Mercurial > libavformat.hg
annotate rtpenc.c @ 6099:ed24da0b0cee libavformat
bgr/rgb444 for nut
author | michael |
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date | Sun, 06 Jun 2010 11:43:53 +0000 |
parents | 201152a121b5 |
children | 6c0e0ad77315 |
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 |
0 | 22 #include "avformat.h" |
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23 #include "mpegts.h" |
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24 #include "internal.h" |
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25 #include "libavutil/random_seed.h" |
0 | 26 |
27 #include <unistd.h> | |
28 | |
4388 | 29 #include "rtpenc.h" |
1419 | 30 |
0 | 31 //#define DEBUG |
32 | |
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33 #define RTCP_SR_SIZE 28 |
0 | 34 |
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35 static int is_supported(enum CodecID id) |
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36 { |
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37 switch(id) { |
4814 | 38 case CODEC_ID_H263: |
39 case CODEC_ID_H263P: | |
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40 case CODEC_ID_H264: |
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41 case CODEC_ID_MPEG1VIDEO: |
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42 case CODEC_ID_MPEG2VIDEO: |
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43 case CODEC_ID_MPEG4: |
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44 case CODEC_ID_AAC: |
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45 case CODEC_ID_MP2: |
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46 case CODEC_ID_MP3: |
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47 case CODEC_ID_PCM_ALAW: |
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48 case CODEC_ID_PCM_MULAW: |
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49 case CODEC_ID_PCM_S8: |
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50 case CODEC_ID_PCM_S16BE: |
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51 case CODEC_ID_PCM_S16LE: |
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52 case CODEC_ID_PCM_U16BE: |
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53 case CODEC_ID_PCM_U16LE: |
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54 case CODEC_ID_PCM_U8: |
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55 case CODEC_ID_MPEG2TS: |
4836 | 56 case CODEC_ID_AMR_NB: |
57 case CODEC_ID_AMR_WB: | |
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58 return 1; |
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59 default: |
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60 return 0; |
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61 } |
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62 } |
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63 |
0 | 64 static int rtp_write_header(AVFormatContext *s1) |
65 { | |
4388 | 66 RTPMuxContext *s = s1->priv_data; |
5477 | 67 int max_packet_size, n; |
0 | 68 AVStream *st; |
69 | |
70 if (s1->nb_streams != 1) | |
71 return -1; | |
72 st = s1->streams[0]; | |
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73 if (!is_supported(st->codec->codec_id)) { |
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74 av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", st->codec->codec_id); |
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75 |
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76 return -1; |
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77 } |
0 | 78 |
5477 | 79 s->payload_type = ff_rtp_get_payload_type(st->codec); |
80 if (s->payload_type < 0) | |
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81 s->payload_type = RTP_PT_PRIVATE + (st->codec->codec_type == AVMEDIA_TYPE_AUDIO); |
0 | 82 |
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83 s->base_timestamp = av_get_random_seed(); |
0 | 84 s->timestamp = s->base_timestamp; |
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85 s->cur_timestamp = 0; |
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86 s->ssrc = av_get_random_seed(); |
0 | 87 s->first_packet = 1; |
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88 s->first_rtcp_ntp_time = ff_ntp_time(); |
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89 if (s1->start_time_realtime) |
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90 /* Round the NTP time to whole milliseconds. */ |
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91 s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 + |
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92 NTP_OFFSET_US; |
0 | 93 |
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94 max_packet_size = url_fget_max_packet_size(s1->pb); |
0 | 95 if (max_packet_size <= 12) |
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96 return AVERROR(EIO); |
4391 | 97 s->buf = av_malloc(max_packet_size); |
98 if (s->buf == NULL) { | |
99 return AVERROR(ENOMEM); | |
100 } | |
0 | 101 s->max_payload_size = max_packet_size - 12; |
102 | |
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103 s->max_frames_per_packet = 0; |
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104 if (s1->max_delay) { |
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105 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { |
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106 if (st->codec->frame_size == 0) { |
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107 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n"); |
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108 } else { |
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109 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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110 } |
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111 } |
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112 if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) { |
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113 /* FIXME: We should round down here... */ |
3500 | 114 s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base); |
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115 } |
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116 } |
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117 |
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118 av_set_pts_info(st, 32, 1, 90000); |
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119 switch(st->codec->codec_id) { |
0 | 120 case CODEC_ID_MP2: |
232 | 121 case CODEC_ID_MP3: |
0 | 122 s->buf_ptr = s->buf + 4; |
123 break; | |
124 case CODEC_ID_MPEG1VIDEO: | |
2760 | 125 case CODEC_ID_MPEG2VIDEO: |
0 | 126 break; |
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127 case CODEC_ID_MPEG2TS: |
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128 n = s->max_payload_size / TS_PACKET_SIZE; |
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129 if (n < 1) |
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130 n = 1; |
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131 s->max_payload_size = n * TS_PACKET_SIZE; |
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132 s->buf_ptr = s->buf; |
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133 break; |
4836 | 134 case CODEC_ID_AMR_NB: |
135 case CODEC_ID_AMR_WB: | |
136 if (!s->max_frames_per_packet) | |
137 s->max_frames_per_packet = 12; | |
138 if (st->codec->codec_id == CODEC_ID_AMR_NB) | |
139 n = 31; | |
140 else | |
141 n = 61; | |
142 /* max_header_toc_size + the largest AMR payload must fit */ | |
143 if (1 + s->max_frames_per_packet + n > s->max_payload_size) { | |
144 av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n"); | |
145 return -1; | |
146 } | |
147 if (st->codec->channels != 1) { | |
148 av_log(s1, AV_LOG_ERROR, "Only mono is supported\n"); | |
149 return -1; | |
150 } | |
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151 case CODEC_ID_AAC: |
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152 s->num_frames = 0; |
0 | 153 default: |
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154 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { |
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155 av_set_pts_info(st, 32, 1, st->codec->sample_rate); |
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156 } |
0 | 157 s->buf_ptr = s->buf; |
158 break; | |
159 } | |
160 | |
161 return 0; | |
162 } | |
163 | |
164 /* send an rtcp sender report packet */ | |
65 | 165 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time) |
0 | 166 { |
4388 | 167 RTPMuxContext *s = s1->priv_data; |
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168 uint32_t rtp_ts; |
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169 |
3579 | 170 dprintf(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp); |
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171 |
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172 s->last_rtcp_ntp_time = ntp_time; |
3500 | 173 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000}, |
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174 s1->streams[0]->time_base) + s->base_timestamp; |
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175 put_byte(s1->pb, (RTP_VERSION << 6)); |
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176 put_byte(s1->pb, 200); |
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177 put_be16(s1->pb, 6); /* length in words - 1 */ |
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178 put_be32(s1->pb, s->ssrc); |
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179 put_be32(s1->pb, ntp_time / 1000000); |
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180 put_be32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000); |
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181 put_be32(s1->pb, rtp_ts); |
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182 put_be32(s1->pb, s->packet_count); |
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183 put_be32(s1->pb, s->octet_count); |
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184 put_flush_packet(s1->pb); |
0 | 185 } |
186 | |
187 /* send an rtp packet. sequence number is incremented, but the caller | |
188 must update the timestamp itself */ | |
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189 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m) |
0 | 190 { |
4388 | 191 RTPMuxContext *s = s1->priv_data; |
0 | 192 |
3579 | 193 dprintf(s1, "rtp_send_data size=%d\n", len); |
0 | 194 |
195 /* build the RTP header */ | |
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196 put_byte(s1->pb, (RTP_VERSION << 6)); |
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197 put_byte(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7)); |
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198 put_be16(s1->pb, s->seq); |
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199 put_be32(s1->pb, s->timestamp); |
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200 put_be32(s1->pb, s->ssrc); |
885 | 201 |
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202 put_buffer(s1->pb, buf1, len); |
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203 put_flush_packet(s1->pb); |
885 | 204 |
0 | 205 s->seq++; |
206 s->octet_count += len; | |
207 s->packet_count++; | |
208 } | |
209 | |
210 /* send an integer number of samples and compute time stamp and fill | |
211 the rtp send buffer before sending. */ | |
212 static void rtp_send_samples(AVFormatContext *s1, | |
241 | 213 const uint8_t *buf1, int size, int sample_size) |
0 | 214 { |
4388 | 215 RTPMuxContext *s = s1->priv_data; |
0 | 216 int len, max_packet_size, n; |
217 | |
218 max_packet_size = (s->max_payload_size / sample_size) * sample_size; | |
219 /* not needed, but who nows */ | |
220 if ((size % sample_size) != 0) | |
221 av_abort(); | |
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222 n = 0; |
0 | 223 while (size > 0) { |
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224 s->buf_ptr = s->buf; |
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225 len = FFMIN(max_packet_size, size); |
0 | 226 |
227 /* copy data */ | |
228 memcpy(s->buf_ptr, buf1, len); | |
229 s->buf_ptr += len; | |
230 buf1 += len; | |
231 size -= len; | |
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232 s->timestamp = s->cur_timestamp + n / sample_size; |
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233 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); |
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234 n += (s->buf_ptr - s->buf); |
0 | 235 } |
885 | 236 } |
0 | 237 |
238 static void rtp_send_mpegaudio(AVFormatContext *s1, | |
241 | 239 const uint8_t *buf1, int size) |
0 | 240 { |
4388 | 241 RTPMuxContext *s = s1->priv_data; |
0 | 242 int len, count, max_packet_size; |
243 | |
244 max_packet_size = s->max_payload_size; | |
245 | |
246 /* test if we must flush because not enough space */ | |
247 len = (s->buf_ptr - s->buf); | |
248 if ((len + size) > max_packet_size) { | |
249 if (len > 4) { | |
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250 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); |
0 | 251 s->buf_ptr = s->buf + 4; |
252 } | |
253 } | |
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254 if (s->buf_ptr == s->buf + 4) { |
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255 s->timestamp = s->cur_timestamp; |
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256 } |
0 | 257 |
258 /* add the packet */ | |
259 if (size > max_packet_size) { | |
260 /* big packet: fragment */ | |
261 count = 0; | |
262 while (size > 0) { | |
263 len = max_packet_size - 4; | |
264 if (len > size) | |
265 len = size; | |
266 /* build fragmented packet */ | |
267 s->buf[0] = 0; | |
268 s->buf[1] = 0; | |
269 s->buf[2] = count >> 8; | |
270 s->buf[3] = count; | |
271 memcpy(s->buf + 4, buf1, len); | |
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272 ff_rtp_send_data(s1, s->buf, len + 4, 0); |
0 | 273 size -= len; |
274 buf1 += len; | |
275 count += len; | |
276 } | |
277 } else { | |
278 if (s->buf_ptr == s->buf + 4) { | |
279 /* no fragmentation possible */ | |
280 s->buf[0] = 0; | |
281 s->buf[1] = 0; | |
282 s->buf[2] = 0; | |
283 s->buf[3] = 0; | |
284 } | |
285 memcpy(s->buf_ptr, buf1, size); | |
286 s->buf_ptr += size; | |
287 } | |
288 } | |
289 | |
290 static void rtp_send_raw(AVFormatContext *s1, | |
241 | 291 const uint8_t *buf1, int size) |
0 | 292 { |
4388 | 293 RTPMuxContext *s = s1->priv_data; |
0 | 294 int len, max_packet_size; |
295 | |
296 max_packet_size = s->max_payload_size; | |
297 | |
298 while (size > 0) { | |
299 len = max_packet_size; | |
300 if (len > size) | |
301 len = size; | |
302 | |
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303 s->timestamp = s->cur_timestamp; |
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304 ff_rtp_send_data(s1, buf1, len, (len == size)); |
0 | 305 |
306 buf1 += len; | |
307 size -= len; | |
308 } | |
309 } | |
310 | |
294
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311 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */ |
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312 static void rtp_send_mpegts_raw(AVFormatContext *s1, |
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313 const uint8_t *buf1, int size) |
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314 { |
4388 | 315 RTPMuxContext *s = s1->priv_data; |
294
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316 int len, out_len; |
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317 |
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318 while (size >= TS_PACKET_SIZE) { |
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319 len = s->max_payload_size - (s->buf_ptr - s->buf); |
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320 if (len > size) |
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321 len = size; |
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322 memcpy(s->buf_ptr, buf1, len); |
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323 buf1 += len; |
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324 size -= len; |
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325 s->buf_ptr += len; |
885 | 326 |
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327 out_len = s->buf_ptr - s->buf; |
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328 if (out_len >= s->max_payload_size) { |
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329 ff_rtp_send_data(s1, s->buf, out_len, 0); |
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330 s->buf_ptr = s->buf; |
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331 } |
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332 } |
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333 } |
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334 |
468 | 335 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) |
0 | 336 { |
4388 | 337 RTPMuxContext *s = s1->priv_data; |
0 | 338 AVStream *st = s1->streams[0]; |
339 int rtcp_bytes; | |
468 | 340 int size= pkt->size; |
885 | 341 |
3579 | 342 dprintf(s1, "%d: write len=%d\n", pkt->stream_index, size); |
0 | 343 |
885 | 344 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / |
0 | 345 RTCP_TX_RATIO_DEN; |
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346 if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) && |
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347 (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) { |
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348 rtcp_send_sr(s1, ff_ntp_time()); |
0 | 349 s->last_octet_count = s->octet_count; |
350 s->first_packet = 0; | |
351 } | |
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352 s->cur_timestamp = s->base_timestamp + pkt->pts; |
0 | 353 |
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354 switch(st->codec->codec_id) { |
0 | 355 case CODEC_ID_PCM_MULAW: |
356 case CODEC_ID_PCM_ALAW: | |
357 case CODEC_ID_PCM_U8: | |
358 case CODEC_ID_PCM_S8: | |
5479 | 359 rtp_send_samples(s1, pkt->data, size, 1 * st->codec->channels); |
0 | 360 break; |
361 case CODEC_ID_PCM_U16BE: | |
362 case CODEC_ID_PCM_U16LE: | |
363 case CODEC_ID_PCM_S16BE: | |
364 case CODEC_ID_PCM_S16LE: | |
5479 | 365 rtp_send_samples(s1, pkt->data, size, 2 * st->codec->channels); |
0 | 366 break; |
367 case CODEC_ID_MP2: | |
232 | 368 case CODEC_ID_MP3: |
5479 | 369 rtp_send_mpegaudio(s1, pkt->data, size); |
0 | 370 break; |
371 case CODEC_ID_MPEG1VIDEO: | |
2760 | 372 case CODEC_ID_MPEG2VIDEO: |
5479 | 373 ff_rtp_send_mpegvideo(s1, pkt->data, size); |
0 | 374 break; |
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375 case CODEC_ID_AAC: |
5479 | 376 ff_rtp_send_aac(s1, pkt->data, size); |
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377 break; |
4836 | 378 case CODEC_ID_AMR_NB: |
379 case CODEC_ID_AMR_WB: | |
5479 | 380 ff_rtp_send_amr(s1, pkt->data, size); |
4836 | 381 break; |
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382 case CODEC_ID_MPEG2TS: |
5479 | 383 rtp_send_mpegts_raw(s1, pkt->data, size); |
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384 break; |
2960 | 385 case CODEC_ID_H264: |
5479 | 386 ff_rtp_send_h264(s1, pkt->data, size); |
2960 | 387 break; |
4814 | 388 case CODEC_ID_H263: |
389 case CODEC_ID_H263P: | |
5479 | 390 ff_rtp_send_h263(s1, pkt->data, size); |
4814 | 391 break; |
0 | 392 default: |
393 /* better than nothing : send the codec raw data */ | |
5479 | 394 rtp_send_raw(s1, pkt->data, size); |
0 | 395 break; |
396 } | |
397 return 0; | |
398 } | |
399 | |
4391 | 400 static int rtp_write_trailer(AVFormatContext *s1) |
401 { | |
402 RTPMuxContext *s = s1->priv_data; | |
403 | |
404 av_freep(&s->buf); | |
405 | |
406 return 0; | |
407 } | |
408 | |
1167 | 409 AVOutputFormat rtp_muxer = { |
0 | 410 "rtp", |
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411 NULL_IF_CONFIG_SMALL("RTP output format"), |
0 | 412 NULL, |
413 NULL, | |
4388 | 414 sizeof(RTPMuxContext), |
0 | 415 CODEC_ID_PCM_MULAW, |
416 CODEC_ID_NONE, | |
417 rtp_write_header, | |
418 rtp_write_packet, | |
4391 | 419 rtp_write_trailer, |
0 | 420 }; |