Mercurial > libavformat.hg
annotate rtpenc.c @ 5477:2ba624c6c869 libavformat
Remove an unneeded local variable.
author | lucabe |
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date | Mon, 28 Dec 2009 13:19:54 +0000 |
parents | 770a7de45356 |
children | c75f52005f22 |
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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cosmetics: Remove pointless period after copyright statement non-sentences.
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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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Use a symbolic name for the payload size of an RTCP Sender Report packet
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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; |
5477 | 74 int 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 |
5477 | 86 s->payload_type = ff_rtp_get_payload_type(st->codec); |
87 if (s->payload_type < 0) | |
88 s->payload_type = RTP_PT_PRIVATE + (st->codec->codec_type == CODEC_TYPE_AUDIO); | |
0 | 89 |
2790 | 90 // 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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91 s->base_timestamp = 0; /* FIXME: was random(), what should this be? */ |
0 | 92 s->timestamp = s->base_timestamp; |
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93 s->cur_timestamp = 0; |
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94 s->ssrc = 0; /* FIXME: was random(), what should this be? */ |
0 | 95 s->first_packet = 1; |
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96 s->first_rtcp_ntp_time = AV_NOPTS_VALUE; |
0 | 97 |
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98 max_packet_size = url_fget_max_packet_size(s1->pb); |
0 | 99 if (max_packet_size <= 12) |
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100 return AVERROR(EIO); |
4391 | 101 s->buf = av_malloc(max_packet_size); |
102 if (s->buf == NULL) { | |
103 return AVERROR(ENOMEM); | |
104 } | |
0 | 105 s->max_payload_size = max_packet_size - 12; |
106 | |
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107 s->max_frames_per_packet = 0; |
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108 if (s1->max_delay) { |
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109 if (st->codec->codec_type == CODEC_TYPE_AUDIO) { |
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110 if (st->codec->frame_size == 0) { |
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111 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n"); |
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112 } else { |
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113 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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114 } |
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115 } |
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116 if (st->codec->codec_type == CODEC_TYPE_VIDEO) { |
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117 /* FIXME: We should round down here... */ |
3500 | 118 s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base); |
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119 } |
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120 } |
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121 |
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122 av_set_pts_info(st, 32, 1, 90000); |
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123 switch(st->codec->codec_id) { |
0 | 124 case CODEC_ID_MP2: |
232 | 125 case CODEC_ID_MP3: |
0 | 126 s->buf_ptr = s->buf + 4; |
127 break; | |
128 case CODEC_ID_MPEG1VIDEO: | |
2760 | 129 case CODEC_ID_MPEG2VIDEO: |
0 | 130 break; |
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131 case CODEC_ID_MPEG2TS: |
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132 n = s->max_payload_size / TS_PACKET_SIZE; |
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133 if (n < 1) |
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134 n = 1; |
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135 s->max_payload_size = n * TS_PACKET_SIZE; |
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136 s->buf_ptr = s->buf; |
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137 break; |
4836 | 138 case CODEC_ID_AMR_NB: |
139 case CODEC_ID_AMR_WB: | |
140 if (!s->max_frames_per_packet) | |
141 s->max_frames_per_packet = 12; | |
142 if (st->codec->codec_id == CODEC_ID_AMR_NB) | |
143 n = 31; | |
144 else | |
145 n = 61; | |
146 /* max_header_toc_size + the largest AMR payload must fit */ | |
147 if (1 + s->max_frames_per_packet + n > s->max_payload_size) { | |
148 av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n"); | |
149 return -1; | |
150 } | |
151 if (st->codec->channels != 1) { | |
152 av_log(s1, AV_LOG_ERROR, "Only mono is supported\n"); | |
153 return -1; | |
154 } | |
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155 case CODEC_ID_AAC: |
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156 s->num_frames = 0; |
0 | 157 default: |
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158 if (st->codec->codec_type == CODEC_TYPE_AUDIO) { |
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159 av_set_pts_info(st, 32, 1, st->codec->sample_rate); |
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160 } |
0 | 161 s->buf_ptr = s->buf; |
162 break; | |
163 } | |
164 | |
165 return 0; | |
166 } | |
167 | |
168 /* send an rtcp sender report packet */ | |
65 | 169 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time) |
0 | 170 { |
4388 | 171 RTPMuxContext *s = s1->priv_data; |
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172 uint32_t rtp_ts; |
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173 |
3579 | 174 dprintf(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp); |
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175 |
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176 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = ntp_time; |
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177 s->last_rtcp_ntp_time = ntp_time; |
3500 | 178 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000}, |
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179 s1->streams[0]->time_base) + s->base_timestamp; |
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180 put_byte(s1->pb, (RTP_VERSION << 6)); |
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181 put_byte(s1->pb, 200); |
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182 put_be16(s1->pb, 6); /* length in words - 1 */ |
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183 put_be32(s1->pb, s->ssrc); |
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184 put_be32(s1->pb, ntp_time / 1000000); |
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185 put_be32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000); |
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186 put_be32(s1->pb, rtp_ts); |
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187 put_be32(s1->pb, s->packet_count); |
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188 put_be32(s1->pb, s->octet_count); |
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189 put_flush_packet(s1->pb); |
0 | 190 } |
191 | |
192 /* send an rtp packet. sequence number is incremented, but the caller | |
193 must update the timestamp itself */ | |
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194 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m) |
0 | 195 { |
4388 | 196 RTPMuxContext *s = s1->priv_data; |
0 | 197 |
3579 | 198 dprintf(s1, "rtp_send_data size=%d\n", len); |
0 | 199 |
200 /* build the RTP header */ | |
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201 put_byte(s1->pb, (RTP_VERSION << 6)); |
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202 put_byte(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7)); |
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203 put_be16(s1->pb, s->seq); |
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204 put_be32(s1->pb, s->timestamp); |
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205 put_be32(s1->pb, s->ssrc); |
885 | 206 |
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207 put_buffer(s1->pb, buf1, len); |
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208 put_flush_packet(s1->pb); |
885 | 209 |
0 | 210 s->seq++; |
211 s->octet_count += len; | |
212 s->packet_count++; | |
213 } | |
214 | |
215 /* send an integer number of samples and compute time stamp and fill | |
216 the rtp send buffer before sending. */ | |
217 static void rtp_send_samples(AVFormatContext *s1, | |
241 | 218 const uint8_t *buf1, int size, int sample_size) |
0 | 219 { |
4388 | 220 RTPMuxContext *s = s1->priv_data; |
0 | 221 int len, max_packet_size, n; |
222 | |
223 max_packet_size = (s->max_payload_size / sample_size) * sample_size; | |
224 /* not needed, but who nows */ | |
225 if ((size % sample_size) != 0) | |
226 av_abort(); | |
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227 n = 0; |
0 | 228 while (size > 0) { |
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229 s->buf_ptr = s->buf; |
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230 len = FFMIN(max_packet_size, size); |
0 | 231 |
232 /* copy data */ | |
233 memcpy(s->buf_ptr, buf1, len); | |
234 s->buf_ptr += len; | |
235 buf1 += len; | |
236 size -= len; | |
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237 s->timestamp = s->cur_timestamp + n / sample_size; |
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238 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); |
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239 n += (s->buf_ptr - s->buf); |
0 | 240 } |
885 | 241 } |
0 | 242 |
243 /* NOTE: we suppose that exactly one frame is given as argument here */ | |
244 /* XXX: test it */ | |
245 static void rtp_send_mpegaudio(AVFormatContext *s1, | |
241 | 246 const uint8_t *buf1, int size) |
0 | 247 { |
4388 | 248 RTPMuxContext *s = s1->priv_data; |
0 | 249 int len, count, max_packet_size; |
250 | |
251 max_packet_size = s->max_payload_size; | |
252 | |
253 /* test if we must flush because not enough space */ | |
254 len = (s->buf_ptr - s->buf); | |
255 if ((len + size) > max_packet_size) { | |
256 if (len > 4) { | |
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257 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); |
0 | 258 s->buf_ptr = s->buf + 4; |
259 } | |
260 } | |
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261 if (s->buf_ptr == s->buf + 4) { |
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262 s->timestamp = s->cur_timestamp; |
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263 } |
0 | 264 |
265 /* add the packet */ | |
266 if (size > max_packet_size) { | |
267 /* big packet: fragment */ | |
268 count = 0; | |
269 while (size > 0) { | |
270 len = max_packet_size - 4; | |
271 if (len > size) | |
272 len = size; | |
273 /* build fragmented packet */ | |
274 s->buf[0] = 0; | |
275 s->buf[1] = 0; | |
276 s->buf[2] = count >> 8; | |
277 s->buf[3] = count; | |
278 memcpy(s->buf + 4, buf1, len); | |
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279 ff_rtp_send_data(s1, s->buf, len + 4, 0); |
0 | 280 size -= len; |
281 buf1 += len; | |
282 count += len; | |
283 } | |
284 } else { | |
285 if (s->buf_ptr == s->buf + 4) { | |
286 /* no fragmentation possible */ | |
287 s->buf[0] = 0; | |
288 s->buf[1] = 0; | |
289 s->buf[2] = 0; | |
290 s->buf[3] = 0; | |
291 } | |
292 memcpy(s->buf_ptr, buf1, size); | |
293 s->buf_ptr += size; | |
294 } | |
295 } | |
296 | |
297 static void rtp_send_raw(AVFormatContext *s1, | |
241 | 298 const uint8_t *buf1, int size) |
0 | 299 { |
4388 | 300 RTPMuxContext *s = s1->priv_data; |
0 | 301 int len, max_packet_size; |
302 | |
303 max_packet_size = s->max_payload_size; | |
304 | |
305 while (size > 0) { | |
306 len = max_packet_size; | |
307 if (len > size) | |
308 len = size; | |
309 | |
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310 s->timestamp = s->cur_timestamp; |
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311 ff_rtp_send_data(s1, buf1, len, (len == size)); |
0 | 312 |
313 buf1 += len; | |
314 size -= len; | |
315 } | |
316 } | |
317 | |
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318 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */ |
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319 static void rtp_send_mpegts_raw(AVFormatContext *s1, |
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320 const uint8_t *buf1, int size) |
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321 { |
4388 | 322 RTPMuxContext *s = s1->priv_data; |
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323 int len, out_len; |
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324 |
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325 while (size >= TS_PACKET_SIZE) { |
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326 len = s->max_payload_size - (s->buf_ptr - s->buf); |
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327 if (len > size) |
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328 len = size; |
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329 memcpy(s->buf_ptr, buf1, len); |
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330 buf1 += len; |
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331 size -= len; |
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332 s->buf_ptr += len; |
885 | 333 |
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334 out_len = s->buf_ptr - s->buf; |
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335 if (out_len >= s->max_payload_size) { |
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336 ff_rtp_send_data(s1, s->buf, out_len, 0); |
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337 s->buf_ptr = s->buf; |
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338 } |
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339 } |
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340 } |
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341 |
0 | 342 /* write an RTP packet. 'buf1' must contain a single specific frame. */ |
468 | 343 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) |
0 | 344 { |
4388 | 345 RTPMuxContext *s = s1->priv_data; |
0 | 346 AVStream *st = s1->streams[0]; |
347 int rtcp_bytes; | |
468 | 348 int size= pkt->size; |
349 uint8_t *buf1= pkt->data; | |
885 | 350 |
3579 | 351 dprintf(s1, "%d: write len=%d\n", pkt->stream_index, size); |
0 | 352 |
885 | 353 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / |
0 | 354 RTCP_TX_RATIO_DEN; |
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355 if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) && |
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356 (ntp_time() - s->last_rtcp_ntp_time > 5000000))) { |
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357 rtcp_send_sr(s1, ntp_time()); |
0 | 358 s->last_octet_count = s->octet_count; |
359 s->first_packet = 0; | |
360 } | |
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361 s->cur_timestamp = s->base_timestamp + pkt->pts; |
0 | 362 |
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363 switch(st->codec->codec_id) { |
0 | 364 case CODEC_ID_PCM_MULAW: |
365 case CODEC_ID_PCM_ALAW: | |
366 case CODEC_ID_PCM_U8: | |
367 case CODEC_ID_PCM_S8: | |
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368 rtp_send_samples(s1, buf1, size, 1 * st->codec->channels); |
0 | 369 break; |
370 case CODEC_ID_PCM_U16BE: | |
371 case CODEC_ID_PCM_U16LE: | |
372 case CODEC_ID_PCM_S16BE: | |
373 case CODEC_ID_PCM_S16LE: | |
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374 rtp_send_samples(s1, buf1, size, 2 * st->codec->channels); |
0 | 375 break; |
376 case CODEC_ID_MP2: | |
232 | 377 case CODEC_ID_MP3: |
0 | 378 rtp_send_mpegaudio(s1, buf1, size); |
379 break; | |
380 case CODEC_ID_MPEG1VIDEO: | |
2760 | 381 case CODEC_ID_MPEG2VIDEO: |
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382 ff_rtp_send_mpegvideo(s1, buf1, size); |
0 | 383 break; |
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384 case CODEC_ID_AAC: |
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385 ff_rtp_send_aac(s1, buf1, size); |
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386 break; |
4836 | 387 case CODEC_ID_AMR_NB: |
388 case CODEC_ID_AMR_WB: | |
389 ff_rtp_send_amr(s1, buf1, size); | |
390 break; | |
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391 case CODEC_ID_MPEG2TS: |
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392 rtp_send_mpegts_raw(s1, buf1, size); |
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393 break; |
2960 | 394 case CODEC_ID_H264: |
395 ff_rtp_send_h264(s1, buf1, size); | |
396 break; | |
4814 | 397 case CODEC_ID_H263: |
398 case CODEC_ID_H263P: | |
399 ff_rtp_send_h263(s1, buf1, size); | |
400 break; | |
0 | 401 default: |
402 /* better than nothing : send the codec raw data */ | |
403 rtp_send_raw(s1, buf1, size); | |
404 break; | |
405 } | |
406 return 0; | |
407 } | |
408 | |
4391 | 409 static int rtp_write_trailer(AVFormatContext *s1) |
410 { | |
411 RTPMuxContext *s = s1->priv_data; | |
412 | |
413 av_freep(&s->buf); | |
414 | |
415 return 0; | |
416 } | |
417 | |
1167 | 418 AVOutputFormat rtp_muxer = { |
0 | 419 "rtp", |
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420 NULL_IF_CONFIG_SMALL("RTP output format"), |
0 | 421 NULL, |
422 NULL, | |
4388 | 423 sizeof(RTPMuxContext), |
0 | 424 CODEC_ID_PCM_MULAW, |
425 CODEC_ID_NONE, | |
426 rtp_write_header, | |
427 rtp_write_packet, | |
4391 | 428 rtp_write_trailer, |
0 | 429 }; |