Mercurial > libavformat.hg
annotate rtp.c @ 2702:6a7b82888880 libavformat
Enable debug variable only when debug code is enabled, fixes the warning:
rtp_h264.c: In function ¡Æh264_handle_packet¡Ç:
rtp_h264.c:168: warning: unused variable ¡Ædata¡Ç
author | diego |
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date | Sun, 04 Nov 2007 12:46:16 +0000 |
parents | d751acab2622 |
children | cc693f9e80ee |
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> | |
1754 | 26 #include "network.h" |
0 | 27 |
1419 | 28 #include "rtp_internal.h" |
1460 | 29 #include "rtp_h264.h" |
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30 #include "rtp_mpv.h" |
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31 #include "rtp_aac.h" |
1419 | 32 |
0 | 33 //#define DEBUG |
34 | |
35 | |
36 /* TODO: - add RTCP statistics reporting (should be optional). | |
37 | |
38 - add support for h263/mpeg4 packetized output : IDEA: send a | |
39 buffer to 'rtp_write_packet' contains all the packets for ONE | |
40 frame. Each packet should have a four byte header containing | |
41 the length in big endian format (same trick as | |
885 | 42 'url_open_dyn_packet_buf') |
0 | 43 */ |
44 | |
774 | 45 /* from http://www.iana.org/assignments/rtp-parameters last updated 05 January 2005 */ |
46 AVRtpPayloadType_t AVRtpPayloadTypes[]= | |
47 { | |
48 {0, "PCMU", CODEC_TYPE_AUDIO, CODEC_ID_PCM_MULAW, 8000, 1}, | |
49 {1, "Reserved", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
50 {2, "Reserved", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
51 {3, "GSM", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
52 {4, "G723", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
53 {5, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
54 {6, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 16000, 1}, | |
55 {7, "LPC", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
56 {8, "PCMA", CODEC_TYPE_AUDIO, CODEC_ID_PCM_ALAW, 8000, 1}, | |
57 {9, "G722", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
58 {10, "L16", CODEC_TYPE_AUDIO, CODEC_ID_PCM_S16BE, 44100, 2}, | |
59 {11, "L16", CODEC_TYPE_AUDIO, CODEC_ID_PCM_S16BE, 44100, 1}, | |
1259 | 60 {12, "QCELP", CODEC_TYPE_AUDIO, CODEC_ID_QCELP, 8000, 1}, |
774 | 61 {13, "CN", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, |
62 {14, "MPA", CODEC_TYPE_AUDIO, CODEC_ID_MP2, 90000, -1}, | |
63 {15, "G728", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
64 {16, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 11025, 1}, | |
65 {17, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 22050, 1}, | |
66 {18, "G729", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
67 {19, "reserved", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, | |
68 {20, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, | |
69 {21, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, | |
70 {22, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, | |
71 {23, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, | |
72 {24, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1}, | |
73 {25, "CelB", CODEC_TYPE_VIDEO, CODEC_ID_NONE, 90000, -1}, | |
74 {26, "JPEG", CODEC_TYPE_VIDEO, CODEC_ID_MJPEG, 90000, -1}, | |
75 {27, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1}, | |
76 {28, "nv", CODEC_TYPE_VIDEO, CODEC_ID_NONE, 90000, -1}, | |
77 {29, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1}, | |
78 {30, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1}, | |
79 {31, "H261", CODEC_TYPE_VIDEO, CODEC_ID_H261, 90000, -1}, | |
80 {32, "MPV", CODEC_TYPE_VIDEO, CODEC_ID_MPEG1VIDEO, 90000, -1}, | |
81 {33, "MP2T", CODEC_TYPE_DATA, CODEC_ID_MPEG2TS, 90000, -1}, | |
82 {34, "H263", CODEC_TYPE_VIDEO, CODEC_ID_H263, 90000, -1}, | |
83 {35, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
84 {36, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
85 {37, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
86 {38, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
87 {39, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
88 {40, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
89 {41, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
90 {42, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
91 {43, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
92 {44, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
93 {45, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
94 {46, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
95 {47, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
96 {48, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
97 {49, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
98 {50, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
99 {51, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
100 {52, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
101 {53, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
102 {54, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
103 {55, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
104 {56, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
105 {57, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
106 {58, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
107 {59, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
108 {60, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
109 {61, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
110 {62, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
111 {63, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
112 {64, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
113 {65, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
114 {66, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
115 {67, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
116 {68, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
117 {69, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
118 {70, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
119 {71, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
120 {72, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
121 {73, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
122 {74, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
123 {75, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
124 {76, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
125 {77, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
126 {78, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
127 {79, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
128 {80, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
129 {81, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
130 {82, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
131 {83, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
132 {84, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
133 {85, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
134 {86, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
135 {87, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
136 {88, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
137 {89, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
138 {90, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
139 {91, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
140 {92, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
141 {93, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
142 {94, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
143 {95, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
144 {96, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
145 {97, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
146 {98, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
147 {99, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
148 {100, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
149 {101, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
150 {102, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
151 {103, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
152 {104, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
153 {105, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
154 {106, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
155 {107, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
156 {108, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
157 {109, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
158 {110, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
159 {111, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
160 {112, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
161 {113, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
162 {114, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
163 {115, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
164 {116, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
165 {117, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
166 {118, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
167 {119, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
168 {120, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
169 {121, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
170 {122, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
171 {123, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
172 {124, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
173 {125, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
174 {126, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
175 {127, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
176 {-1, "", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1} | |
177 }; | |
0 | 178 |
1419 | 179 /* statistics functions */ |
180 RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL; | |
181 | |
182 static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4}; | |
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183 static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC}; |
0 | 184 |
1419 | 185 static void register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler) |
186 { | |
187 handler->next= RTPFirstDynamicPayloadHandler; | |
188 RTPFirstDynamicPayloadHandler= handler; | |
189 } | |
885 | 190 |
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191 void av_register_rtp_dynamic_payload_handlers(void) |
1419 | 192 { |
193 register_dynamic_payload_handler(&mp4v_es_handler); | |
194 register_dynamic_payload_handler(&mpeg4_generic_handler); | |
195 register_dynamic_payload_handler(&ff_h264_dynamic_handler); | |
196 } | |
0 | 197 |
198 int rtp_get_codec_info(AVCodecContext *codec, int payload_type) | |
199 { | |
774 | 200 if (AVRtpPayloadTypes[payload_type].codec_id != CODEC_ID_NONE) { |
201 codec->codec_type = AVRtpPayloadTypes[payload_type].codec_type; | |
1116 | 202 codec->codec_id = AVRtpPayloadTypes[payload_type].codec_id; |
774 | 203 if (AVRtpPayloadTypes[payload_type].audio_channels > 0) |
204 codec->channels = AVRtpPayloadTypes[payload_type].audio_channels; | |
205 if (AVRtpPayloadTypes[payload_type].clock_rate > 0) | |
206 codec->sample_rate = AVRtpPayloadTypes[payload_type].clock_rate; | |
207 return 0; | |
0 | 208 } |
774 | 209 return -1; |
0 | 210 } |
211 | |
212 int rtp_get_payload_type(AVCodecContext *codec) | |
213 { | |
774 | 214 int i, payload_type; |
0 | 215 |
216 /* compute the payload type */ | |
774 | 217 for (payload_type = -1, i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i) |
218 if (AVRtpPayloadTypes[i].codec_id == codec->codec_id) { | |
219 if (codec->codec_id == CODEC_ID_PCM_S16BE) | |
220 if (codec->channels != AVRtpPayloadTypes[i].audio_channels) | |
221 continue; | |
222 payload_type = AVRtpPayloadTypes[i].pt; | |
0 | 223 } |
224 return payload_type; | |
225 } | |
226 | |
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227 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len) |
0 | 228 { |
229 if (buf[1] != 200) | |
230 return -1; | |
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231 s->last_rtcp_ntp_time = AV_RB64(buf + 8); |
173 | 232 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) |
233 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time; | |
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234 s->last_rtcp_timestamp = AV_RB32(buf + 16); |
0 | 235 return 0; |
236 } | |
237 | |
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238 #define RTP_SEQ_MOD (1<<16) |
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239 |
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240 /** |
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241 * called on parse open packet |
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242 */ |
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243 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet. |
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244 { |
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245 memset(s, 0, sizeof(RTPStatistics)); |
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246 s->max_seq= base_sequence; |
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247 s->probation= 1; |
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248 } |
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249 |
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250 /** |
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251 * called whenever there is a large jump in sequence numbers, or when they get out of probation... |
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252 */ |
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253 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq) |
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254 { |
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255 s->max_seq= seq; |
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256 s->cycles= 0; |
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257 s->base_seq= seq -1; |
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258 s->bad_seq= RTP_SEQ_MOD + 1; |
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259 s->received= 0; |
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260 s->expected_prior= 0; |
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261 s->received_prior= 0; |
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262 s->jitter= 0; |
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263 s->transit= 0; |
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264 } |
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265 |
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266 /** |
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267 * returns 1 if we should handle this packet. |
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268 */ |
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269 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq) |
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270 { |
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271 uint16_t udelta= seq - s->max_seq; |
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272 const int MAX_DROPOUT= 3000; |
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273 const int MAX_MISORDER = 100; |
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274 const int MIN_SEQUENTIAL = 2; |
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275 |
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276 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */ |
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277 if(s->probation) |
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278 { |
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279 if(seq==s->max_seq + 1) { |
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280 s->probation--; |
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281 s->max_seq= seq; |
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282 if(s->probation==0) { |
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283 rtp_init_sequence(s, seq); |
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284 s->received++; |
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285 return 1; |
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286 } |
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287 } else { |
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288 s->probation= MIN_SEQUENTIAL - 1; |
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289 s->max_seq = seq; |
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290 } |
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291 } else if (udelta < MAX_DROPOUT) { |
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292 // in order, with permissible gap |
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293 if(seq < s->max_seq) { |
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294 //sequence number wrapped; count antother 64k cycles |
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295 s->cycles += RTP_SEQ_MOD; |
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296 } |
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297 s->max_seq= seq; |
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298 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) { |
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299 // sequence made a large jump... |
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300 if(seq==s->bad_seq) { |
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301 // two sequential packets-- assume that the other side restarted without telling us; just resync. |
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302 rtp_init_sequence(s, seq); |
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303 } else { |
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304 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1); |
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305 return 0; |
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306 } |
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307 } else { |
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308 // duplicate or reordered packet... |
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309 } |
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310 s->received++; |
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311 return 1; |
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312 } |
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313 |
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314 #if 0 |
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315 /** |
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316 * This function is currently unused; without a valid local ntp time, I don't see how we could calculate the |
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317 * difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values |
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318 * never change. I left this in in case someone else can see a way. (rdm) |
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319 */ |
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320 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp) |
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321 { |
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322 uint32_t transit= arrival_timestamp - sent_timestamp; |
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323 int d; |
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324 s->transit= transit; |
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325 d= FFABS(transit - s->transit); |
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326 s->jitter += d - ((s->jitter + 8)>>4); |
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327 } |
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328 #endif |
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329 |
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330 int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count) |
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331 { |
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332 ByteIOContext pb; |
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333 uint8_t *buf; |
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334 int len; |
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335 int rtcp_bytes; |
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336 RTPStatistics *stats= &s->statistics; |
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337 uint32_t lost; |
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338 uint32_t extended_max; |
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339 uint32_t expected_interval; |
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340 uint32_t received_interval; |
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341 uint32_t lost_interval; |
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342 uint32_t expected; |
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343 uint32_t fraction; |
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344 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time? |
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345 |
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346 if (!s->rtp_ctx || (count < 1)) |
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347 return -1; |
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348 |
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349 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */ |
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350 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */ |
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351 s->octet_count += count; |
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352 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / |
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353 RTCP_TX_RATIO_DEN; |
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354 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !? |
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355 if (rtcp_bytes < 28) |
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356 return -1; |
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357 s->last_octet_count = s->octet_count; |
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358 |
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359 if (url_open_dyn_buf(&pb) < 0) |
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360 return -1; |
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361 |
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362 // Receiver Report |
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363 put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */ |
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364 put_byte(&pb, 201); |
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365 put_be16(&pb, 7); /* length in words - 1 */ |
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366 put_be32(&pb, s->ssrc); // our own SSRC |
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367 put_be32(&pb, s->ssrc); // XXX: should be the server's here! |
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368 // some placeholders we should really fill... |
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369 // RFC 1889/p64 |
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370 extended_max= stats->cycles + stats->max_seq; |
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371 expected= extended_max - stats->base_seq + 1; |
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372 lost= expected - stats->received; |
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373 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits... |
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374 expected_interval= expected - stats->expected_prior; |
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375 stats->expected_prior= expected; |
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376 received_interval= stats->received - stats->received_prior; |
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377 stats->received_prior= stats->received; |
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378 lost_interval= expected_interval - received_interval; |
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379 if (expected_interval==0 || lost_interval<=0) fraction= 0; |
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380 else fraction = (lost_interval<<8)/expected_interval; |
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381 |
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382 fraction= (fraction<<24) | lost; |
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383 |
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384 put_be32(&pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */ |
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385 put_be32(&pb, extended_max); /* max sequence received */ |
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386 put_be32(&pb, stats->jitter>>4); /* jitter */ |
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387 |
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388 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE) |
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389 { |
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390 put_be32(&pb, 0); /* last SR timestamp */ |
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391 put_be32(&pb, 0); /* delay since last SR */ |
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392 } else { |
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393 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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394 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time; |
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395 |
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396 put_be32(&pb, middle_32_bits); /* last SR timestamp */ |
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397 put_be32(&pb, delay_since_last); /* delay since last SR */ |
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398 } |
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399 |
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400 // CNAME |
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401 put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */ |
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402 put_byte(&pb, 202); |
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403 len = strlen(s->hostname); |
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404 put_be16(&pb, (6 + len + 3) / 4); /* length in words - 1 */ |
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405 put_be32(&pb, s->ssrc); |
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406 put_byte(&pb, 0x01); |
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407 put_byte(&pb, len); |
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408 put_buffer(&pb, s->hostname, len); |
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409 // padding |
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410 for (len = (6 + len) % 4; len % 4; len++) { |
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411 put_byte(&pb, 0); |
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412 } |
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413 |
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414 put_flush_packet(&pb); |
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415 len = url_close_dyn_buf(&pb, &buf); |
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416 if ((len > 0) && buf) { |
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417 int result; |
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418 #if defined(DEBUG) |
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419 printf("sending %d bytes of RR\n", len); |
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420 #endif |
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421 result= url_write(s->rtp_ctx, buf, len); |
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422 #if defined(DEBUG) |
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423 printf("result from url_write: %d\n", result); |
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424 #endif |
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425 av_free(buf); |
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426 } |
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427 return 0; |
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428 } |
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429 |
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430 /** |
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431 * open a new RTP parse context for stream 'st'. 'st' can be NULL for |
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432 * MPEG2TS streams to indicate that they should be demuxed inside the |
885 | 433 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned) |
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434 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system. |
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435 */ |
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436 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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437 { |
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438 RTPDemuxContext *s; |
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439 |
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440 s = av_mallocz(sizeof(RTPDemuxContext)); |
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441 if (!s) |
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442 return NULL; |
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443 s->payload_type = payload_type; |
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444 s->last_rtcp_ntp_time = AV_NOPTS_VALUE; |
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445 s->first_rtcp_ntp_time = AV_NOPTS_VALUE; |
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446 s->ic = s1; |
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447 s->st = st; |
774 | 448 s->rtp_payload_data = rtp_payload_data; |
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449 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp? |
774 | 450 if (!strcmp(AVRtpPayloadTypes[payload_type].enc_name, "MP2T")) { |
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451 s->ts = mpegts_parse_open(s->ic); |
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452 if (s->ts == NULL) { |
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453 av_free(s); |
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454 return NULL; |
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455 } |
305 | 456 } else { |
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457 switch(st->codec->codec_id) { |
305 | 458 case CODEC_ID_MPEG1VIDEO: |
459 case CODEC_ID_MPEG2VIDEO: | |
460 case CODEC_ID_MP2: | |
461 case CODEC_ID_MP3: | |
462 case CODEC_ID_MPEG4: | |
1419 | 463 case CODEC_ID_H264: |
2023 | 464 st->need_parsing = AVSTREAM_PARSE_FULL; |
305 | 465 break; |
466 default: | |
467 break; | |
468 } | |
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469 } |
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470 // needed to send back RTCP RR in RTSP sessions |
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471 s->rtp_ctx = rtpc; |
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472 gethostname(s->hostname, sizeof(s->hostname)); |
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473 return s; |
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474 } |
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475 |
774 | 476 static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf) |
477 { | |
478 int au_headers_length, au_header_size, i; | |
479 GetBitContext getbitcontext; | |
480 rtp_payload_data_t *infos; | |
481 | |
482 infos = s->rtp_payload_data; | |
483 | |
484 if (infos == NULL) | |
485 return -1; | |
486 | |
487 /* decode the first 2 bytes where are stored the AUHeader sections | |
488 length in bits */ | |
1673 | 489 au_headers_length = AV_RB16(buf); |
774 | 490 |
491 if (au_headers_length > RTP_MAX_PACKET_LENGTH) | |
492 return -1; | |
493 | |
494 infos->au_headers_length_bytes = (au_headers_length + 7) / 8; | |
495 | |
496 /* skip AU headers length section (2 bytes) */ | |
497 buf += 2; | |
498 | |
499 init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8); | |
500 | |
501 /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */ | |
502 au_header_size = infos->sizelength + infos->indexlength; | |
503 if (au_header_size <= 0 || (au_headers_length % au_header_size != 0)) | |
504 return -1; | |
505 | |
506 infos->nb_au_headers = au_headers_length / au_header_size; | |
507 infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers); | |
508 | |
509 /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving) | |
2164 | 510 In my test, the FAAD decoder does not behave correctly when sending each AU one by one |
774 | 511 but does when sending the whole as one big packet... */ |
512 infos->au_headers[0].size = 0; | |
513 infos->au_headers[0].index = 0; | |
514 for (i = 0; i < infos->nb_au_headers; ++i) { | |
515 infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength); | |
516 infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength); | |
517 } | |
518 | |
519 infos->nb_au_headers = 1; | |
520 | |
521 return 0; | |
522 } | |
523 | |
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524 /** |
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525 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc. |
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526 */ |
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527 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp) |
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528 { |
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529 switch(s->st->codec->codec_id) { |
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530 case CODEC_ID_MP2: |
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531 case CODEC_ID_MPEG1VIDEO: |
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532 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) { |
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533 int64_t addend; |
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534 |
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535 int delta_timestamp; |
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536 /* XXX: is it really necessary to unify the timestamp base ? */ |
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537 /* compute pts from timestamp with received ntp_time */ |
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538 delta_timestamp = timestamp - s->last_rtcp_timestamp; |
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539 /* convert to 90 kHz without overflow */ |
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540 addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14; |
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541 addend = (addend * 5625) >> 14; |
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542 pkt->pts = addend + delta_timestamp; |
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543 } |
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544 break; |
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545 case CODEC_ID_AAC: |
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546 case CODEC_ID_H264: |
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547 case CODEC_ID_MPEG4: |
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548 pkt->pts = timestamp; |
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549 break; |
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550 default: |
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551 /* no timestamp info yet */ |
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552 break; |
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553 } |
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554 pkt->stream_index = s->st->index; |
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555 } |
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556 |
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557 /** |
885 | 558 * Parse an RTP or RTCP packet directly sent as a buffer. |
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559 * @param s RTP parse context. |
0 | 560 * @param pkt returned packet |
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561 * @param buf input buffer or NULL to read the next packets |
0 | 562 * @param len buffer len |
885 | 563 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow |
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564 * (use buf as NULL to read the next). -1 if no packet (error or no more packet). |
0 | 565 */ |
885 | 566 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, |
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567 const uint8_t *buf, int len) |
0 | 568 { |
569 unsigned int ssrc, h; | |
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570 int payload_type, seq, ret; |
0 | 571 AVStream *st; |
65 | 572 uint32_t timestamp; |
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573 int rv= 0; |
885 | 574 |
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575 if (!buf) { |
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576 /* return the next packets, if any */ |
1419 | 577 if(s->st && s->parse_packet) { |
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|
578 timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned.... |
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
579 rv= s->parse_packet(s, pkt, ×tamp, NULL, 0); |
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
580 finalize_packet(s, pkt, timestamp); |
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
581 return rv; |
1419 | 582 } else { |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
583 // TODO: Move to a dynamic packet handler (like above) |
1424
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
584 if (s->read_buf_index >= s->read_buf_size) |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
585 return -1; |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
586 ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index, |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
587 s->read_buf_size - s->read_buf_index); |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
588 if (ret < 0) |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
589 return -1; |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
590 s->read_buf_index += ret; |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
591 if (s->read_buf_index < s->read_buf_size) |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
592 return 1; |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
593 else |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
594 return 0; |
1419 | 595 } |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
596 } |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
597 |
0 | 598 if (len < 12) |
599 return -1; | |
600 | |
601 if ((buf[0] & 0xc0) != (RTP_VERSION << 6)) | |
602 return -1; | |
603 if (buf[1] >= 200 && buf[1] <= 204) { | |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
604 rtcp_parse_packet(s, buf, len); |
0 | 605 return -1; |
606 } | |
607 payload_type = buf[1] & 0x7f; | |
2222 | 608 seq = AV_RB16(buf + 2); |
2167
c4244900f986
remove decode_be32/64 and replace them by AV_RB32/64
benoit
parents:
2164
diff
changeset
|
609 timestamp = AV_RB32(buf + 4); |
c4244900f986
remove decode_be32/64 and replace them by AV_RB32/64
benoit
parents:
2164
diff
changeset
|
610 ssrc = AV_RB32(buf + 8); |
1425
00d9393a126f
make ffmpeg able to send back a RTCP receiver report.
gpoirier
parents:
1424
diff
changeset
|
611 /* store the ssrc in the RTPDemuxContext */ |
00d9393a126f
make ffmpeg able to send back a RTCP receiver report.
gpoirier
parents:
1424
diff
changeset
|
612 s->ssrc = ssrc; |
885 | 613 |
0 | 614 /* NOTE: we can handle only one payload type */ |
615 if (s->payload_type != payload_type) | |
616 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
|
617 |
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
|
618 st = s->st; |
1445
db97355877b1
add valid statistics for the RTCP receiver report.
gpoirier
parents:
1443
diff
changeset
|
619 // only do something with this if all the rtp checks pass... |
db97355877b1
add valid statistics for the RTCP receiver report.
gpoirier
parents:
1443
diff
changeset
|
620 if(!rtp_valid_packet_in_sequence(&s->statistics, seq)) |
db97355877b1
add valid statistics for the RTCP receiver report.
gpoirier
parents:
1443
diff
changeset
|
621 { |
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
|
622 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n", |
0 | 623 payload_type, seq, ((s->seq + 1) & 0xffff)); |
1445
db97355877b1
add valid statistics for the RTCP receiver report.
gpoirier
parents:
1443
diff
changeset
|
624 return -1; |
0 | 625 } |
1445
db97355877b1
add valid statistics for the RTCP receiver report.
gpoirier
parents:
1443
diff
changeset
|
626 |
0 | 627 s->seq = seq; |
628 len -= 12; | |
629 buf += 12; | |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
630 |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
631 if (!st) { |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
632 /* specific MPEG2TS demux support */ |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
633 ret = mpegts_parse_packet(s->ts, pkt, buf, len); |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
634 if (ret < 0) |
0 | 635 return -1; |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
636 if (ret < len) { |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
637 s->read_buf_size = len - ret; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
638 memcpy(s->buf, buf + ret, s->read_buf_size); |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
639 s->read_buf_index = 0; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
640 return 1; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
641 } |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
642 } else { |
1419 | 643 // 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
|
644 switch(st->codec->codec_id) { |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
645 case CODEC_ID_MP2: |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
646 /* better than nothing: skip mpeg audio RTP header */ |
0 | 647 if (len <= 4) |
648 return -1; | |
2167
c4244900f986
remove decode_be32/64 and replace them by AV_RB32/64
benoit
parents:
2164
diff
changeset
|
649 h = AV_RB32(buf); |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
650 len -= 4; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
651 buf += 4; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
652 av_new_packet(pkt, len); |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
653 memcpy(pkt->data, buf, len); |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
654 break; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
655 case CODEC_ID_MPEG1VIDEO: |
305 | 656 /* better than nothing: skip mpeg video RTP header */ |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
657 if (len <= 4) |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
658 return -1; |
2167
c4244900f986
remove decode_be32/64 and replace them by AV_RB32/64
benoit
parents:
2164
diff
changeset
|
659 h = AV_RB32(buf); |
0 | 660 buf += 4; |
661 len -= 4; | |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
662 if (h & (1 << 26)) { |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
663 /* mpeg2 */ |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
664 if (len <= 4) |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
665 return -1; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
666 buf += 4; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
667 len -= 4; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
668 } |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
669 av_new_packet(pkt, len); |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
670 memcpy(pkt->data, buf, len); |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
671 break; |
1427
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
672 // 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
|
673 // timestamps. |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
674 // 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
|
675 case CODEC_ID_AAC: |
1427
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
676 if (rtp_parse_mp4_au(s, buf)) |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
677 return -1; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
678 { |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
679 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
|
680 if (infos == NULL) |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
681 return -1; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
682 buf += infos->au_headers_length_bytes + 2; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
683 len -= infos->au_headers_length_bytes + 2; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
684 |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
685 /* 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
|
686 one au_header */ |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
687 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
|
688 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
|
689 buf += infos->au_headers[0].size; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
690 len -= infos->au_headers[0].size; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
691 } |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
692 s->read_buf_size = len; |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
693 rv= 0; |
1427
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
694 break; |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
695 default: |
1419 | 696 if(s->parse_packet) { |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
697 rv= s->parse_packet(s, pkt, ×tamp, buf, len); |
1419 | 698 } else { |
1424
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
699 av_new_packet(pkt, len); |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
700 memcpy(pkt->data, buf, len); |
1419 | 701 } |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
702 break; |
0 | 703 } |
885 | 704 |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
705 // now perform timestamp things.... |
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
706 finalize_packet(s, pkt, timestamp); |
0 | 707 } |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
708 return rv; |
0 | 709 } |
710 | |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
711 void rtp_parse_close(RTPDemuxContext *s) |
0 | 712 { |
1419 | 713 // TODO: fold this into the protocol specific data fields. |
774 | 714 if (!strcmp(AVRtpPayloadTypes[s->payload_type].enc_name, "MP2T")) { |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
715 mpegts_parse_close(s->ts); |
0 | 716 } |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
717 av_free(s); |
0 | 718 } |
719 | |
720 /* rtp output */ | |
721 | |
722 static int rtp_write_header(AVFormatContext *s1) | |
723 { | |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
724 RTPDemuxContext *s = s1->priv_data; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
725 int payload_type, max_packet_size, n; |
0 | 726 AVStream *st; |
727 | |
728 if (s1->nb_streams != 1) | |
729 return -1; | |
730 st = s1->streams[0]; | |
731 | |
820
feca73904e67
changing AVCodecContext codec -> *codec in AVStream so additions to AVCodecContext dont randomize AVStream and break binary compatibility
michael
parents:
777
diff
changeset
|
732 payload_type = rtp_get_payload_type(st->codec); |
0 | 733 if (payload_type < 0) |
734 payload_type = RTP_PT_PRIVATE; /* private payload type */ | |
735 s->payload_type = payload_type; | |
736 | |
1099 | 737 // 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
|
738 s->base_timestamp = 0; /* FIXME: was random(), what should this be? */ |
0 | 739 s->timestamp = s->base_timestamp; |
2540
ca3cba9c641f
Fix timestamps in RTP packets (now, MPEG1 video with B frames works correctly)
lucabe
parents:
2539
diff
changeset
|
740 s->cur_timestamp = 0; |
1045
7f1b7f811f01
fix constraint violation: libavformat is not allowed to modify state of caller, including rng state
rfelker
parents:
986
diff
changeset
|
741 s->ssrc = 0; /* FIXME: was random(), what should this be? */ |
0 | 742 s->first_packet = 1; |
2539
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743 s->first_rtcp_ntp_time = AV_NOPTS_VALUE; |
0 | 744 |
745 max_packet_size = url_fget_max_packet_size(&s1->pb); | |
746 if (max_packet_size <= 12) | |
2274
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takis
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2222
diff
changeset
|
747 return AVERROR(EIO); |
0 | 748 s->max_payload_size = max_packet_size - 12; |
749 | |
2587
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|
750 s->max_frames_per_packet = 0; |
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751 if (s1->max_delay) { |
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diff
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|
752 if (st->codec->codec_type == CODEC_TYPE_AUDIO) { |
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|
753 if (st->codec->frame_size == 0) { |
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diff
changeset
|
754 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n"); |
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755 } else { |
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756 s->max_frames_per_packet = av_rescale_rnd(s1->max_delay, st->codec->sample_rate, AV_TIME_BASE * st->codec->frame_size, AV_ROUND_DOWN); |
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757 } |
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758 } |
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|
759 if (st->codec->codec_type == CODEC_TYPE_VIDEO) { |
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760 /* FIXME: We should round down here... */ |
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761 s->max_frames_per_packet = av_rescale_q(s1->max_delay, AV_TIME_BASE_Q, st->codec->time_base); |
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762 } |
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763 } |
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|
764 |
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|
765 av_set_pts_info(st, 32, 1, 90000); |
820
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changing AVCodecContext codec -> *codec in AVStream so additions to AVCodecContext dont randomize AVStream and break binary compatibility
michael
parents:
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766 switch(st->codec->codec_id) { |
0 | 767 case CODEC_ID_MP2: |
232 | 768 case CODEC_ID_MP3: |
0 | 769 s->buf_ptr = s->buf + 4; |
770 break; | |
771 case CODEC_ID_MPEG1VIDEO: | |
772 break; | |
294
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773 case CODEC_ID_MPEG2TS: |
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774 n = s->max_payload_size / TS_PACKET_SIZE; |
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775 if (n < 1) |
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776 n = 1; |
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777 s->max_payload_size = n * TS_PACKET_SIZE; |
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778 s->buf_ptr = s->buf; |
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779 break; |
2550
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780 case CODEC_ID_AAC: |
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781 s->read_buf_index = 0; |
0 | 782 default: |
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783 if (st->codec->codec_type == CODEC_TYPE_AUDIO) { |
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784 av_set_pts_info(st, 32, 1, st->codec->sample_rate); |
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785 } |
0 | 786 s->buf_ptr = s->buf; |
787 break; | |
788 } | |
789 | |
790 return 0; | |
791 } | |
792 | |
793 /* send an rtcp sender report packet */ | |
65 | 794 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time) |
0 | 795 { |
294
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796 RTPDemuxContext *s = s1->priv_data; |
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797 uint32_t rtp_ts; |
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|
798 |
0 | 799 #if defined(DEBUG) |
1443
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diego
parents:
1431
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|
800 printf("RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp); |
0 | 801 #endif |
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802 |
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803 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = ntp_time; |
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diff
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|
804 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, AV_TIME_BASE_Q, |
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parents:
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805 s1->streams[0]->time_base) + s->base_timestamp; |
0 | 806 put_byte(&s1->pb, (RTP_VERSION << 6)); |
807 put_byte(&s1->pb, 200); | |
808 put_be16(&s1->pb, 6); /* length in words - 1 */ | |
809 put_be32(&s1->pb, s->ssrc); | |
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diff
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810 put_be32(&s1->pb, ntp_time / 1000000); |
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811 put_be32(&s1->pb, ((ntp_time % 1000000) << 32) / 1000000); |
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812 put_be32(&s1->pb, rtp_ts); |
0 | 813 put_be32(&s1->pb, s->packet_count); |
814 put_be32(&s1->pb, s->octet_count); | |
815 put_flush_packet(&s1->pb); | |
816 } | |
817 | |
818 /* send an rtp packet. sequence number is incremented, but the caller | |
819 must update the timestamp itself */ | |
2406
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lucabe
parents:
2274
diff
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|
820 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m) |
0 | 821 { |
294
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822 RTPDemuxContext *s = s1->priv_data; |
0 | 823 |
824 #ifdef DEBUG | |
825 printf("rtp_send_data size=%d\n", len); | |
826 #endif | |
827 | |
828 /* build the RTP header */ | |
829 put_byte(&s1->pb, (RTP_VERSION << 6)); | |
826
ee7ba4017c8d
MPEG4 streaming over RTP patch by (Luca Abeni: lucabe72, email it)
michael
parents:
820
diff
changeset
|
830 put_byte(&s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7)); |
0 | 831 put_be16(&s1->pb, s->seq); |
832 put_be32(&s1->pb, s->timestamp); | |
833 put_be32(&s1->pb, s->ssrc); | |
885 | 834 |
0 | 835 put_buffer(&s1->pb, buf1, len); |
836 put_flush_packet(&s1->pb); | |
885 | 837 |
0 | 838 s->seq++; |
839 s->octet_count += len; | |
840 s->packet_count++; | |
841 } | |
842 | |
843 /* send an integer number of samples and compute time stamp and fill | |
844 the rtp send buffer before sending. */ | |
845 static void rtp_send_samples(AVFormatContext *s1, | |
241 | 846 const uint8_t *buf1, int size, int sample_size) |
0 | 847 { |
294
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848 RTPDemuxContext *s = s1->priv_data; |
0 | 849 int len, max_packet_size, n; |
850 | |
851 max_packet_size = (s->max_payload_size / sample_size) * sample_size; | |
852 /* not needed, but who nows */ | |
853 if ((size % sample_size) != 0) | |
854 av_abort(); | |
2540
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Fix timestamps in RTP packets (now, MPEG1 video with B frames works correctly)
lucabe
parents:
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diff
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|
855 n = 0; |
0 | 856 while (size > 0) { |
2540
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Fix timestamps in RTP packets (now, MPEG1 video with B frames works correctly)
lucabe
parents:
2539
diff
changeset
|
857 s->buf_ptr = s->buf; |
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lucabe
parents:
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diff
changeset
|
858 len = FFMIN(max_packet_size, size); |
0 | 859 |
860 /* copy data */ | |
861 memcpy(s->buf_ptr, buf1, len); | |
862 s->buf_ptr += len; | |
863 buf1 += len; | |
864 size -= len; | |
2540
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Fix timestamps in RTP packets (now, MPEG1 video with B frames works correctly)
lucabe
parents:
2539
diff
changeset
|
865 s->timestamp = s->cur_timestamp + n / sample_size; |
ca3cba9c641f
Fix timestamps in RTP packets (now, MPEG1 video with B frames works correctly)
lucabe
parents:
2539
diff
changeset
|
866 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); |
ca3cba9c641f
Fix timestamps in RTP packets (now, MPEG1 video with B frames works correctly)
lucabe
parents:
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diff
changeset
|
867 n += (s->buf_ptr - s->buf); |
0 | 868 } |
885 | 869 } |
0 | 870 |
871 /* NOTE: we suppose that exactly one frame is given as argument here */ | |
872 /* XXX: test it */ | |
873 static void rtp_send_mpegaudio(AVFormatContext *s1, | |
241 | 874 const uint8_t *buf1, int size) |
0 | 875 { |
294
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|
876 RTPDemuxContext *s = s1->priv_data; |
0 | 877 int len, count, max_packet_size; |
878 | |
879 max_packet_size = s->max_payload_size; | |
880 | |
881 /* test if we must flush because not enough space */ | |
882 len = (s->buf_ptr - s->buf); | |
883 if ((len + size) > max_packet_size) { | |
884 if (len > 4) { | |
2406
18e94e5989d8
Move the RTP packetization code for MPEG12 video in its own file (rtp_mpv.c)
lucabe
parents:
2274
diff
changeset
|
885 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); |
0 | 886 s->buf_ptr = s->buf + 4; |
887 } | |
888 } | |
2540
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Fix timestamps in RTP packets (now, MPEG1 video with B frames works correctly)
lucabe
parents:
2539
diff
changeset
|
889 if (s->buf_ptr == s->buf + 4) { |
ca3cba9c641f
Fix timestamps in RTP packets (now, MPEG1 video with B frames works correctly)
lucabe
parents:
2539
diff
changeset
|
890 s->timestamp = s->cur_timestamp; |
ca3cba9c641f
Fix timestamps in RTP packets (now, MPEG1 video with B frames works correctly)
lucabe
parents:
2539
diff
changeset
|
891 } |
0 | 892 |
893 /* add the packet */ | |
894 if (size > max_packet_size) { | |
895 /* big packet: fragment */ | |
896 count = 0; | |
897 while (size > 0) { | |
898 len = max_packet_size - 4; | |
899 if (len > size) | |
900 len = size; | |
901 /* build fragmented packet */ | |
902 s->buf[0] = 0; | |
903 s->buf[1] = 0; | |
904 s->buf[2] = count >> 8; | |
905 s->buf[3] = count; | |
906 memcpy(s->buf + 4, buf1, len); | |
2406
18e94e5989d8
Move the RTP packetization code for MPEG12 video in its own file (rtp_mpv.c)
lucabe
parents:
2274
diff
changeset
|
907 ff_rtp_send_data(s1, s->buf, len + 4, 0); |
0 | 908 size -= len; |
909 buf1 += len; | |
910 count += len; | |
911 } | |
912 } else { | |
913 if (s->buf_ptr == s->buf + 4) { | |
914 /* no fragmentation possible */ | |
915 s->buf[0] = 0; | |
916 s->buf[1] = 0; | |
917 s->buf[2] = 0; | |
918 s->buf[3] = 0; | |
919 } | |
920 memcpy(s->buf_ptr, buf1, size); | |
921 s->buf_ptr += size; | |
922 } | |
923 } | |
924 | |
925 static void rtp_send_raw(AVFormatContext *s1, | |
241 | 926 const uint8_t *buf1, int size) |
0 | 927 { |
294
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|
928 RTPDemuxContext *s = s1->priv_data; |
0 | 929 int len, max_packet_size; |
930 | |
931 max_packet_size = s->max_payload_size; | |
932 | |
933 while (size > 0) { | |
934 len = max_packet_size; | |
935 if (len > size) | |
936 len = size; | |
937 | |
2540
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Fix timestamps in RTP packets (now, MPEG1 video with B frames works correctly)
lucabe
parents:
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diff
changeset
|
938 s->timestamp = s->cur_timestamp; |
2406
18e94e5989d8
Move the RTP packetization code for MPEG12 video in its own file (rtp_mpv.c)
lucabe
parents:
2274
diff
changeset
|
939 ff_rtp_send_data(s1, buf1, len, (len == size)); |
0 | 940 |
941 buf1 += len; | |
942 size -= len; | |
943 } | |
944 } | |
945 | |
294
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|
946 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */ |
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|
947 static void rtp_send_mpegts_raw(AVFormatContext *s1, |
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|
948 const uint8_t *buf1, int size) |
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949 { |
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|
950 RTPDemuxContext *s = s1->priv_data; |
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951 int len, out_len; |
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|
952 |
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|
953 while (size >= TS_PACKET_SIZE) { |
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|
954 len = s->max_payload_size - (s->buf_ptr - s->buf); |
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955 if (len > size) |
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|
956 len = size; |
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|
957 memcpy(s->buf_ptr, buf1, len); |
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|
958 buf1 += len; |
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959 size -= len; |
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960 s->buf_ptr += len; |
885 | 961 |
294
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|
962 out_len = s->buf_ptr - s->buf; |
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963 if (out_len >= s->max_payload_size) { |
2406
18e94e5989d8
Move the RTP packetization code for MPEG12 video in its own file (rtp_mpv.c)
lucabe
parents:
2274
diff
changeset
|
964 ff_rtp_send_data(s1, s->buf, out_len, 0); |
294
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965 s->buf_ptr = s->buf; |
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966 } |
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967 } |
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|
968 } |
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969 |
0 | 970 /* write an RTP packet. 'buf1' must contain a single specific frame. */ |
468 | 971 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) |
0 | 972 { |
294
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241
diff
changeset
|
973 RTPDemuxContext *s = s1->priv_data; |
0 | 974 AVStream *st = s1->streams[0]; |
975 int rtcp_bytes; | |
468 | 976 int size= pkt->size; |
977 uint8_t *buf1= pkt->data; | |
885 | 978 |
0 | 979 #ifdef DEBUG |
468 | 980 printf("%d: write len=%d\n", pkt->stream_index, size); |
0 | 981 #endif |
982 | |
983 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */ | |
885 | 984 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / |
0 | 985 RTCP_TX_RATIO_DEN; |
986 if (s->first_packet || rtcp_bytes >= 28) { | |
2539
ba933dfa4833
Properly set RTP and NTP timestamps in RTCP SR packets
lucabe
parents:
2406
diff
changeset
|
987 rtcp_send_sr(s1, av_gettime()); |
0 | 988 s->last_octet_count = s->octet_count; |
989 s->first_packet = 0; | |
990 } | |
2540
ca3cba9c641f
Fix timestamps in RTP packets (now, MPEG1 video with B frames works correctly)
lucabe
parents:
2539
diff
changeset
|
991 s->cur_timestamp = s->base_timestamp + pkt->pts; |
0 | 992 |
820
feca73904e67
changing AVCodecContext codec -> *codec in AVStream so additions to AVCodecContext dont randomize AVStream and break binary compatibility
michael
parents:
777
diff
changeset
|
993 switch(st->codec->codec_id) { |
0 | 994 case CODEC_ID_PCM_MULAW: |
995 case CODEC_ID_PCM_ALAW: | |
996 case CODEC_ID_PCM_U8: | |
997 case CODEC_ID_PCM_S8: | |
820
feca73904e67
changing AVCodecContext codec -> *codec in AVStream so additions to AVCodecContext dont randomize AVStream and break binary compatibility
michael
parents:
777
diff
changeset
|
998 rtp_send_samples(s1, buf1, size, 1 * st->codec->channels); |
0 | 999 break; |
1000 case CODEC_ID_PCM_U16BE: | |
1001 case CODEC_ID_PCM_U16LE: | |
1002 case CODEC_ID_PCM_S16BE: | |
1003 case CODEC_ID_PCM_S16LE: | |
820
feca73904e67
changing AVCodecContext codec -> *codec in AVStream so additions to AVCodecContext dont randomize AVStream and break binary compatibility
michael
parents:
777
diff
changeset
|
1004 rtp_send_samples(s1, buf1, size, 2 * st->codec->channels); |
0 | 1005 break; |
1006 case CODEC_ID_MP2: | |
232 | 1007 case CODEC_ID_MP3: |
0 | 1008 rtp_send_mpegaudio(s1, buf1, size); |
1009 break; | |
1010 case CODEC_ID_MPEG1VIDEO: | |
2406
18e94e5989d8
Move the RTP packetization code for MPEG12 video in its own file (rtp_mpv.c)
lucabe
parents:
2274
diff
changeset
|
1011 ff_rtp_send_mpegvideo(s1, buf1, size); |
0 | 1012 break; |
2550
e9c34ec665c6
Support for AAC streaming over RTP. Fragmentation is not implemented yet
lucabe
parents:
2540
diff
changeset
|
1013 case CODEC_ID_AAC: |
e9c34ec665c6
Support for AAC streaming over RTP. Fragmentation is not implemented yet
lucabe
parents:
2540
diff
changeset
|
1014 ff_rtp_send_aac(s1, buf1, size); |
e9c34ec665c6
Support for AAC streaming over RTP. Fragmentation is not implemented yet
lucabe
parents:
2540
diff
changeset
|
1015 break; |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
1016 case CODEC_ID_MPEG2TS: |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
1017 rtp_send_mpegts_raw(s1, buf1, size); |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
1018 break; |
0 | 1019 default: |
1020 /* better than nothing : send the codec raw data */ | |
1021 rtp_send_raw(s1, buf1, size); | |
1022 break; | |
1023 } | |
1024 return 0; | |
1025 } | |
1026 | |
1167 | 1027 AVOutputFormat rtp_muxer = { |
0 | 1028 "rtp", |
1029 "RTP output format", | |
1030 NULL, | |
1031 NULL, | |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
1032 sizeof(RTPDemuxContext), |
0 | 1033 CODEC_ID_PCM_MULAW, |
1034 CODEC_ID_NONE, | |
1035 rtp_write_header, | |
1036 rtp_write_packet, | |
1037 }; |