Mercurial > libavformat.hg
annotate rtpenc.c @ 4980:3f9b378936f2 libavformat
move new_pes_av_stream to avoid forward declaration
author | bcoudurier |
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date | Sun, 31 May 2009 03:12:25 +0000 |
parents | 304a0ea063f0 |
children | 770a7de45356 |
rev | line source |
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0 | 1 /* |
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Split the RTP muxer out of rtp.c, to simplify the RTSP demuxer's dependencies
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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 |
4872 | 22 #include "libavcodec/get_bits.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 |
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42 static int is_supported(enum CodecID id) |
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43 { |
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44 switch(id) { |
4814 | 45 case CODEC_ID_H263: |
46 case CODEC_ID_H263P: | |
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47 case CODEC_ID_H264: |
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48 case CODEC_ID_MPEG1VIDEO: |
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49 case CODEC_ID_MPEG2VIDEO: |
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50 case CODEC_ID_MPEG4: |
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51 case CODEC_ID_AAC: |
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52 case CODEC_ID_MP2: |
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53 case CODEC_ID_MP3: |
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54 case CODEC_ID_PCM_ALAW: |
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55 case CODEC_ID_PCM_MULAW: |
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56 case CODEC_ID_PCM_S8: |
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57 case CODEC_ID_PCM_S16BE: |
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58 case CODEC_ID_PCM_S16LE: |
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59 case CODEC_ID_PCM_U16BE: |
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60 case CODEC_ID_PCM_U16LE: |
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61 case CODEC_ID_PCM_U8: |
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62 case CODEC_ID_MPEG2TS: |
4836 | 63 case CODEC_ID_AMR_NB: |
64 case CODEC_ID_AMR_WB: | |
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65 return 1; |
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66 default: |
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67 return 0; |
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68 } |
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69 } |
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70 |
0 | 71 static int rtp_write_header(AVFormatContext *s1) |
72 { | |
4388 | 73 RTPMuxContext *s = s1->priv_data; |
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74 int payload_type, max_packet_size, n; |
0 | 75 AVStream *st; |
76 | |
77 if (s1->nb_streams != 1) | |
78 return -1; | |
79 st = s1->streams[0]; | |
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80 if (!is_supported(st->codec->codec_id)) { |
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81 av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", st->codec->codec_id); |
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82 |
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83 return -1; |
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84 } |
0 | 85 |
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86 payload_type = ff_rtp_get_payload_type(st->codec); |
0 | 87 if (payload_type < 0) |
88 payload_type = RTP_PT_PRIVATE; /* private payload type */ | |
89 s->payload_type = payload_type; | |
90 | |
2790 | 91 // 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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92 s->base_timestamp = 0; /* FIXME: was random(), what should this be? */ |
0 | 93 s->timestamp = s->base_timestamp; |
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94 s->cur_timestamp = 0; |
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95 s->ssrc = 0; /* FIXME: was random(), what should this be? */ |
0 | 96 s->first_packet = 1; |
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97 s->first_rtcp_ntp_time = AV_NOPTS_VALUE; |
0 | 98 |
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99 max_packet_size = url_fget_max_packet_size(s1->pb); |
0 | 100 if (max_packet_size <= 12) |
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101 return AVERROR(EIO); |
4391 | 102 s->buf = av_malloc(max_packet_size); |
103 if (s->buf == NULL) { | |
104 return AVERROR(ENOMEM); | |
105 } | |
0 | 106 s->max_payload_size = max_packet_size - 12; |
107 | |
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108 s->max_frames_per_packet = 0; |
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109 if (s1->max_delay) { |
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110 if (st->codec->codec_type == CODEC_TYPE_AUDIO) { |
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111 if (st->codec->frame_size == 0) { |
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112 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n"); |
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113 } else { |
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114 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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115 } |
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116 } |
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117 if (st->codec->codec_type == CODEC_TYPE_VIDEO) { |
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118 /* FIXME: We should round down here... */ |
3500 | 119 s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base); |
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120 } |
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121 } |
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122 |
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123 av_set_pts_info(st, 32, 1, 90000); |
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124 switch(st->codec->codec_id) { |
0 | 125 case CODEC_ID_MP2: |
232 | 126 case CODEC_ID_MP3: |
0 | 127 s->buf_ptr = s->buf + 4; |
128 break; | |
129 case CODEC_ID_MPEG1VIDEO: | |
2760 | 130 case CODEC_ID_MPEG2VIDEO: |
0 | 131 break; |
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132 case CODEC_ID_MPEG2TS: |
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133 n = s->max_payload_size / TS_PACKET_SIZE; |
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134 if (n < 1) |
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135 n = 1; |
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136 s->max_payload_size = n * TS_PACKET_SIZE; |
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137 s->buf_ptr = s->buf; |
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138 break; |
4836 | 139 case CODEC_ID_AMR_NB: |
140 case CODEC_ID_AMR_WB: | |
141 if (!s->max_frames_per_packet) | |
142 s->max_frames_per_packet = 12; | |
143 if (st->codec->codec_id == CODEC_ID_AMR_NB) | |
144 n = 31; | |
145 else | |
146 n = 61; | |
147 /* max_header_toc_size + the largest AMR payload must fit */ | |
148 if (1 + s->max_frames_per_packet + n > s->max_payload_size) { | |
149 av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n"); | |
150 return -1; | |
151 } | |
152 if (st->codec->channels != 1) { | |
153 av_log(s1, AV_LOG_ERROR, "Only mono is supported\n"); | |
154 return -1; | |
155 } | |
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156 case CODEC_ID_AAC: |
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157 s->num_frames = 0; |
0 | 158 default: |
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159 if (st->codec->codec_type == CODEC_TYPE_AUDIO) { |
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160 av_set_pts_info(st, 32, 1, st->codec->sample_rate); |
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161 } |
0 | 162 s->buf_ptr = s->buf; |
163 break; | |
164 } | |
165 | |
166 return 0; | |
167 } | |
168 | |
169 /* send an rtcp sender report packet */ | |
65 | 170 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time) |
0 | 171 { |
4388 | 172 RTPMuxContext *s = s1->priv_data; |
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173 uint32_t rtp_ts; |
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174 |
3579 | 175 dprintf(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp); |
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176 |
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177 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = ntp_time; |
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178 s->last_rtcp_ntp_time = ntp_time; |
3500 | 179 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000}, |
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180 s1->streams[0]->time_base) + s->base_timestamp; |
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181 put_byte(s1->pb, (RTP_VERSION << 6)); |
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182 put_byte(s1->pb, 200); |
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183 put_be16(s1->pb, 6); /* length in words - 1 */ |
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184 put_be32(s1->pb, s->ssrc); |
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185 put_be32(s1->pb, ntp_time / 1000000); |
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186 put_be32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000); |
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187 put_be32(s1->pb, rtp_ts); |
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188 put_be32(s1->pb, s->packet_count); |
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189 put_be32(s1->pb, s->octet_count); |
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190 put_flush_packet(s1->pb); |
0 | 191 } |
192 | |
193 /* send an rtp packet. sequence number is incremented, but the caller | |
194 must update the timestamp itself */ | |
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195 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m) |
0 | 196 { |
4388 | 197 RTPMuxContext *s = s1->priv_data; |
0 | 198 |
3579 | 199 dprintf(s1, "rtp_send_data size=%d\n", len); |
0 | 200 |
201 /* build the RTP header */ | |
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202 put_byte(s1->pb, (RTP_VERSION << 6)); |
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203 put_byte(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7)); |
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204 put_be16(s1->pb, s->seq); |
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205 put_be32(s1->pb, s->timestamp); |
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206 put_be32(s1->pb, s->ssrc); |
885 | 207 |
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208 put_buffer(s1->pb, buf1, len); |
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209 put_flush_packet(s1->pb); |
885 | 210 |
0 | 211 s->seq++; |
212 s->octet_count += len; | |
213 s->packet_count++; | |
214 } | |
215 | |
216 /* send an integer number of samples and compute time stamp and fill | |
217 the rtp send buffer before sending. */ | |
218 static void rtp_send_samples(AVFormatContext *s1, | |
241 | 219 const uint8_t *buf1, int size, int sample_size) |
0 | 220 { |
4388 | 221 RTPMuxContext *s = s1->priv_data; |
0 | 222 int len, max_packet_size, n; |
223 | |
224 max_packet_size = (s->max_payload_size / sample_size) * sample_size; | |
225 /* not needed, but who nows */ | |
226 if ((size % sample_size) != 0) | |
227 av_abort(); | |
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228 n = 0; |
0 | 229 while (size > 0) { |
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230 s->buf_ptr = s->buf; |
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231 len = FFMIN(max_packet_size, size); |
0 | 232 |
233 /* copy data */ | |
234 memcpy(s->buf_ptr, buf1, len); | |
235 s->buf_ptr += len; | |
236 buf1 += len; | |
237 size -= len; | |
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238 s->timestamp = s->cur_timestamp + n / sample_size; |
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239 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); |
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240 n += (s->buf_ptr - s->buf); |
0 | 241 } |
885 | 242 } |
0 | 243 |
244 /* NOTE: we suppose that exactly one frame is given as argument here */ | |
245 /* XXX: test it */ | |
246 static void rtp_send_mpegaudio(AVFormatContext *s1, | |
241 | 247 const uint8_t *buf1, int size) |
0 | 248 { |
4388 | 249 RTPMuxContext *s = s1->priv_data; |
0 | 250 int len, count, max_packet_size; |
251 | |
252 max_packet_size = s->max_payload_size; | |
253 | |
254 /* test if we must flush because not enough space */ | |
255 len = (s->buf_ptr - s->buf); | |
256 if ((len + size) > max_packet_size) { | |
257 if (len > 4) { | |
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258 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); |
0 | 259 s->buf_ptr = s->buf + 4; |
260 } | |
261 } | |
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262 if (s->buf_ptr == s->buf + 4) { |
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263 s->timestamp = s->cur_timestamp; |
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264 } |
0 | 265 |
266 /* add the packet */ | |
267 if (size > max_packet_size) { | |
268 /* big packet: fragment */ | |
269 count = 0; | |
270 while (size > 0) { | |
271 len = max_packet_size - 4; | |
272 if (len > size) | |
273 len = size; | |
274 /* build fragmented packet */ | |
275 s->buf[0] = 0; | |
276 s->buf[1] = 0; | |
277 s->buf[2] = count >> 8; | |
278 s->buf[3] = count; | |
279 memcpy(s->buf + 4, buf1, len); | |
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280 ff_rtp_send_data(s1, s->buf, len + 4, 0); |
0 | 281 size -= len; |
282 buf1 += len; | |
283 count += len; | |
284 } | |
285 } else { | |
286 if (s->buf_ptr == s->buf + 4) { | |
287 /* no fragmentation possible */ | |
288 s->buf[0] = 0; | |
289 s->buf[1] = 0; | |
290 s->buf[2] = 0; | |
291 s->buf[3] = 0; | |
292 } | |
293 memcpy(s->buf_ptr, buf1, size); | |
294 s->buf_ptr += size; | |
295 } | |
296 } | |
297 | |
298 static void rtp_send_raw(AVFormatContext *s1, | |
241 | 299 const uint8_t *buf1, int size) |
0 | 300 { |
4388 | 301 RTPMuxContext *s = s1->priv_data; |
0 | 302 int len, max_packet_size; |
303 | |
304 max_packet_size = s->max_payload_size; | |
305 | |
306 while (size > 0) { | |
307 len = max_packet_size; | |
308 if (len > size) | |
309 len = size; | |
310 | |
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311 s->timestamp = s->cur_timestamp; |
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312 ff_rtp_send_data(s1, buf1, len, (len == size)); |
0 | 313 |
314 buf1 += len; | |
315 size -= len; | |
316 } | |
317 } | |
318 | |
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319 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */ |
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320 static void rtp_send_mpegts_raw(AVFormatContext *s1, |
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321 const uint8_t *buf1, int size) |
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322 { |
4388 | 323 RTPMuxContext *s = s1->priv_data; |
294
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324 int len, out_len; |
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325 |
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326 while (size >= TS_PACKET_SIZE) { |
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327 len = s->max_payload_size - (s->buf_ptr - s->buf); |
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328 if (len > size) |
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329 len = size; |
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330 memcpy(s->buf_ptr, buf1, len); |
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331 buf1 += len; |
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332 size -= len; |
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333 s->buf_ptr += len; |
885 | 334 |
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335 out_len = s->buf_ptr - s->buf; |
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336 if (out_len >= s->max_payload_size) { |
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337 ff_rtp_send_data(s1, s->buf, out_len, 0); |
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338 s->buf_ptr = s->buf; |
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339 } |
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340 } |
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341 } |
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342 |
0 | 343 /* write an RTP packet. 'buf1' must contain a single specific frame. */ |
468 | 344 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) |
0 | 345 { |
4388 | 346 RTPMuxContext *s = s1->priv_data; |
0 | 347 AVStream *st = s1->streams[0]; |
348 int rtcp_bytes; | |
468 | 349 int size= pkt->size; |
350 uint8_t *buf1= pkt->data; | |
885 | 351 |
3579 | 352 dprintf(s1, "%d: write len=%d\n", pkt->stream_index, size); |
0 | 353 |
885 | 354 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / |
0 | 355 RTCP_TX_RATIO_DEN; |
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356 if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) && |
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357 (ntp_time() - s->last_rtcp_ntp_time > 5000000))) { |
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358 rtcp_send_sr(s1, ntp_time()); |
0 | 359 s->last_octet_count = s->octet_count; |
360 s->first_packet = 0; | |
361 } | |
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362 s->cur_timestamp = s->base_timestamp + pkt->pts; |
0 | 363 |
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364 switch(st->codec->codec_id) { |
0 | 365 case CODEC_ID_PCM_MULAW: |
366 case CODEC_ID_PCM_ALAW: | |
367 case CODEC_ID_PCM_U8: | |
368 case CODEC_ID_PCM_S8: | |
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369 rtp_send_samples(s1, buf1, size, 1 * st->codec->channels); |
0 | 370 break; |
371 case CODEC_ID_PCM_U16BE: | |
372 case CODEC_ID_PCM_U16LE: | |
373 case CODEC_ID_PCM_S16BE: | |
374 case CODEC_ID_PCM_S16LE: | |
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375 rtp_send_samples(s1, buf1, size, 2 * st->codec->channels); |
0 | 376 break; |
377 case CODEC_ID_MP2: | |
232 | 378 case CODEC_ID_MP3: |
0 | 379 rtp_send_mpegaudio(s1, buf1, size); |
380 break; | |
381 case CODEC_ID_MPEG1VIDEO: | |
2760 | 382 case CODEC_ID_MPEG2VIDEO: |
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383 ff_rtp_send_mpegvideo(s1, buf1, size); |
0 | 384 break; |
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385 case CODEC_ID_AAC: |
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386 ff_rtp_send_aac(s1, buf1, size); |
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387 break; |
4836 | 388 case CODEC_ID_AMR_NB: |
389 case CODEC_ID_AMR_WB: | |
390 ff_rtp_send_amr(s1, buf1, size); | |
391 break; | |
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392 case CODEC_ID_MPEG2TS: |
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393 rtp_send_mpegts_raw(s1, buf1, size); |
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394 break; |
2960 | 395 case CODEC_ID_H264: |
396 ff_rtp_send_h264(s1, buf1, size); | |
397 break; | |
4814 | 398 case CODEC_ID_H263: |
399 case CODEC_ID_H263P: | |
400 ff_rtp_send_h263(s1, buf1, size); | |
401 break; | |
0 | 402 default: |
403 /* better than nothing : send the codec raw data */ | |
404 rtp_send_raw(s1, buf1, size); | |
405 break; | |
406 } | |
407 return 0; | |
408 } | |
409 | |
4391 | 410 static int rtp_write_trailer(AVFormatContext *s1) |
411 { | |
412 RTPMuxContext *s = s1->priv_data; | |
413 | |
414 av_freep(&s->buf); | |
415 | |
416 return 0; | |
417 } | |
418 | |
1167 | 419 AVOutputFormat rtp_muxer = { |
0 | 420 "rtp", |
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421 NULL_IF_CONFIG_SMALL("RTP output format"), |
0 | 422 NULL, |
423 NULL, | |
4388 | 424 sizeof(RTPMuxContext), |
0 | 425 CODEC_ID_PCM_MULAW, |
426 CODEC_ID_NONE, | |
427 rtp_write_header, | |
428 rtp_write_packet, | |
4391 | 429 rtp_write_trailer, |
0 | 430 }; |