590 /* return the next packets, if any */ |
1419 | 591 if(s->st && s->parse_packet) { |
1431
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|
592 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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|
593 rv= s->parse_packet(s, pkt, ×tamp, NULL, 0); |
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594 finalize_packet(s, pkt, timestamp); |
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595 return rv; |
1419 | 596 } else { |
1431
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fix more dynamic protocol stuff, needed by the forthcoming h264
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|
597 // TODO: Move to a dynamic packet handler (like above) |
1424
1c39ce5c6a5d
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gpoirier
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1419
diff
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|
598 if (s->read_buf_index >= s->read_buf_size) |
1c39ce5c6a5d
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gpoirier
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599 return -1; |
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|
600 ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index, |
1c39ce5c6a5d
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|
601 s->read_buf_size - s->read_buf_index); |
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gpoirier
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|
602 if (ret < 0) |
1c39ce5c6a5d
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gpoirier
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|
603 return -1; |
1c39ce5c6a5d
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gpoirier
parents:
1419
diff
changeset
|
604 s->read_buf_index += ret; |
1c39ce5c6a5d
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gpoirier
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1419
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|
605 if (s->read_buf_index < s->read_buf_size) |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
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|
606 return 1; |
1c39ce5c6a5d
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gpoirier
parents:
1419
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|
607 else |
1c39ce5c6a5d
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gpoirier
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|
608 return 0; |
1419 | 609 } |
294
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610 } |
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611 |
0 | 612 if (len < 12) |
613 return -1; | |
614 | |
615 if ((buf[0] & 0xc0) != (RTP_VERSION << 6)) | |
616 return -1; | |
617 if (buf[1] >= 200 && buf[1] <= 204) { | |
294
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618 rtcp_parse_packet(s, buf, len); |
0 | 619 return -1; |
620 } | |
621 payload_type = buf[1] & 0x7f; | |
622 seq = (buf[2] << 8) | buf[3]; | |
623 timestamp = decode_be32(buf + 4); | |
624 ssrc = decode_be32(buf + 8); | |
1425
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make ffmpeg able to send back a RTCP receiver report.
gpoirier
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changeset
|
625 /* store the ssrc in the RTPDemuxContext */ |
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|
626 s->ssrc = ssrc; |
885 | 627 |
0 | 628 /* NOTE: we can handle only one payload type */ |
629 if (s->payload_type != payload_type) | |
630 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
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changeset
|
631 |
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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diff
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|
632 st = s->st; |
1445
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add valid statistics for the RTCP receiver report.
gpoirier
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|
633 // only do something with this if all the rtp checks pass... |
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add valid statistics for the RTCP receiver report.
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|
634 if(!rtp_valid_packet_in_sequence(&s->statistics, seq)) |
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|
635 { |
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
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896
diff
changeset
|
636 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n", |
0 | 637 payload_type, seq, ((s->seq + 1) & 0xffff)); |
1445
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add valid statistics for the RTCP receiver report.
gpoirier
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|
638 return -1; |
0 | 639 } |
1445
db97355877b1
add valid statistics for the RTCP receiver report.
gpoirier
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1443
diff
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|
640 |
0 | 641 s->seq = seq; |
642 len -= 12; | |
643 buf += 12; | |
294
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644 |
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645 if (!st) { |
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646 /* specific MPEG2TS demux support */ |
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647 ret = mpegts_parse_packet(s->ts, pkt, buf, len); |
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648 if (ret < 0) |
0 | 649 return -1; |
294
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650 if (ret < len) { |
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651 s->read_buf_size = len - ret; |
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652 memcpy(s->buf, buf + ret, s->read_buf_size); |
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653 s->read_buf_index = 0; |
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654 return 1; |
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655 } |
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656 } else { |
1419 | 657 // 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
|
658 switch(st->codec->codec_id) { |
294
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659 case CODEC_ID_MP2: |
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660 /* better than nothing: skip mpeg audio RTP header */ |
0 | 661 if (len <= 4) |
662 return -1; | |
294
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663 h = decode_be32(buf); |
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664 len -= 4; |
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diff
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|
665 buf += 4; |
6091b76cfc2a
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diff
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|
666 av_new_packet(pkt, len); |
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667 memcpy(pkt->data, buf, len); |
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changeset
|
668 break; |
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669 case CODEC_ID_MPEG1VIDEO: |
305 | 670 /* better than nothing: skip mpeg video RTP header */ |
294
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bellard
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|
671 if (len <= 4) |
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bellard
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changeset
|
672 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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|
673 h = decode_be32(buf); |
0 | 674 buf += 4; |
675 len -= 4; | |
294
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241
diff
changeset
|
676 if (h & (1 << 26)) { |
6091b76cfc2a
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|
677 /* mpeg2 */ |
6091b76cfc2a
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diff
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|
678 if (len <= 4) |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
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|
679 return -1; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
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changeset
|
680 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
|
681 len -= 4; |
6091b76cfc2a
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bellard
parents:
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diff
changeset
|
682 } |
6091b76cfc2a
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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diff
changeset
|
683 av_new_packet(pkt, len); |
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|
684 memcpy(pkt->data, buf, len); |
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diff
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|
685 break; |
1427
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
686 // 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
|
687 // timestamps. |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
688 // 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
|
689 case CODEC_ID_AAC: |
1427
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
690 if (rtp_parse_mp4_au(s, buf)) |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
691 return -1; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
692 { |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
693 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
|
694 if (infos == NULL) |
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 buf += infos->au_headers_length_bytes + 2; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
697 len -= infos->au_headers_length_bytes + 2; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
698 |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
699 /* 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
|
700 one au_header */ |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
701 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
|
702 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
|
703 buf += infos->au_headers[0].size; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
704 len -= infos->au_headers[0].size; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
705 } |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
706 s->read_buf_size = len; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
707 s->buf_ptr = buf; |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
708 rv= 0; |
1427
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
709 break; |
294
6091b76cfc2a
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bellard
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diff
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|
710 default: |
1419 | 711 if(s->parse_packet) { |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
712 rv= s->parse_packet(s, pkt, ×tamp, buf, len); |
1419 | 713 } else { |
1424
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
714 av_new_packet(pkt, len); |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
715 memcpy(pkt->data, buf, len); |
1419 | 716 } |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
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diff
changeset
|
717 break; |
0 | 718 } |
885 | 719 |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
720 // now perform timestamp things.... |
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
721 finalize_packet(s, pkt, timestamp); |
0 | 722 } |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
723 return rv; |
0 | 724 } |
725 | |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
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241
diff
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|
726 void rtp_parse_close(RTPDemuxContext *s) |
0 | 727 { |
1419 | 728 // TODO: fold this into the protocol specific data fields. |
774 | 729 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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diff
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|
730 mpegts_parse_close(s->ts); |
0 | 731 } |
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
|
732 av_free(s); |
0 | 733 } |
734 | |
735 /* rtp output */ | |
736 | |
737 static int rtp_write_header(AVFormatContext *s1) | |
738 { | |
294
6091b76cfc2a
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diff
changeset
|
739 RTPDemuxContext *s = s1->priv_data; |
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changeset
|
740 int payload_type, max_packet_size, n; |
0 | 741 AVStream *st; |
742 | |
743 if (s1->nb_streams != 1) | |
744 return -1; | |
745 st = s1->streams[0]; | |
746 | |
820
feca73904e67
changing AVCodecContext codec -> *codec in AVStream so additions to AVCodecContext dont randomize AVStream and break binary compatibility
michael
parents:
777
diff
changeset
|
747 payload_type = rtp_get_payload_type(st->codec); |
0 | 748 if (payload_type < 0) |
749 payload_type = RTP_PT_PRIVATE; /* private payload type */ | |
750 s->payload_type = payload_type; | |
751 | |
1099 | 752 // 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
|
753 s->base_timestamp = 0; /* FIXME: was random(), what should this be? */ |
0 | 754 s->timestamp = s->base_timestamp; |
1045
7f1b7f811f01
fix constraint violation: libavformat is not allowed to modify state of caller, including rng state
rfelker
parents:
986
diff
changeset
|
755 s->ssrc = 0; /* FIXME: was random(), what should this be? */ |
0 | 756 s->first_packet = 1; |
757 | |
758 max_packet_size = url_fget_max_packet_size(&s1->pb); | |
759 if (max_packet_size <= 12) | |
760 return AVERROR_IO; | |
761 s->max_payload_size = max_packet_size - 12; | |
762 | |
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763 switch(st->codec->codec_id) { |
0 | 764 case CODEC_ID_MP2: |
232 | 765 case CODEC_ID_MP3: |
0 | 766 s->buf_ptr = s->buf + 4; |
767 s->cur_timestamp = 0; | |
768 break; | |
769 case CODEC_ID_MPEG1VIDEO: | |
770 s->cur_timestamp = 0; | |
771 break; | |
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772 case CODEC_ID_MPEG2TS: |
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773 n = s->max_payload_size / TS_PACKET_SIZE; |
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774 if (n < 1) |
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775 n = 1; |
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776 s->max_payload_size = n * TS_PACKET_SIZE; |
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777 s->buf_ptr = s->buf; |
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778 break; |
0 | 779 default: |
780 s->buf_ptr = s->buf; | |
781 break; | |
782 } | |
783 | |
784 return 0; | |
785 } | |
786 | |
787 /* send an rtcp sender report packet */ | |
65 | 788 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time) |
0 | 789 { |
294
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790 RTPDemuxContext *s = s1->priv_data; |
0 | 791 #if defined(DEBUG) |
1443
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792 printf("RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp); |
0 | 793 #endif |
794 put_byte(&s1->pb, (RTP_VERSION << 6)); | |
795 put_byte(&s1->pb, 200); | |
796 put_be16(&s1->pb, 6); /* length in words - 1 */ | |
797 put_be32(&s1->pb, s->ssrc); | |
798 put_be64(&s1->pb, ntp_time); | |
799 put_be32(&s1->pb, s->timestamp); | |
800 put_be32(&s1->pb, s->packet_count); | |
801 put_be32(&s1->pb, s->octet_count); | |
802 put_flush_packet(&s1->pb); | |
803 } | |
804 | |
805 /* send an rtp packet. sequence number is incremented, but the caller | |
806 must update the timestamp itself */ | |
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807 static void rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m) |
0 | 808 { |
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809 RTPDemuxContext *s = s1->priv_data; |
0 | 810 |
811 #ifdef DEBUG | |
812 printf("rtp_send_data size=%d\n", len); | |
813 #endif | |
814 | |
815 /* build the RTP header */ | |
816 put_byte(&s1->pb, (RTP_VERSION << 6)); | |
826
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817 put_byte(&s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7)); |
0 | 818 put_be16(&s1->pb, s->seq); |
819 put_be32(&s1->pb, s->timestamp); | |
820 put_be32(&s1->pb, s->ssrc); | |
885 | 821 |
0 | 822 put_buffer(&s1->pb, buf1, len); |
823 put_flush_packet(&s1->pb); | |
885 | 824 |
0 | 825 s->seq++; |
826 s->octet_count += len; | |
827 s->packet_count++; | |
828 } | |
829 | |
830 /* send an integer number of samples and compute time stamp and fill | |
831 the rtp send buffer before sending. */ | |
832 static void rtp_send_samples(AVFormatContext *s1, | |
241 | 833 const uint8_t *buf1, int size, int sample_size) |
0 | 834 { |
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835 RTPDemuxContext *s = s1->priv_data; |
0 | 836 int len, max_packet_size, n; |
837 | |
838 max_packet_size = (s->max_payload_size / sample_size) * sample_size; | |
839 /* not needed, but who nows */ | |
840 if ((size % sample_size) != 0) | |
841 av_abort(); | |
842 while (size > 0) { | |
843 len = (max_packet_size - (s->buf_ptr - s->buf)); | |
844 if (len > size) | |
845 len = size; | |
846 | |
847 /* copy data */ | |
848 memcpy(s->buf_ptr, buf1, len); | |
849 s->buf_ptr += len; | |
850 buf1 += len; | |
851 size -= len; | |
852 n = (s->buf_ptr - s->buf); | |
853 /* if buffer full, then send it */ | |
854 if (n >= max_packet_size) { | |
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855 rtp_send_data(s1, s->buf, n, 0); |
0 | 856 s->buf_ptr = s->buf; |
857 /* update timestamp */ | |
858 s->timestamp += n / sample_size; | |
859 } | |
860 } | |
885 | 861 } |
0 | 862 |
863 /* NOTE: we suppose that exactly one frame is given as argument here */ | |
864 /* XXX: test it */ | |
865 static void rtp_send_mpegaudio(AVFormatContext *s1, | |
241 | 866 const uint8_t *buf1, int size) |
0 | 867 { |
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868 RTPDemuxContext *s = s1->priv_data; |
0 | 869 AVStream *st = s1->streams[0]; |
870 int len, count, max_packet_size; | |
871 | |
872 max_packet_size = s->max_payload_size; | |
873 | |
874 /* test if we must flush because not enough space */ | |
875 len = (s->buf_ptr - s->buf); | |
876 if ((len + size) > max_packet_size) { | |
877 if (len > 4) { | |
826
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878 rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); |
0 | 879 s->buf_ptr = s->buf + 4; |
880 /* 90 KHz time stamp */ | |
885 | 881 s->timestamp = s->base_timestamp + |
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882 (s->cur_timestamp * 90000LL) / st->codec->sample_rate; |
0 | 883 } |
884 } | |
885 | |
886 /* add the packet */ | |
887 if (size > max_packet_size) { | |
888 /* big packet: fragment */ | |
889 count = 0; | |
890 while (size > 0) { | |
891 len = max_packet_size - 4; | |
892 if (len > size) | |
893 len = size; | |
894 /* build fragmented packet */ | |
895 s->buf[0] = 0; | |
896 s->buf[1] = 0; | |
897 s->buf[2] = count >> 8; | |
898 s->buf[3] = count; | |
899 memcpy(s->buf + 4, buf1, len); | |
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900 rtp_send_data(s1, s->buf, len + 4, 0); |
0 | 901 size -= len; |
902 buf1 += len; | |
903 count += len; | |
904 } | |
905 } else { | |
906 if (s->buf_ptr == s->buf + 4) { | |
907 /* no fragmentation possible */ | |
908 s->buf[0] = 0; | |
909 s->buf[1] = 0; | |
910 s->buf[2] = 0; | |
911 s->buf[3] = 0; | |
912 } | |
913 memcpy(s->buf_ptr, buf1, size); | |
914 s->buf_ptr += size; | |
915 } | |
820
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916 s->cur_timestamp += st->codec->frame_size; |
0 | 917 } |
918 | |
919 /* NOTE: a single frame must be passed with sequence header if | |
920 needed. XXX: use slices. */ | |
921 static void rtp_send_mpegvideo(AVFormatContext *s1, | |
241 | 922 const uint8_t *buf1, int size) |
0 | 923 { |
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924 RTPDemuxContext *s = s1->priv_data; |
0 | 925 AVStream *st = s1->streams[0]; |
926 int len, h, max_packet_size; | |
65 | 927 uint8_t *q; |
0 | 928 |
929 max_packet_size = s->max_payload_size; | |
930 | |
931 while (size > 0) { | |
932 /* XXX: more correct headers */ | |
933 h = 0; | |
820
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934 if (st->codec->sub_id == 2) |
0 | 935 h |= 1 << 26; /* mpeg 2 indicator */ |
936 q = s->buf; | |
937 *q++ = h >> 24; | |
938 *q++ = h >> 16; | |
939 *q++ = h >> 8; | |
940 *q++ = h; | |
941 | |
820
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michael
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942 if (st->codec->sub_id == 2) { |
0 | 943 h = 0; |
944 *q++ = h >> 24; | |
945 *q++ = h >> 16; | |
946 *q++ = h >> 8; | |
947 *q++ = h; | |
948 } | |
885 | 949 |
0 | 950 len = max_packet_size - (q - s->buf); |
951 if (len > size) | |
952 len = size; | |
953 | |
954 memcpy(q, buf1, len); | |
955 q += len; | |
956 | |
957 /* 90 KHz time stamp */ | |
885 | 958 s->timestamp = s->base_timestamp + |
820
feca73904e67
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michael
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959 av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps |
827
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michael
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826
diff
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960 rtp_send_data(s1, s->buf, q - s->buf, (len == size)); |
0 | 961 |
962 buf1 += len; | |
963 size -= len; | |
964 } | |
965 s->cur_timestamp++; | |
966 } | |
967 | |
968 static void rtp_send_raw(AVFormatContext *s1, | |
241 | 969 const uint8_t *buf1, int size) |
0 | 970 { |
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971 RTPDemuxContext *s = s1->priv_data; |
0 | 972 AVStream *st = s1->streams[0]; |
973 int len, max_packet_size; | |
974 | |
975 max_packet_size = s->max_payload_size; | |
976 | |
977 while (size > 0) { | |
978 len = max_packet_size; | |
979 if (len > size) | |
980 len = size; | |
981 | |
982 /* 90 KHz time stamp */ | |
885 | 983 s->timestamp = s->base_timestamp + |
820
feca73904e67
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michael
parents:
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984 av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps |
826
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michael
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985 rtp_send_data(s1, buf1, len, (len == size)); |
0 | 986 |
987 buf1 += len; | |
988 size -= len; | |
989 } | |
990 s->cur_timestamp++; | |
991 } | |
992 | |
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993 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */ |
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994 static void rtp_send_mpegts_raw(AVFormatContext *s1, |
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995 const uint8_t *buf1, int size) |
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996 { |
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997 RTPDemuxContext *s = s1->priv_data; |
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998 int len, out_len; |
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999 |
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1000 while (size >= TS_PACKET_SIZE) { |
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1001 len = s->max_payload_size - (s->buf_ptr - s->buf); |
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1002 if (len > size) |
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1003 len = size; |
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1004 memcpy(s->buf_ptr, buf1, len); |
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1005 buf1 += len; |
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1006 size -= len; |
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1007 s->buf_ptr += len; |
885 | 1008 |
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1009 out_len = s->buf_ptr - s->buf; |
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1010 if (out_len >= s->max_payload_size) { |
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1011 rtp_send_data(s1, s->buf, out_len, 0); |
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1012 s->buf_ptr = s->buf; |
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1013 } |
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1014 } |
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1015 } |
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1016 |
0 | 1017 /* write an RTP packet. 'buf1' must contain a single specific frame. */ |
468 | 1018 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) |
0 | 1019 { |
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1020 RTPDemuxContext *s = s1->priv_data; |
0 | 1021 AVStream *st = s1->streams[0]; |
1022 int rtcp_bytes; | |
65 | 1023 int64_t ntp_time; |
468 | 1024 int size= pkt->size; |
1025 uint8_t *buf1= pkt->data; | |
885 | 1026 |
0 | 1027 #ifdef DEBUG |
468 | 1028 printf("%d: write len=%d\n", pkt->stream_index, size); |
0 | 1029 #endif |
1030 | |
1031 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */ | |
885 | 1032 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / |
0 | 1033 RTCP_TX_RATIO_DEN; |
1034 if (s->first_packet || rtcp_bytes >= 28) { | |
1035 /* compute NTP time */ | |
173 | 1036 /* XXX: 90 kHz timestamp hardcoded */ |
468 | 1037 ntp_time = (pkt->pts << 28) / 5625; |
885 | 1038 rtcp_send_sr(s1, ntp_time); |
0 | 1039 s->last_octet_count = s->octet_count; |
1040 s->first_packet = 0; | |
1041 } | |
1042 | |
820
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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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|
1043 switch(st->codec->codec_id) { |
0 | 1044 case CODEC_ID_PCM_MULAW: |
1045 case CODEC_ID_PCM_ALAW: | |
1046 case CODEC_ID_PCM_U8: | |
1047 case CODEC_ID_PCM_S8: | |
820
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1048 rtp_send_samples(s1, buf1, size, 1 * st->codec->channels); |
0 | 1049 break; |
1050 case CODEC_ID_PCM_U16BE: | |
1051 case CODEC_ID_PCM_U16LE: | |
1052 case CODEC_ID_PCM_S16BE: | |
1053 case CODEC_ID_PCM_S16LE: | |
820
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777
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|
1054 rtp_send_samples(s1, buf1, size, 2 * st->codec->channels); |
0 | 1055 break; |
1056 case CODEC_ID_MP2: | |
232 | 1057 case CODEC_ID_MP3: |
0 | 1058 rtp_send_mpegaudio(s1, buf1, size); |
1059 break; | |
1060 case CODEC_ID_MPEG1VIDEO: | |
1061 rtp_send_mpegvideo(s1, buf1, size); | |
1062 break; | |
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1063 case CODEC_ID_MPEG2TS: |
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1064 rtp_send_mpegts_raw(s1, buf1, size); |
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1065 break; |
0 | 1066 default: |
1067 /* better than nothing : send the codec raw data */ | |
1068 rtp_send_raw(s1, buf1, size); | |
1069 break; | |
1070 } | |
1071 return 0; | |
1072 } | |
1073 | |
1074 static int rtp_write_trailer(AVFormatContext *s1) | |
1075 { | |
294
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added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
1076 // RTPDemuxContext *s = s1->priv_data; |
0 | 1077 return 0; |
1078 } | |
1079 | |
1167 | 1080 AVOutputFormat rtp_muxer = { |
0 | 1081 "rtp", |
1082 "RTP output format", | |
1083 NULL, | |
1084 NULL, | |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
1085 sizeof(RTPDemuxContext), |
0 | 1086 CODEC_ID_PCM_MULAW, |
1087 CODEC_ID_NONE, | |
1088 rtp_write_header, | |
1089 rtp_write_packet, | |
1090 rtp_write_trailer, | |
1091 }; |