Mercurial > libavformat.hg
annotate rtpenc.c @ 4389:25430a31705d libavformat
Merge rtp_aac.h and rtp_mpv.h in rtpenc.h
author | lucabe |
---|---|
date | Fri, 06 Feb 2009 15:30:41 +0000 |
parents | 80f21f72d7d6 |
children | 3c0d674bd232 |
rev | line source |
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0 | 1 /* |
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2 * RTP output format |
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3 * Copyright (c) 2002 Fabrice Bellard |
0 | 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 */ |
3286 | 21 |
22 #include "libavcodec/bitstream.h" | |
0 | 23 #include "avformat.h" |
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24 #include "mpegts.h" |
0 | 25 |
26 #include <unistd.h> | |
1754 | 27 #include "network.h" |
0 | 28 |
4388 | 29 #include "rtpenc.h" |
1419 | 30 |
0 | 31 //#define DEBUG |
32 | |
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33 #define RTCP_SR_SIZE 28 |
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34 #define NTP_OFFSET 2208988800ULL |
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35 #define NTP_OFFSET_US (NTP_OFFSET * 1000000ULL) |
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36 |
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37 static uint64_t ntp_time(void) |
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38 { |
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39 return (av_gettime() / 1000) * 1000 + NTP_OFFSET_US; |
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40 } |
0 | 41 |
42 static int rtp_write_header(AVFormatContext *s1) | |
43 { | |
4388 | 44 RTPMuxContext *s = s1->priv_data; |
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45 int payload_type, max_packet_size, n; |
0 | 46 AVStream *st; |
47 | |
48 if (s1->nb_streams != 1) | |
49 return -1; | |
50 st = s1->streams[0]; | |
51 | |
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52 payload_type = rtp_get_payload_type(st->codec); |
0 | 53 if (payload_type < 0) |
54 payload_type = RTP_PT_PRIVATE; /* private payload type */ | |
55 s->payload_type = payload_type; | |
56 | |
2790 | 57 // following 2 FIXMEs could be set based on the current time, there is normally no info leak, as RTP will likely be transmitted immediately |
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58 s->base_timestamp = 0; /* FIXME: was random(), what should this be? */ |
0 | 59 s->timestamp = s->base_timestamp; |
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60 s->cur_timestamp = 0; |
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61 s->ssrc = 0; /* FIXME: was random(), what should this be? */ |
0 | 62 s->first_packet = 1; |
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63 s->first_rtcp_ntp_time = AV_NOPTS_VALUE; |
0 | 64 |
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65 max_packet_size = url_fget_max_packet_size(s1->pb); |
0 | 66 if (max_packet_size <= 12) |
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67 return AVERROR(EIO); |
0 | 68 s->max_payload_size = max_packet_size - 12; |
69 | |
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70 s->max_frames_per_packet = 0; |
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71 if (s1->max_delay) { |
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72 if (st->codec->codec_type == CODEC_TYPE_AUDIO) { |
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73 if (st->codec->frame_size == 0) { |
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74 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n"); |
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75 } else { |
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76 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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77 } |
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78 } |
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79 if (st->codec->codec_type == CODEC_TYPE_VIDEO) { |
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80 /* FIXME: We should round down here... */ |
3500 | 81 s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base); |
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82 } |
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83 } |
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84 |
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85 av_set_pts_info(st, 32, 1, 90000); |
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86 switch(st->codec->codec_id) { |
0 | 87 case CODEC_ID_MP2: |
232 | 88 case CODEC_ID_MP3: |
0 | 89 s->buf_ptr = s->buf + 4; |
90 break; | |
91 case CODEC_ID_MPEG1VIDEO: | |
2760 | 92 case CODEC_ID_MPEG2VIDEO: |
0 | 93 break; |
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94 case CODEC_ID_MPEG2TS: |
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95 n = s->max_payload_size / TS_PACKET_SIZE; |
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96 if (n < 1) |
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97 n = 1; |
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98 s->max_payload_size = n * TS_PACKET_SIZE; |
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99 s->buf_ptr = s->buf; |
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100 break; |
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101 case CODEC_ID_AAC: |
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102 s->num_frames = 0; |
0 | 103 default: |
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104 if (st->codec->codec_type == CODEC_TYPE_AUDIO) { |
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105 av_set_pts_info(st, 32, 1, st->codec->sample_rate); |
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106 } |
0 | 107 s->buf_ptr = s->buf; |
108 break; | |
109 } | |
110 | |
111 return 0; | |
112 } | |
113 | |
114 /* send an rtcp sender report packet */ | |
65 | 115 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time) |
0 | 116 { |
4388 | 117 RTPMuxContext *s = s1->priv_data; |
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118 uint32_t rtp_ts; |
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119 |
3579 | 120 dprintf(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp); |
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121 |
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122 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = ntp_time; |
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123 s->last_rtcp_ntp_time = ntp_time; |
3500 | 124 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000}, |
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125 s1->streams[0]->time_base) + s->base_timestamp; |
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126 put_byte(s1->pb, (RTP_VERSION << 6)); |
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127 put_byte(s1->pb, 200); |
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128 put_be16(s1->pb, 6); /* length in words - 1 */ |
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129 put_be32(s1->pb, s->ssrc); |
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130 put_be32(s1->pb, ntp_time / 1000000); |
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131 put_be32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000); |
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132 put_be32(s1->pb, rtp_ts); |
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133 put_be32(s1->pb, s->packet_count); |
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134 put_be32(s1->pb, s->octet_count); |
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135 put_flush_packet(s1->pb); |
0 | 136 } |
137 | |
138 /* send an rtp packet. sequence number is incremented, but the caller | |
139 must update the timestamp itself */ | |
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140 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m) |
0 | 141 { |
4388 | 142 RTPMuxContext *s = s1->priv_data; |
0 | 143 |
3579 | 144 dprintf(s1, "rtp_send_data size=%d\n", len); |
0 | 145 |
146 /* build the RTP header */ | |
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147 put_byte(s1->pb, (RTP_VERSION << 6)); |
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148 put_byte(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7)); |
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149 put_be16(s1->pb, s->seq); |
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150 put_be32(s1->pb, s->timestamp); |
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151 put_be32(s1->pb, s->ssrc); |
885 | 152 |
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153 put_buffer(s1->pb, buf1, len); |
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154 put_flush_packet(s1->pb); |
885 | 155 |
0 | 156 s->seq++; |
157 s->octet_count += len; | |
158 s->packet_count++; | |
159 } | |
160 | |
161 /* send an integer number of samples and compute time stamp and fill | |
162 the rtp send buffer before sending. */ | |
163 static void rtp_send_samples(AVFormatContext *s1, | |
241 | 164 const uint8_t *buf1, int size, int sample_size) |
0 | 165 { |
4388 | 166 RTPMuxContext *s = s1->priv_data; |
0 | 167 int len, max_packet_size, n; |
168 | |
169 max_packet_size = (s->max_payload_size / sample_size) * sample_size; | |
170 /* not needed, but who nows */ | |
171 if ((size % sample_size) != 0) | |
172 av_abort(); | |
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173 n = 0; |
0 | 174 while (size > 0) { |
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175 s->buf_ptr = s->buf; |
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176 len = FFMIN(max_packet_size, size); |
0 | 177 |
178 /* copy data */ | |
179 memcpy(s->buf_ptr, buf1, len); | |
180 s->buf_ptr += len; | |
181 buf1 += len; | |
182 size -= len; | |
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183 s->timestamp = s->cur_timestamp + n / sample_size; |
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184 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); |
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185 n += (s->buf_ptr - s->buf); |
0 | 186 } |
885 | 187 } |
0 | 188 |
189 /* NOTE: we suppose that exactly one frame is given as argument here */ | |
190 /* XXX: test it */ | |
191 static void rtp_send_mpegaudio(AVFormatContext *s1, | |
241 | 192 const uint8_t *buf1, int size) |
0 | 193 { |
4388 | 194 RTPMuxContext *s = s1->priv_data; |
0 | 195 int len, count, max_packet_size; |
196 | |
197 max_packet_size = s->max_payload_size; | |
198 | |
199 /* test if we must flush because not enough space */ | |
200 len = (s->buf_ptr - s->buf); | |
201 if ((len + size) > max_packet_size) { | |
202 if (len > 4) { | |
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203 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); |
0 | 204 s->buf_ptr = s->buf + 4; |
205 } | |
206 } | |
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207 if (s->buf_ptr == s->buf + 4) { |
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208 s->timestamp = s->cur_timestamp; |
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209 } |
0 | 210 |
211 /* add the packet */ | |
212 if (size > max_packet_size) { | |
213 /* big packet: fragment */ | |
214 count = 0; | |
215 while (size > 0) { | |
216 len = max_packet_size - 4; | |
217 if (len > size) | |
218 len = size; | |
219 /* build fragmented packet */ | |
220 s->buf[0] = 0; | |
221 s->buf[1] = 0; | |
222 s->buf[2] = count >> 8; | |
223 s->buf[3] = count; | |
224 memcpy(s->buf + 4, buf1, len); | |
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225 ff_rtp_send_data(s1, s->buf, len + 4, 0); |
0 | 226 size -= len; |
227 buf1 += len; | |
228 count += len; | |
229 } | |
230 } else { | |
231 if (s->buf_ptr == s->buf + 4) { | |
232 /* no fragmentation possible */ | |
233 s->buf[0] = 0; | |
234 s->buf[1] = 0; | |
235 s->buf[2] = 0; | |
236 s->buf[3] = 0; | |
237 } | |
238 memcpy(s->buf_ptr, buf1, size); | |
239 s->buf_ptr += size; | |
240 } | |
241 } | |
242 | |
243 static void rtp_send_raw(AVFormatContext *s1, | |
241 | 244 const uint8_t *buf1, int size) |
0 | 245 { |
4388 | 246 RTPMuxContext *s = s1->priv_data; |
0 | 247 int len, max_packet_size; |
248 | |
249 max_packet_size = s->max_payload_size; | |
250 | |
251 while (size > 0) { | |
252 len = max_packet_size; | |
253 if (len > size) | |
254 len = size; | |
255 | |
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256 s->timestamp = s->cur_timestamp; |
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257 ff_rtp_send_data(s1, buf1, len, (len == size)); |
0 | 258 |
259 buf1 += len; | |
260 size -= len; | |
261 } | |
262 } | |
263 | |
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264 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */ |
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265 static void rtp_send_mpegts_raw(AVFormatContext *s1, |
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266 const uint8_t *buf1, int size) |
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267 { |
4388 | 268 RTPMuxContext *s = s1->priv_data; |
294
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269 int len, out_len; |
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270 |
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271 while (size >= TS_PACKET_SIZE) { |
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272 len = s->max_payload_size - (s->buf_ptr - s->buf); |
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273 if (len > size) |
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274 len = size; |
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275 memcpy(s->buf_ptr, buf1, len); |
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276 buf1 += len; |
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277 size -= len; |
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278 s->buf_ptr += len; |
885 | 279 |
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280 out_len = s->buf_ptr - s->buf; |
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281 if (out_len >= s->max_payload_size) { |
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282 ff_rtp_send_data(s1, s->buf, out_len, 0); |
294
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283 s->buf_ptr = s->buf; |
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284 } |
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285 } |
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286 } |
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287 |
0 | 288 /* write an RTP packet. 'buf1' must contain a single specific frame. */ |
468 | 289 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) |
0 | 290 { |
4388 | 291 RTPMuxContext *s = s1->priv_data; |
0 | 292 AVStream *st = s1->streams[0]; |
293 int rtcp_bytes; | |
468 | 294 int size= pkt->size; |
295 uint8_t *buf1= pkt->data; | |
885 | 296 |
3579 | 297 dprintf(s1, "%d: write len=%d\n", pkt->stream_index, size); |
0 | 298 |
885 | 299 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / |
0 | 300 RTCP_TX_RATIO_DEN; |
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301 if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) && |
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302 (ntp_time() - s->last_rtcp_ntp_time > 5000000))) { |
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303 rtcp_send_sr(s1, ntp_time()); |
0 | 304 s->last_octet_count = s->octet_count; |
305 s->first_packet = 0; | |
306 } | |
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307 s->cur_timestamp = s->base_timestamp + pkt->pts; |
0 | 308 |
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309 switch(st->codec->codec_id) { |
0 | 310 case CODEC_ID_PCM_MULAW: |
311 case CODEC_ID_PCM_ALAW: | |
312 case CODEC_ID_PCM_U8: | |
313 case CODEC_ID_PCM_S8: | |
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314 rtp_send_samples(s1, buf1, size, 1 * st->codec->channels); |
0 | 315 break; |
316 case CODEC_ID_PCM_U16BE: | |
317 case CODEC_ID_PCM_U16LE: | |
318 case CODEC_ID_PCM_S16BE: | |
319 case CODEC_ID_PCM_S16LE: | |
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320 rtp_send_samples(s1, buf1, size, 2 * st->codec->channels); |
0 | 321 break; |
322 case CODEC_ID_MP2: | |
232 | 323 case CODEC_ID_MP3: |
0 | 324 rtp_send_mpegaudio(s1, buf1, size); |
325 break; | |
326 case CODEC_ID_MPEG1VIDEO: | |
2760 | 327 case CODEC_ID_MPEG2VIDEO: |
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328 ff_rtp_send_mpegvideo(s1, buf1, size); |
0 | 329 break; |
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330 case CODEC_ID_AAC: |
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331 ff_rtp_send_aac(s1, buf1, size); |
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332 break; |
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333 case CODEC_ID_MPEG2TS: |
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334 rtp_send_mpegts_raw(s1, buf1, size); |
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335 break; |
2960 | 336 case CODEC_ID_H264: |
337 ff_rtp_send_h264(s1, buf1, size); | |
338 break; | |
0 | 339 default: |
340 /* better than nothing : send the codec raw data */ | |
341 rtp_send_raw(s1, buf1, size); | |
342 break; | |
343 } | |
344 return 0; | |
345 } | |
346 | |
1167 | 347 AVOutputFormat rtp_muxer = { |
0 | 348 "rtp", |
3424
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349 NULL_IF_CONFIG_SMALL("RTP output format"), |
0 | 350 NULL, |
351 NULL, | |
4388 | 352 sizeof(RTPMuxContext), |
0 | 353 CODEC_ID_PCM_MULAW, |
354 CODEC_ID_NONE, | |
355 rtp_write_header, | |
356 rtp_write_packet, | |
357 }; |