Mercurial > libavformat.hg
annotate rtp.c @ 2772:b43dc2901f83 libavformat
Make av_read_frame with rtsp client return EINTR on interrupt
patch from elupusateccedotse
author | lu_zero |
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date | Wed, 21 Nov 2007 10:26:11 +0000 |
parents | d52c718e83f9 |
children | dd1fbe36d103 |
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 | |
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35 #define RTCP_SR_SIZE 28 |
0 | 36 |
37 /* TODO: - add RTCP statistics reporting (should be optional). | |
38 | |
39 - add support for h263/mpeg4 packetized output : IDEA: send a | |
40 buffer to 'rtp_write_packet' contains all the packets for ONE | |
41 frame. Each packet should have a four byte header containing | |
42 the length in big endian format (same trick as | |
885 | 43 'url_open_dyn_packet_buf') |
0 | 44 */ |
45 | |
774 | 46 /* from http://www.iana.org/assignments/rtp-parameters last updated 05 January 2005 */ |
47 AVRtpPayloadType_t AVRtpPayloadTypes[]= | |
48 { | |
49 {0, "PCMU", CODEC_TYPE_AUDIO, CODEC_ID_PCM_MULAW, 8000, 1}, | |
50 {1, "Reserved", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
51 {2, "Reserved", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
52 {3, "GSM", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
53 {4, "G723", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
54 {5, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
55 {6, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 16000, 1}, | |
56 {7, "LPC", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
57 {8, "PCMA", CODEC_TYPE_AUDIO, CODEC_ID_PCM_ALAW, 8000, 1}, | |
58 {9, "G722", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
59 {10, "L16", CODEC_TYPE_AUDIO, CODEC_ID_PCM_S16BE, 44100, 2}, | |
60 {11, "L16", CODEC_TYPE_AUDIO, CODEC_ID_PCM_S16BE, 44100, 1}, | |
1259 | 61 {12, "QCELP", CODEC_TYPE_AUDIO, CODEC_ID_QCELP, 8000, 1}, |
774 | 62 {13, "CN", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, |
63 {14, "MPA", CODEC_TYPE_AUDIO, CODEC_ID_MP2, 90000, -1}, | |
64 {15, "G728", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
65 {16, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 11025, 1}, | |
66 {17, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 22050, 1}, | |
67 {18, "G729", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, | |
68 {19, "reserved", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, | |
69 {20, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, | |
70 {21, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, | |
71 {22, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, | |
72 {23, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, | |
73 {24, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1}, | |
74 {25, "CelB", CODEC_TYPE_VIDEO, CODEC_ID_NONE, 90000, -1}, | |
75 {26, "JPEG", CODEC_TYPE_VIDEO, CODEC_ID_MJPEG, 90000, -1}, | |
76 {27, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1}, | |
77 {28, "nv", CODEC_TYPE_VIDEO, CODEC_ID_NONE, 90000, -1}, | |
78 {29, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1}, | |
79 {30, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1}, | |
80 {31, "H261", CODEC_TYPE_VIDEO, CODEC_ID_H261, 90000, -1}, | |
81 {32, "MPV", CODEC_TYPE_VIDEO, CODEC_ID_MPEG1VIDEO, 90000, -1}, | |
2760 | 82 {32, "MPV", CODEC_TYPE_VIDEO, CODEC_ID_MPEG2VIDEO, 90000, -1}, |
774 | 83 {33, "MP2T", CODEC_TYPE_DATA, CODEC_ID_MPEG2TS, 90000, -1}, |
84 {34, "H263", CODEC_TYPE_VIDEO, CODEC_ID_H263, 90000, -1}, | |
85 {35, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
86 {36, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
87 {37, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
88 {38, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
89 {39, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
90 {40, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
91 {41, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
92 {42, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
93 {43, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
94 {44, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
95 {45, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
96 {46, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
97 {47, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
98 {48, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
99 {49, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
100 {50, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
101 {51, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
102 {52, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
103 {53, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
104 {54, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
105 {55, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
106 {56, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
107 {57, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
108 {58, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
109 {59, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
110 {60, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
111 {61, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
112 {62, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
113 {63, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
114 {64, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
115 {65, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
116 {66, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
117 {67, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
118 {68, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
119 {69, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
120 {70, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
121 {71, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
122 {72, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
123 {73, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
124 {74, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
125 {75, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
126 {76, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
127 {77, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
128 {78, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
129 {79, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
130 {80, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
131 {81, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
132 {82, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
133 {83, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
134 {84, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
135 {85, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
136 {86, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
137 {87, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
138 {88, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
139 {89, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
140 {90, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
141 {91, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
142 {92, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
143 {93, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
144 {94, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
145 {95, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
146 {96, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
147 {97, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
148 {98, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
149 {99, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
150 {100, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
151 {101, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
152 {102, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
153 {103, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
154 {104, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
155 {105, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
156 {106, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
157 {107, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
158 {108, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
159 {109, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
160 {110, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
161 {111, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
162 {112, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
163 {113, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
164 {114, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
165 {115, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
166 {116, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
167 {117, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
168 {118, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
169 {119, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
170 {120, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
171 {121, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
172 {122, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
173 {123, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
174 {124, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
175 {125, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
176 {126, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
177 {127, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, | |
178 {-1, "", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1} | |
179 }; | |
0 | 180 |
1419 | 181 /* statistics functions */ |
182 RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL; | |
183 | |
184 static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4}; | |
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185 static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC}; |
0 | 186 |
1419 | 187 static void register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler) |
188 { | |
189 handler->next= RTPFirstDynamicPayloadHandler; | |
190 RTPFirstDynamicPayloadHandler= handler; | |
191 } | |
885 | 192 |
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193 void av_register_rtp_dynamic_payload_handlers(void) |
1419 | 194 { |
195 register_dynamic_payload_handler(&mp4v_es_handler); | |
196 register_dynamic_payload_handler(&mpeg4_generic_handler); | |
197 register_dynamic_payload_handler(&ff_h264_dynamic_handler); | |
198 } | |
0 | 199 |
200 int rtp_get_codec_info(AVCodecContext *codec, int payload_type) | |
201 { | |
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202 int i = 0; |
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203 |
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204 for (i = 0; AVRtpPayloadTypes[i].pt >= 0; i++) |
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205 if (AVRtpPayloadTypes[i].pt == payload_type) { |
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206 if (AVRtpPayloadTypes[i].codec_id != CODEC_ID_NONE) { |
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207 codec->codec_type = AVRtpPayloadTypes[i].codec_type; |
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208 codec->codec_id = AVRtpPayloadTypes[i].codec_id; |
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209 if (AVRtpPayloadTypes[i].audio_channels > 0) |
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210 codec->channels = AVRtpPayloadTypes[i].audio_channels; |
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211 if (AVRtpPayloadTypes[i].clock_rate > 0) |
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212 codec->sample_rate = AVRtpPayloadTypes[i].clock_rate; |
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213 return 0; |
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214 } |
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215 } |
774 | 216 return -1; |
0 | 217 } |
218 | |
219 int rtp_get_payload_type(AVCodecContext *codec) | |
220 { | |
774 | 221 int i, payload_type; |
0 | 222 |
223 /* compute the payload type */ | |
774 | 224 for (payload_type = -1, i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i) |
225 if (AVRtpPayloadTypes[i].codec_id == codec->codec_id) { | |
226 if (codec->codec_id == CODEC_ID_PCM_S16BE) | |
227 if (codec->channels != AVRtpPayloadTypes[i].audio_channels) | |
228 continue; | |
229 payload_type = AVRtpPayloadTypes[i].pt; | |
0 | 230 } |
231 return payload_type; | |
232 } | |
233 | |
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234 const char *ff_rtp_enc_name(int payload_type) |
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235 { |
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236 int i; |
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237 |
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238 for (i = 0; AVRtpPayloadTypes[i].pt >= 0; i++) |
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239 if (AVRtpPayloadTypes[i].pt == payload_type) { |
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240 return AVRtpPayloadTypes[i].enc_name; |
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241 } |
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242 |
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243 return ""; |
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244 } |
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245 |
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246 enum CodecID ff_rtp_codec_id(const char *buf, enum CodecType codec_type) |
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247 { |
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248 int i; |
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249 |
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250 for (i = 0; AVRtpPayloadTypes[i].pt >= 0; i++) |
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251 if (!strcmp(buf, AVRtpPayloadTypes[i].enc_name) && (codec_type == AVRtpPayloadTypes[i].codec_type)){ |
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252 return AVRtpPayloadTypes[i].codec_id; |
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253 } |
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254 |
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255 return CODEC_ID_NONE; |
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256 } |
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257 |
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258 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len) |
0 | 259 { |
260 if (buf[1] != 200) | |
261 return -1; | |
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262 s->last_rtcp_ntp_time = AV_RB64(buf + 8); |
173 | 263 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) |
264 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time; | |
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265 s->last_rtcp_timestamp = AV_RB32(buf + 16); |
0 | 266 return 0; |
267 } | |
268 | |
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269 #define RTP_SEQ_MOD (1<<16) |
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270 |
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271 /** |
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272 * called on parse open packet |
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273 */ |
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274 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet. |
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275 { |
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276 memset(s, 0, sizeof(RTPStatistics)); |
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277 s->max_seq= base_sequence; |
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278 s->probation= 1; |
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279 } |
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280 |
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281 /** |
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282 * called whenever there is a large jump in sequence numbers, or when they get out of probation... |
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283 */ |
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284 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq) |
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285 { |
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286 s->max_seq= seq; |
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287 s->cycles= 0; |
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288 s->base_seq= seq -1; |
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289 s->bad_seq= RTP_SEQ_MOD + 1; |
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290 s->received= 0; |
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291 s->expected_prior= 0; |
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292 s->received_prior= 0; |
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293 s->jitter= 0; |
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294 s->transit= 0; |
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295 } |
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296 |
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297 /** |
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298 * returns 1 if we should handle this packet. |
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299 */ |
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300 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq) |
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301 { |
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302 uint16_t udelta= seq - s->max_seq; |
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303 const int MAX_DROPOUT= 3000; |
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304 const int MAX_MISORDER = 100; |
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305 const int MIN_SEQUENTIAL = 2; |
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306 |
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307 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */ |
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308 if(s->probation) |
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309 { |
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310 if(seq==s->max_seq + 1) { |
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311 s->probation--; |
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312 s->max_seq= seq; |
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313 if(s->probation==0) { |
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314 rtp_init_sequence(s, seq); |
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315 s->received++; |
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316 return 1; |
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317 } |
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318 } else { |
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319 s->probation= MIN_SEQUENTIAL - 1; |
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320 s->max_seq = seq; |
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321 } |
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322 } else if (udelta < MAX_DROPOUT) { |
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323 // in order, with permissible gap |
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324 if(seq < s->max_seq) { |
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325 //sequence number wrapped; count antother 64k cycles |
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326 s->cycles += RTP_SEQ_MOD; |
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327 } |
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328 s->max_seq= seq; |
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329 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) { |
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330 // sequence made a large jump... |
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331 if(seq==s->bad_seq) { |
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332 // two sequential packets-- assume that the other side restarted without telling us; just resync. |
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333 rtp_init_sequence(s, seq); |
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334 } else { |
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335 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1); |
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336 return 0; |
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337 } |
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338 } else { |
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339 // duplicate or reordered packet... |
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340 } |
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341 s->received++; |
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342 return 1; |
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343 } |
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344 |
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345 #if 0 |
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346 /** |
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347 * This function is currently unused; without a valid local ntp time, I don't see how we could calculate the |
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348 * difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values |
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349 * never change. I left this in in case someone else can see a way. (rdm) |
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350 */ |
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351 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp) |
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352 { |
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353 uint32_t transit= arrival_timestamp - sent_timestamp; |
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354 int d; |
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355 s->transit= transit; |
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356 d= FFABS(transit - s->transit); |
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357 s->jitter += d - ((s->jitter + 8)>>4); |
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358 } |
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359 #endif |
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360 |
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361 int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count) |
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362 { |
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363 ByteIOContext *pb; |
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364 uint8_t *buf; |
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365 int len; |
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366 int rtcp_bytes; |
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367 RTPStatistics *stats= &s->statistics; |
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368 uint32_t lost; |
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369 uint32_t extended_max; |
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370 uint32_t expected_interval; |
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371 uint32_t received_interval; |
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372 uint32_t lost_interval; |
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373 uint32_t expected; |
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374 uint32_t fraction; |
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375 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time? |
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376 |
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377 if (!s->rtp_ctx || (count < 1)) |
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378 return -1; |
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379 |
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380 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */ |
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381 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */ |
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382 s->octet_count += count; |
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383 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / |
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384 RTCP_TX_RATIO_DEN; |
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385 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !? |
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386 if (rtcp_bytes < 28) |
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387 return -1; |
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388 s->last_octet_count = s->octet_count; |
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389 |
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390 if (url_open_dyn_buf(&pb) < 0) |
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391 return -1; |
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392 |
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393 // Receiver Report |
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394 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */ |
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395 put_byte(pb, 201); |
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396 put_be16(pb, 7); /* length in words - 1 */ |
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397 put_be32(pb, s->ssrc); // our own SSRC |
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398 put_be32(pb, s->ssrc); // XXX: should be the server's here! |
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399 // some placeholders we should really fill... |
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400 // RFC 1889/p64 |
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401 extended_max= stats->cycles + stats->max_seq; |
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402 expected= extended_max - stats->base_seq + 1; |
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403 lost= expected - stats->received; |
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404 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits... |
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405 expected_interval= expected - stats->expected_prior; |
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406 stats->expected_prior= expected; |
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407 received_interval= stats->received - stats->received_prior; |
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408 stats->received_prior= stats->received; |
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409 lost_interval= expected_interval - received_interval; |
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410 if (expected_interval==0 || lost_interval<=0) fraction= 0; |
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411 else fraction = (lost_interval<<8)/expected_interval; |
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412 |
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413 fraction= (fraction<<24) | lost; |
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414 |
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415 put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */ |
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416 put_be32(pb, extended_max); /* max sequence received */ |
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417 put_be32(pb, stats->jitter>>4); /* jitter */ |
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418 |
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419 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE) |
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420 { |
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421 put_be32(pb, 0); /* last SR timestamp */ |
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422 put_be32(pb, 0); /* delay since last SR */ |
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423 } else { |
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424 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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425 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time; |
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426 |
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427 put_be32(pb, middle_32_bits); /* last SR timestamp */ |
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428 put_be32(pb, delay_since_last); /* delay since last SR */ |
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429 } |
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430 |
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431 // CNAME |
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432 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */ |
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433 put_byte(pb, 202); |
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434 len = strlen(s->hostname); |
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435 put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */ |
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436 put_be32(pb, s->ssrc); |
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437 put_byte(pb, 0x01); |
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438 put_byte(pb, len); |
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439 put_buffer(pb, s->hostname, len); |
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440 // padding |
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441 for (len = (6 + len) % 4; len % 4; len++) { |
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442 put_byte(pb, 0); |
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443 } |
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444 |
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445 put_flush_packet(pb); |
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446 len = url_close_dyn_buf(pb, &buf); |
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447 if ((len > 0) && buf) { |
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448 int result; |
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449 #if defined(DEBUG) |
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450 printf("sending %d bytes of RR\n", len); |
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451 #endif |
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452 result= url_write(s->rtp_ctx, buf, len); |
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453 #if defined(DEBUG) |
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454 printf("result from url_write: %d\n", result); |
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455 #endif |
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456 av_free(buf); |
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457 } |
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458 return 0; |
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459 } |
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460 |
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461 /** |
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462 * open a new RTP parse context for stream 'st'. 'st' can be NULL for |
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463 * MPEG2TS streams to indicate that they should be demuxed inside the |
885 | 464 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned) |
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465 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system. |
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466 */ |
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467 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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468 { |
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469 RTPDemuxContext *s; |
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470 |
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471 s = av_mallocz(sizeof(RTPDemuxContext)); |
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472 if (!s) |
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473 return NULL; |
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474 s->payload_type = payload_type; |
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475 s->last_rtcp_ntp_time = AV_NOPTS_VALUE; |
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476 s->first_rtcp_ntp_time = AV_NOPTS_VALUE; |
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477 s->ic = s1; |
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478 s->st = st; |
774 | 479 s->rtp_payload_data = rtp_payload_data; |
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480 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp? |
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481 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) { |
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482 s->ts = mpegts_parse_open(s->ic); |
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483 if (s->ts == NULL) { |
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484 av_free(s); |
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485 return NULL; |
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486 } |
305 | 487 } else { |
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488 switch(st->codec->codec_id) { |
305 | 489 case CODEC_ID_MPEG1VIDEO: |
490 case CODEC_ID_MPEG2VIDEO: | |
491 case CODEC_ID_MP2: | |
492 case CODEC_ID_MP3: | |
493 case CODEC_ID_MPEG4: | |
1419 | 494 case CODEC_ID_H264: |
2023 | 495 st->need_parsing = AVSTREAM_PARSE_FULL; |
305 | 496 break; |
497 default: | |
498 break; | |
499 } | |
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500 } |
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501 // needed to send back RTCP RR in RTSP sessions |
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502 s->rtp_ctx = rtpc; |
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503 gethostname(s->hostname, sizeof(s->hostname)); |
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504 return s; |
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505 } |
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506 |
774 | 507 static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf) |
508 { | |
509 int au_headers_length, au_header_size, i; | |
510 GetBitContext getbitcontext; | |
511 rtp_payload_data_t *infos; | |
512 | |
513 infos = s->rtp_payload_data; | |
514 | |
515 if (infos == NULL) | |
516 return -1; | |
517 | |
518 /* decode the first 2 bytes where are stored the AUHeader sections | |
519 length in bits */ | |
1673 | 520 au_headers_length = AV_RB16(buf); |
774 | 521 |
522 if (au_headers_length > RTP_MAX_PACKET_LENGTH) | |
523 return -1; | |
524 | |
525 infos->au_headers_length_bytes = (au_headers_length + 7) / 8; | |
526 | |
527 /* skip AU headers length section (2 bytes) */ | |
528 buf += 2; | |
529 | |
530 init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8); | |
531 | |
532 /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */ | |
533 au_header_size = infos->sizelength + infos->indexlength; | |
534 if (au_header_size <= 0 || (au_headers_length % au_header_size != 0)) | |
535 return -1; | |
536 | |
537 infos->nb_au_headers = au_headers_length / au_header_size; | |
538 infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers); | |
539 | |
540 /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving) | |
2164 | 541 In my test, the FAAD decoder does not behave correctly when sending each AU one by one |
774 | 542 but does when sending the whole as one big packet... */ |
543 infos->au_headers[0].size = 0; | |
544 infos->au_headers[0].index = 0; | |
545 for (i = 0; i < infos->nb_au_headers; ++i) { | |
546 infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength); | |
547 infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength); | |
548 } | |
549 | |
550 infos->nb_au_headers = 1; | |
551 | |
552 return 0; | |
553 } | |
554 | |
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555 /** |
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556 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc. |
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557 */ |
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558 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp) |
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559 { |
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560 switch(s->st->codec->codec_id) { |
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gpoirier
parents:
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|
561 case CODEC_ID_MP2: |
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gpoirier
parents:
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changeset
|
562 case CODEC_ID_MPEG1VIDEO: |
2760 | 563 case CODEC_ID_MPEG2VIDEO: |
1431
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parents:
1427
diff
changeset
|
564 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) { |
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gpoirier
parents:
1427
diff
changeset
|
565 int64_t addend; |
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gpoirier
parents:
1427
diff
changeset
|
566 |
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parents:
1427
diff
changeset
|
567 int delta_timestamp; |
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changeset
|
568 /* XXX: is it really necessary to unify the timestamp base ? */ |
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parents:
1427
diff
changeset
|
569 /* compute pts from timestamp with received ntp_time */ |
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gpoirier
parents:
1427
diff
changeset
|
570 delta_timestamp = timestamp - s->last_rtcp_timestamp; |
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gpoirier
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571 /* convert to 90 kHz without overflow */ |
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572 addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14; |
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573 addend = (addend * 5625) >> 14; |
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|
574 pkt->pts = addend + delta_timestamp; |
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changeset
|
575 } |
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changeset
|
576 break; |
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
|
577 case CODEC_ID_AAC: |
1431
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parents:
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diff
changeset
|
578 case CODEC_ID_H264: |
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parents:
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changeset
|
579 case CODEC_ID_MPEG4: |
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changeset
|
580 pkt->pts = timestamp; |
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parents:
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diff
changeset
|
581 break; |
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gpoirier
parents:
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diff
changeset
|
582 default: |
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parents:
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diff
changeset
|
583 /* no timestamp info yet */ |
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parents:
1427
diff
changeset
|
584 break; |
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parents:
1427
diff
changeset
|
585 } |
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diff
changeset
|
586 pkt->stream_index = s->st->index; |
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diff
changeset
|
587 } |
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diff
changeset
|
588 |
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changeset
|
589 /** |
885 | 590 * Parse an RTP or RTCP packet directly sent as a buffer. |
294
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|
591 * @param s RTP parse context. |
0 | 592 * @param pkt returned packet |
294
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|
593 * @param buf input buffer or NULL to read the next packets |
0 | 594 * @param len buffer len |
885 | 595 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow |
294
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|
596 * (use buf as NULL to read the next). -1 if no packet (error or no more packet). |
0 | 597 */ |
885 | 598 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, |
294
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241
diff
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|
599 const uint8_t *buf, int len) |
0 | 600 { |
601 unsigned int ssrc, h; | |
1431
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gpoirier
parents:
1427
diff
changeset
|
602 int payload_type, seq, ret; |
0 | 603 AVStream *st; |
65 | 604 uint32_t timestamp; |
1431
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gpoirier
parents:
1427
diff
changeset
|
605 int rv= 0; |
885 | 606 |
294
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diff
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|
607 if (!buf) { |
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diff
changeset
|
608 /* return the next packets, if any */ |
1419 | 609 if(s->st && s->parse_packet) { |
1431
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gpoirier
parents:
1427
diff
changeset
|
610 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
|
611 rv= s->parse_packet(s, pkt, ×tamp, NULL, 0); |
2d8a17631520
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gpoirier
parents:
1427
diff
changeset
|
612 finalize_packet(s, pkt, timestamp); |
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fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
613 return rv; |
1419 | 614 } else { |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
615 // 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
|
616 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
|
617 return -1; |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
618 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
|
619 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
|
620 if (ret < 0) |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
621 return -1; |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
622 s->read_buf_index += ret; |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
623 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
|
624 return 1; |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
625 else |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
626 return 0; |
1419 | 627 } |
294
6091b76cfc2a
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parents:
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diff
changeset
|
628 } |
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parents:
241
diff
changeset
|
629 |
0 | 630 if (len < 12) |
631 return -1; | |
632 | |
633 if ((buf[0] & 0xc0) != (RTP_VERSION << 6)) | |
634 return -1; | |
635 if (buf[1] >= 200 && buf[1] <= 204) { | |
294
6091b76cfc2a
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bellard
parents:
241
diff
changeset
|
636 rtcp_parse_packet(s, buf, len); |
0 | 637 return -1; |
638 } | |
639 payload_type = buf[1] & 0x7f; | |
2222 | 640 seq = AV_RB16(buf + 2); |
2167
c4244900f986
remove decode_be32/64 and replace them by AV_RB32/64
benoit
parents:
2164
diff
changeset
|
641 timestamp = AV_RB32(buf + 4); |
c4244900f986
remove decode_be32/64 and replace them by AV_RB32/64
benoit
parents:
2164
diff
changeset
|
642 ssrc = AV_RB32(buf + 8); |
1425
00d9393a126f
make ffmpeg able to send back a RTCP receiver report.
gpoirier
parents:
1424
diff
changeset
|
643 /* store the ssrc in the RTPDemuxContext */ |
00d9393a126f
make ffmpeg able to send back a RTCP receiver report.
gpoirier
parents:
1424
diff
changeset
|
644 s->ssrc = ssrc; |
885 | 645 |
0 | 646 /* NOTE: we can handle only one payload type */ |
647 if (s->payload_type != payload_type) | |
648 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
|
649 |
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
|
650 st = s->st; |
1445
db97355877b1
add valid statistics for the RTCP receiver report.
gpoirier
parents:
1443
diff
changeset
|
651 // 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
|
652 if(!rtp_valid_packet_in_sequence(&s->statistics, seq)) |
db97355877b1
add valid statistics for the RTCP receiver report.
gpoirier
parents:
1443
diff
changeset
|
653 { |
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
|
654 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n", |
0 | 655 payload_type, seq, ((s->seq + 1) & 0xffff)); |
1445
db97355877b1
add valid statistics for the RTCP receiver report.
gpoirier
parents:
1443
diff
changeset
|
656 return -1; |
0 | 657 } |
1445
db97355877b1
add valid statistics for the RTCP receiver report.
gpoirier
parents:
1443
diff
changeset
|
658 |
0 | 659 s->seq = seq; |
660 len -= 12; | |
661 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
|
662 |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
663 if (!st) { |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
664 /* specific MPEG2TS demux support */ |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
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parents:
241
diff
changeset
|
665 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
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parents:
241
diff
changeset
|
666 if (ret < 0) |
0 | 667 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
|
668 if (ret < len) { |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
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parents:
241
diff
changeset
|
669 s->read_buf_size = len - ret; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
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diff
changeset
|
670 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
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parents:
241
diff
changeset
|
671 s->read_buf_index = 0; |
6091b76cfc2a
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diff
changeset
|
672 return 1; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
673 } |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
674 } else { |
1419 | 675 // 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
|
676 switch(st->codec->codec_id) { |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
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diff
changeset
|
677 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
|
678 /* better than nothing: skip mpeg audio RTP header */ |
0 | 679 if (len <= 4) |
680 return -1; | |
2167
c4244900f986
remove decode_be32/64 and replace them by AV_RB32/64
benoit
parents:
2164
diff
changeset
|
681 h = AV_RB32(buf); |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
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diff
changeset
|
682 len -= 4; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
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diff
changeset
|
683 buf += 4; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
684 av_new_packet(pkt, len); |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
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241
diff
changeset
|
685 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
|
686 break; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
687 case CODEC_ID_MPEG1VIDEO: |
2760 | 688 case CODEC_ID_MPEG2VIDEO: |
305 | 689 /* 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
|
690 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
|
691 return -1; |
2167
c4244900f986
remove decode_be32/64 and replace them by AV_RB32/64
benoit
parents:
2164
diff
changeset
|
692 h = AV_RB32(buf); |
0 | 693 buf += 4; |
694 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
|
695 if (h & (1 << 26)) { |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
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241
diff
changeset
|
696 /* mpeg2 */ |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
697 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
|
698 return -1; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
699 buf += 4; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
700 len -= 4; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
701 } |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
702 av_new_packet(pkt, len); |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
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parents:
241
diff
changeset
|
703 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
|
704 break; |
1427
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
705 // 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
|
706 // timestamps. |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
707 // 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
|
708 case CODEC_ID_AAC: |
1427
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
709 if (rtp_parse_mp4_au(s, buf)) |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
710 return -1; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
711 { |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
712 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
|
713 if (infos == NULL) |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
714 return -1; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
715 buf += infos->au_headers_length_bytes + 2; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
716 len -= infos->au_headers_length_bytes + 2; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
717 |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
718 /* 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
|
719 one au_header */ |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
720 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
|
721 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
|
722 buf += infos->au_headers[0].size; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
723 len -= infos->au_headers[0].size; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
724 } |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
725 s->read_buf_size = len; |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
726 rv= 0; |
1427
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
727 break; |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
728 default: |
1419 | 729 if(s->parse_packet) { |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
730 rv= s->parse_packet(s, pkt, ×tamp, buf, len); |
1419 | 731 } else { |
1424
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
732 av_new_packet(pkt, len); |
1c39ce5c6a5d
indentation fix, patch by Ryan Martell % rdm4 A martellventures P com %
gpoirier
parents:
1419
diff
changeset
|
733 memcpy(pkt->data, buf, len); |
1419 | 734 } |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
735 break; |
0 | 736 } |
885 | 737 |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
738 // now perform timestamp things.... |
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
739 finalize_packet(s, pkt, timestamp); |
0 | 740 } |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
741 return rv; |
0 | 742 } |
743 | |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
744 void rtp_parse_close(RTPDemuxContext *s) |
0 | 745 { |
1419 | 746 // TODO: fold this into the protocol specific data fields. |
2759
b252e318023a
Remove the "AVRtpPayloadTypes[i].pt == i" assumption from RTP and RTSP
lucabe
parents:
2706
diff
changeset
|
747 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) { |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
748 mpegts_parse_close(s->ts); |
0 | 749 } |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
750 av_free(s); |
0 | 751 } |
752 | |
753 /* rtp output */ | |
754 | |
755 static int rtp_write_header(AVFormatContext *s1) | |
756 { | |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
757 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
|
758 int payload_type, max_packet_size, n; |
0 | 759 AVStream *st; |
760 | |
761 if (s1->nb_streams != 1) | |
762 return -1; | |
763 st = s1->streams[0]; | |
764 | |
820
feca73904e67
changing AVCodecContext codec -> *codec in AVStream so additions to AVCodecContext dont randomize AVStream and break binary compatibility
michael
parents:
777
diff
changeset
|
765 payload_type = rtp_get_payload_type(st->codec); |
0 | 766 if (payload_type < 0) |
767 payload_type = RTP_PT_PRIVATE; /* private payload type */ | |
768 s->payload_type = payload_type; | |
769 | |
1099 | 770 // 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
|
771 s->base_timestamp = 0; /* FIXME: was random(), what should this be? */ |
0 | 772 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
|
773 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
|
774 s->ssrc = 0; /* FIXME: was random(), what should this be? */ |
0 | 775 s->first_packet = 1; |
2539
ba933dfa4833
Properly set RTP and NTP timestamps in RTCP SR packets
lucabe
parents:
2406
diff
changeset
|
776 s->first_rtcp_ntp_time = AV_NOPTS_VALUE; |
0 | 777 |
2771
d52c718e83f9
Use dynamically allocated ByteIOContext in AVFormatContext
andoma
parents:
2760
diff
changeset
|
778 max_packet_size = url_fget_max_packet_size(s1->pb); |
0 | 779 if (max_packet_size <= 12) |
2274
b21c2af60bc9
Replace all occurrences of AVERROR_IO with AVERROR(EIO).
takis
parents:
2222
diff
changeset
|
780 return AVERROR(EIO); |
0 | 781 s->max_payload_size = max_packet_size - 12; |
782 | |
2587
d751acab2622
Allow to set the maximum number of frames per RTP packet (and add support for
lucabe
parents:
2569
diff
changeset
|
783 s->max_frames_per_packet = 0; |
d751acab2622
Allow to set the maximum number of frames per RTP packet (and add support for
lucabe
parents:
2569
diff
changeset
|
784 if (s1->max_delay) { |
d751acab2622
Allow to set the maximum number of frames per RTP packet (and add support for
lucabe
parents:
2569
diff
changeset
|
785 if (st->codec->codec_type == CODEC_TYPE_AUDIO) { |
d751acab2622
Allow to set the maximum number of frames per RTP packet (and add support for
lucabe
parents:
2569
diff
changeset
|
786 if (st->codec->frame_size == 0) { |
d751acab2622
Allow to set the maximum number of frames per RTP packet (and add support for
lucabe
parents:
2569
diff
changeset
|
787 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n"); |
d751acab2622
Allow to set the maximum number of frames per RTP packet (and add support for
lucabe
parents:
2569
diff
changeset
|
788 } else { |
d751acab2622
Allow to set the maximum number of frames per RTP packet (and add support for
lucabe
parents:
2569
diff
changeset
|
789 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); |
d751acab2622
Allow to set the maximum number of frames per RTP packet (and add support for
lucabe
parents:
2569
diff
changeset
|
790 } |
d751acab2622
Allow to set the maximum number of frames per RTP packet (and add support for
lucabe
parents:
2569
diff
changeset
|
791 } |
d751acab2622
Allow to set the maximum number of frames per RTP packet (and add support for
lucabe
parents:
2569
diff
changeset
|
792 if (st->codec->codec_type == CODEC_TYPE_VIDEO) { |
d751acab2622
Allow to set the maximum number of frames per RTP packet (and add support for
lucabe
parents:
2569
diff
changeset
|
793 /* FIXME: We should round down here... */ |
d751acab2622
Allow to set the maximum number of frames per RTP packet (and add support for
lucabe
parents:
2569
diff
changeset
|
794 s->max_frames_per_packet = av_rescale_q(s1->max_delay, AV_TIME_BASE_Q, st->codec->time_base); |
d751acab2622
Allow to set the maximum number of frames per RTP packet (and add support for
lucabe
parents:
2569
diff
changeset
|
795 } |
d751acab2622
Allow to set the maximum number of frames per RTP packet (and add support for
lucabe
parents:
2569
diff
changeset
|
796 } |
d751acab2622
Allow to set the maximum number of frames per RTP packet (and add support for
lucabe
parents:
2569
diff
changeset
|
797 |
2540
ca3cba9c641f
Fix timestamps in RTP packets (now, MPEG1 video with B frames works correctly)
lucabe
parents:
2539
diff
changeset
|
798 av_set_pts_info(st, 32, 1, 90000); |
820
feca73904e67
changing AVCodecContext codec -> *codec in AVStream so additions to AVCodecContext dont randomize AVStream and break binary compatibility
michael
parents:
777
diff
changeset
|
799 switch(st->codec->codec_id) { |
0 | 800 case CODEC_ID_MP2: |
232 | 801 case CODEC_ID_MP3: |
0 | 802 s->buf_ptr = s->buf + 4; |
803 break; | |
804 case CODEC_ID_MPEG1VIDEO: | |
2760 | 805 case CODEC_ID_MPEG2VIDEO: |
0 | 806 break; |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
807 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
|
808 n = s->max_payload_size / TS_PACKET_SIZE; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
809 if (n < 1) |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
810 n = 1; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
811 s->max_payload_size = n * TS_PACKET_SIZE; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
812 s->buf_ptr = s->buf; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
813 break; |
2550
e9c34ec665c6
Support for AAC streaming over RTP. Fragmentation is not implemented yet
lucabe
parents:
2540
diff
changeset
|
814 case CODEC_ID_AAC: |
e9c34ec665c6
Support for AAC streaming over RTP. Fragmentation is not implemented yet
lucabe
parents:
2540
diff
changeset
|
815 s->read_buf_index = 0; |
0 | 816 default: |
2539
ba933dfa4833
Properly set RTP and NTP timestamps in RTCP SR packets
lucabe
parents:
2406
diff
changeset
|
817 if (st->codec->codec_type == CODEC_TYPE_AUDIO) { |
ba933dfa4833
Properly set RTP and NTP timestamps in RTCP SR packets
lucabe
parents:
2406
diff
changeset
|
818 av_set_pts_info(st, 32, 1, st->codec->sample_rate); |
ba933dfa4833
Properly set RTP and NTP timestamps in RTCP SR packets
lucabe
parents:
2406
diff
changeset
|
819 } |
0 | 820 s->buf_ptr = s->buf; |
821 break; | |
822 } | |
823 | |
824 return 0; | |
825 } | |
826 | |
827 /* send an rtcp sender report packet */ | |
65 | 828 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time) |
0 | 829 { |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
830 RTPDemuxContext *s = s1->priv_data; |
2539
ba933dfa4833
Properly set RTP and NTP timestamps in RTCP SR packets
lucabe
parents:
2406
diff
changeset
|
831 uint32_t rtp_ts; |
ba933dfa4833
Properly set RTP and NTP timestamps in RTCP SR packets
lucabe
parents:
2406
diff
changeset
|
832 |
0 | 833 #if defined(DEBUG) |
1443
404048ea11bc
Replace most of the %lld and %llx by their (cleaner) PRI*64 counterparts.
diego
parents:
1431
diff
changeset
|
834 printf("RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp); |
0 | 835 #endif |
2539
ba933dfa4833
Properly set RTP and NTP timestamps in RTCP SR packets
lucabe
parents:
2406
diff
changeset
|
836 |
ba933dfa4833
Properly set RTP and NTP timestamps in RTCP SR packets
lucabe
parents:
2406
diff
changeset
|
837 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = ntp_time; |
2706
b1723b8da595
Do not send too many RTCP packets (according to RFC 3550, the minimum
lucabe
parents:
2705
diff
changeset
|
838 s->last_rtcp_ntp_time = ntp_time; |
2539
ba933dfa4833
Properly set RTP and NTP timestamps in RTCP SR packets
lucabe
parents:
2406
diff
changeset
|
839 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, AV_TIME_BASE_Q, |
ba933dfa4833
Properly set RTP and NTP timestamps in RTCP SR packets
lucabe
parents:
2406
diff
changeset
|
840 s1->streams[0]->time_base) + s->base_timestamp; |
2771
d52c718e83f9
Use dynamically allocated ByteIOContext in AVFormatContext
andoma
parents:
2760
diff
changeset
|
841 put_byte(s1->pb, (RTP_VERSION << 6)); |
d52c718e83f9
Use dynamically allocated ByteIOContext in AVFormatContext
andoma
parents:
2760
diff
changeset
|
842 put_byte(s1->pb, 200); |
d52c718e83f9
Use dynamically allocated ByteIOContext in AVFormatContext
andoma
parents:
2760
diff
changeset
|
843 put_be16(s1->pb, 6); /* length in words - 1 */ |
d52c718e83f9
Use dynamically allocated ByteIOContext in AVFormatContext
andoma
parents:
2760
diff
changeset
|
844 put_be32(s1->pb, s->ssrc); |
d52c718e83f9
Use dynamically allocated ByteIOContext in AVFormatContext
andoma
parents:
2760
diff
changeset
|
845 put_be32(s1->pb, ntp_time / 1000000); |
d52c718e83f9
Use dynamically allocated ByteIOContext in AVFormatContext
andoma
parents:
2760
diff
changeset
|
846 put_be32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000); |
d52c718e83f9
Use dynamically allocated ByteIOContext in AVFormatContext
andoma
parents:
2760
diff
changeset
|
847 put_be32(s1->pb, rtp_ts); |
d52c718e83f9
Use dynamically allocated ByteIOContext in AVFormatContext
andoma
parents:
2760
diff
changeset
|
848 put_be32(s1->pb, s->packet_count); |
d52c718e83f9
Use dynamically allocated ByteIOContext in AVFormatContext
andoma
parents:
2760
diff
changeset
|
849 put_be32(s1->pb, s->octet_count); |
d52c718e83f9
Use dynamically allocated ByteIOContext in AVFormatContext
andoma
parents:
2760
diff
changeset
|
850 put_flush_packet(s1->pb); |
0 | 851 } |
852 | |
853 /* send an rtp packet. sequence number is incremented, but the caller | |
854 must update the timestamp itself */ | |
2406
18e94e5989d8
Move the RTP packetization code for MPEG12 video in its own file (rtp_mpv.c)
lucabe
parents:
2274
diff
changeset
|
855 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m) |
0 | 856 { |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
857 RTPDemuxContext *s = s1->priv_data; |
0 | 858 |
859 #ifdef DEBUG | |
860 printf("rtp_send_data size=%d\n", len); | |
861 #endif | |
862 | |
863 /* build the RTP header */ | |
2771
d52c718e83f9
Use dynamically allocated ByteIOContext in AVFormatContext
andoma
parents:
2760
diff
changeset
|
864 put_byte(s1->pb, (RTP_VERSION << 6)); |
d52c718e83f9
Use dynamically allocated ByteIOContext in AVFormatContext
andoma
parents:
2760
diff
changeset
|
865 put_byte(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7)); |
d52c718e83f9
Use dynamically allocated ByteIOContext in AVFormatContext
andoma
parents:
2760
diff
changeset
|
866 put_be16(s1->pb, s->seq); |
d52c718e83f9
Use dynamically allocated ByteIOContext in AVFormatContext
andoma
parents:
2760
diff
changeset
|
867 put_be32(s1->pb, s->timestamp); |
d52c718e83f9
Use dynamically allocated ByteIOContext in AVFormatContext
andoma
parents:
2760
diff
changeset
|
868 put_be32(s1->pb, s->ssrc); |
885 | 869 |
2771
d52c718e83f9
Use dynamically allocated ByteIOContext in AVFormatContext
andoma
parents:
2760
diff
changeset
|
870 put_buffer(s1->pb, buf1, len); |
d52c718e83f9
Use dynamically allocated ByteIOContext in AVFormatContext
andoma
parents:
2760
diff
changeset
|
871 put_flush_packet(s1->pb); |
885 | 872 |
0 | 873 s->seq++; |
874 s->octet_count += len; | |
875 s->packet_count++; | |
876 } | |
877 | |
878 /* send an integer number of samples and compute time stamp and fill | |
879 the rtp send buffer before sending. */ | |
880 static void rtp_send_samples(AVFormatContext *s1, | |
241 | 881 const uint8_t *buf1, int size, int sample_size) |
0 | 882 { |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
883 RTPDemuxContext *s = s1->priv_data; |
0 | 884 int len, max_packet_size, n; |
885 | |
886 max_packet_size = (s->max_payload_size / sample_size) * sample_size; | |
887 /* not needed, but who nows */ | |
888 if ((size % sample_size) != 0) | |
889 av_abort(); | |
2540
ca3cba9c641f
Fix timestamps in RTP packets (now, MPEG1 video with B frames works correctly)
lucabe
parents:
2539
diff
changeset
|
890 n = 0; |
0 | 891 while (size > 0) { |
2540
ca3cba9c641f
Fix timestamps in RTP packets (now, MPEG1 video with B frames works correctly)
lucabe
parents:
2539
diff
changeset
|
892 s->buf_ptr = s->buf; |
ca3cba9c641f
Fix timestamps in RTP packets (now, MPEG1 video with B frames works correctly)
lucabe
parents:
2539
diff
changeset
|
893 len = FFMIN(max_packet_size, size); |
0 | 894 |
895 /* copy data */ | |
896 memcpy(s->buf_ptr, buf1, len); | |
897 s->buf_ptr += len; | |
898 buf1 += len; | |
899 size -= len; | |
2540
ca3cba9c641f
Fix timestamps in RTP packets (now, MPEG1 video with B frames works correctly)
lucabe
parents:
2539
diff
changeset
|
900 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
|
901 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:
2539
diff
changeset
|
902 n += (s->buf_ptr - s->buf); |
0 | 903 } |
885 | 904 } |
0 | 905 |
906 /* NOTE: we suppose that exactly one frame is given as argument here */ | |
907 /* XXX: test it */ | |
908 static void rtp_send_mpegaudio(AVFormatContext *s1, | |
241 | 909 const uint8_t *buf1, int size) |
0 | 910 { |
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911 RTPDemuxContext *s = s1->priv_data; |
0 | 912 int len, count, max_packet_size; |
913 | |
914 max_packet_size = s->max_payload_size; | |
915 | |
916 /* test if we must flush because not enough space */ | |
917 len = (s->buf_ptr - s->buf); | |
918 if ((len + size) > max_packet_size) { | |
919 if (len > 4) { | |
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920 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); |
0 | 921 s->buf_ptr = s->buf + 4; |
922 } | |
923 } | |
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924 if (s->buf_ptr == s->buf + 4) { |
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925 s->timestamp = s->cur_timestamp; |
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926 } |
0 | 927 |
928 /* add the packet */ | |
929 if (size > max_packet_size) { | |
930 /* big packet: fragment */ | |
931 count = 0; | |
932 while (size > 0) { | |
933 len = max_packet_size - 4; | |
934 if (len > size) | |
935 len = size; | |
936 /* build fragmented packet */ | |
937 s->buf[0] = 0; | |
938 s->buf[1] = 0; | |
939 s->buf[2] = count >> 8; | |
940 s->buf[3] = count; | |
941 memcpy(s->buf + 4, buf1, len); | |
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942 ff_rtp_send_data(s1, s->buf, len + 4, 0); |
0 | 943 size -= len; |
944 buf1 += len; | |
945 count += len; | |
946 } | |
947 } else { | |
948 if (s->buf_ptr == s->buf + 4) { | |
949 /* no fragmentation possible */ | |
950 s->buf[0] = 0; | |
951 s->buf[1] = 0; | |
952 s->buf[2] = 0; | |
953 s->buf[3] = 0; | |
954 } | |
955 memcpy(s->buf_ptr, buf1, size); | |
956 s->buf_ptr += size; | |
957 } | |
958 } | |
959 | |
960 static void rtp_send_raw(AVFormatContext *s1, | |
241 | 961 const uint8_t *buf1, int size) |
0 | 962 { |
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963 RTPDemuxContext *s = s1->priv_data; |
0 | 964 int len, max_packet_size; |
965 | |
966 max_packet_size = s->max_payload_size; | |
967 | |
968 while (size > 0) { | |
969 len = max_packet_size; | |
970 if (len > size) | |
971 len = size; | |
972 | |
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973 s->timestamp = s->cur_timestamp; |
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974 ff_rtp_send_data(s1, buf1, len, (len == size)); |
0 | 975 |
976 buf1 += len; | |
977 size -= len; | |
978 } | |
979 } | |
980 | |
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981 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */ |
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982 static void rtp_send_mpegts_raw(AVFormatContext *s1, |
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983 const uint8_t *buf1, int size) |
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984 { |
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985 RTPDemuxContext *s = s1->priv_data; |
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986 int len, out_len; |
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987 |
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988 while (size >= TS_PACKET_SIZE) { |
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989 len = s->max_payload_size - (s->buf_ptr - s->buf); |
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990 if (len > size) |
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991 len = size; |
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992 memcpy(s->buf_ptr, buf1, len); |
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993 buf1 += len; |
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994 size -= len; |
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995 s->buf_ptr += len; |
885 | 996 |
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997 out_len = s->buf_ptr - s->buf; |
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998 if (out_len >= s->max_payload_size) { |
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999 ff_rtp_send_data(s1, s->buf, out_len, 0); |
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1000 s->buf_ptr = s->buf; |
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1001 } |
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1002 } |
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1003 } |
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1004 |
0 | 1005 /* write an RTP packet. 'buf1' must contain a single specific frame. */ |
468 | 1006 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) |
0 | 1007 { |
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1008 RTPDemuxContext *s = s1->priv_data; |
0 | 1009 AVStream *st = s1->streams[0]; |
1010 int rtcp_bytes; | |
468 | 1011 int size= pkt->size; |
1012 uint8_t *buf1= pkt->data; | |
885 | 1013 |
0 | 1014 #ifdef DEBUG |
468 | 1015 printf("%d: write len=%d\n", pkt->stream_index, size); |
0 | 1016 #endif |
1017 | |
1018 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */ | |
885 | 1019 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / |
0 | 1020 RTCP_TX_RATIO_DEN; |
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1021 if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) && |
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1022 (av_gettime() - s->last_rtcp_ntp_time > 5000000))) { |
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1023 rtcp_send_sr(s1, av_gettime()); |
0 | 1024 s->last_octet_count = s->octet_count; |
1025 s->first_packet = 0; | |
1026 } | |
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1027 s->cur_timestamp = s->base_timestamp + pkt->pts; |
0 | 1028 |
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1029 switch(st->codec->codec_id) { |
0 | 1030 case CODEC_ID_PCM_MULAW: |
1031 case CODEC_ID_PCM_ALAW: | |
1032 case CODEC_ID_PCM_U8: | |
1033 case CODEC_ID_PCM_S8: | |
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1034 rtp_send_samples(s1, buf1, size, 1 * st->codec->channels); |
0 | 1035 break; |
1036 case CODEC_ID_PCM_U16BE: | |
1037 case CODEC_ID_PCM_U16LE: | |
1038 case CODEC_ID_PCM_S16BE: | |
1039 case CODEC_ID_PCM_S16LE: | |
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1040 rtp_send_samples(s1, buf1, size, 2 * st->codec->channels); |
0 | 1041 break; |
1042 case CODEC_ID_MP2: | |
232 | 1043 case CODEC_ID_MP3: |
0 | 1044 rtp_send_mpegaudio(s1, buf1, size); |
1045 break; | |
1046 case CODEC_ID_MPEG1VIDEO: | |
2760 | 1047 case CODEC_ID_MPEG2VIDEO: |
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1048 ff_rtp_send_mpegvideo(s1, buf1, size); |
0 | 1049 break; |
2550
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1050 case CODEC_ID_AAC: |
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1051 ff_rtp_send_aac(s1, buf1, size); |
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1052 break; |
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1053 case CODEC_ID_MPEG2TS: |
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1054 rtp_send_mpegts_raw(s1, buf1, size); |
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1055 break; |
0 | 1056 default: |
1057 /* better than nothing : send the codec raw data */ | |
1058 rtp_send_raw(s1, buf1, size); | |
1059 break; | |
1060 } | |
1061 return 0; | |
1062 } | |
1063 | |
1167 | 1064 AVOutputFormat rtp_muxer = { |
0 | 1065 "rtp", |
1066 "RTP output format", | |
1067 NULL, | |
1068 NULL, | |
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1069 sizeof(RTPDemuxContext), |
0 | 1070 CODEC_ID_PCM_MULAW, |
1071 CODEC_ID_NONE, | |
1072 rtp_write_header, | |
1073 rtp_write_packet, | |
1074 }; |