Mercurial > libavformat.hg
annotate rtpenc.c @ 4006:cf359952a1fc libavformat
force sample rate to 16khz for speex in flv, fix speexaudio.flv
author | bcoudurier |
---|---|
date | Fri, 24 Oct 2008 18:36:26 +0000 |
parents | e6a42f4e5429 |
children | 77e0c7511d41 |
rev | line source |
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0 | 1 /* |
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2 * RTP output format |
0 | 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 */ |
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 |
1419 | 29 #include "rtp_internal.h" |
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30 #include "rtp_mpv.h" |
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31 #include "rtp_aac.h" |
2960 | 32 #include "rtp_h264.h" |
1419 | 33 |
0 | 34 //#define DEBUG |
35 | |
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36 #define RTCP_SR_SIZE 28 |
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37 #define NTP_OFFSET 2208988800ULL |
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38 #define NTP_OFFSET_US (NTP_OFFSET * 1000000ULL) |
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39 |
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40 static uint64_t ntp_time(void) |
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41 { |
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42 return (av_gettime() / 1000) * 1000 + NTP_OFFSET_US; |
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43 } |
0 | 44 |
45 static int rtp_write_header(AVFormatContext *s1) | |
46 { | |
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47 RTPDemuxContext *s = s1->priv_data; |
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48 int payload_type, max_packet_size, n; |
0 | 49 AVStream *st; |
50 | |
51 if (s1->nb_streams != 1) | |
52 return -1; | |
53 st = s1->streams[0]; | |
54 | |
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55 payload_type = rtp_get_payload_type(st->codec); |
0 | 56 if (payload_type < 0) |
57 payload_type = RTP_PT_PRIVATE; /* private payload type */ | |
58 s->payload_type = payload_type; | |
59 | |
2790 | 60 // 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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61 s->base_timestamp = 0; /* FIXME: was random(), what should this be? */ |
0 | 62 s->timestamp = s->base_timestamp; |
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63 s->cur_timestamp = 0; |
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64 s->ssrc = 0; /* FIXME: was random(), what should this be? */ |
0 | 65 s->first_packet = 1; |
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66 s->first_rtcp_ntp_time = AV_NOPTS_VALUE; |
0 | 67 |
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68 max_packet_size = url_fget_max_packet_size(s1->pb); |
0 | 69 if (max_packet_size <= 12) |
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70 return AVERROR(EIO); |
0 | 71 s->max_payload_size = max_packet_size - 12; |
72 | |
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73 s->max_frames_per_packet = 0; |
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74 if (s1->max_delay) { |
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75 if (st->codec->codec_type == CODEC_TYPE_AUDIO) { |
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76 if (st->codec->frame_size == 0) { |
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77 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n"); |
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78 } else { |
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79 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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80 } |
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81 } |
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82 if (st->codec->codec_type == CODEC_TYPE_VIDEO) { |
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83 /* FIXME: We should round down here... */ |
3500 | 84 s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base); |
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85 } |
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86 } |
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87 |
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88 av_set_pts_info(st, 32, 1, 90000); |
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89 switch(st->codec->codec_id) { |
0 | 90 case CODEC_ID_MP2: |
232 | 91 case CODEC_ID_MP3: |
0 | 92 s->buf_ptr = s->buf + 4; |
93 break; | |
94 case CODEC_ID_MPEG1VIDEO: | |
2760 | 95 case CODEC_ID_MPEG2VIDEO: |
0 | 96 break; |
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97 case CODEC_ID_MPEG2TS: |
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98 n = s->max_payload_size / TS_PACKET_SIZE; |
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99 if (n < 1) |
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100 n = 1; |
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101 s->max_payload_size = n * TS_PACKET_SIZE; |
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102 s->buf_ptr = s->buf; |
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103 break; |
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104 case CODEC_ID_AAC: |
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105 s->read_buf_index = 0; |
0 | 106 default: |
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107 if (st->codec->codec_type == CODEC_TYPE_AUDIO) { |
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108 av_set_pts_info(st, 32, 1, st->codec->sample_rate); |
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109 } |
0 | 110 s->buf_ptr = s->buf; |
111 break; | |
112 } | |
113 | |
114 return 0; | |
115 } | |
116 | |
117 /* send an rtcp sender report packet */ | |
65 | 118 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time) |
0 | 119 { |
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120 RTPDemuxContext *s = s1->priv_data; |
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121 uint32_t rtp_ts; |
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122 |
3579 | 123 dprintf(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp); |
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124 |
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125 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = ntp_time; |
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126 s->last_rtcp_ntp_time = ntp_time; |
3500 | 127 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000}, |
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128 s1->streams[0]->time_base) + s->base_timestamp; |
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129 put_byte(s1->pb, (RTP_VERSION << 6)); |
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130 put_byte(s1->pb, 200); |
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131 put_be16(s1->pb, 6); /* length in words - 1 */ |
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132 put_be32(s1->pb, s->ssrc); |
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133 put_be32(s1->pb, ntp_time / 1000000); |
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134 put_be32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000); |
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135 put_be32(s1->pb, rtp_ts); |
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136 put_be32(s1->pb, s->packet_count); |
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137 put_be32(s1->pb, s->octet_count); |
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138 put_flush_packet(s1->pb); |
0 | 139 } |
140 | |
141 /* send an rtp packet. sequence number is incremented, but the caller | |
142 must update the timestamp itself */ | |
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143 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m) |
0 | 144 { |
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145 RTPDemuxContext *s = s1->priv_data; |
0 | 146 |
3579 | 147 dprintf(s1, "rtp_send_data size=%d\n", len); |
0 | 148 |
149 /* build the RTP header */ | |
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150 put_byte(s1->pb, (RTP_VERSION << 6)); |
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151 put_byte(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7)); |
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152 put_be16(s1->pb, s->seq); |
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153 put_be32(s1->pb, s->timestamp); |
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154 put_be32(s1->pb, s->ssrc); |
885 | 155 |
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156 put_buffer(s1->pb, buf1, len); |
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157 put_flush_packet(s1->pb); |
885 | 158 |
0 | 159 s->seq++; |
160 s->octet_count += len; | |
161 s->packet_count++; | |
162 } | |
163 | |
164 /* send an integer number of samples and compute time stamp and fill | |
165 the rtp send buffer before sending. */ | |
166 static void rtp_send_samples(AVFormatContext *s1, | |
241 | 167 const uint8_t *buf1, int size, int sample_size) |
0 | 168 { |
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169 RTPDemuxContext *s = s1->priv_data; |
0 | 170 int len, max_packet_size, n; |
171 | |
172 max_packet_size = (s->max_payload_size / sample_size) * sample_size; | |
173 /* not needed, but who nows */ | |
174 if ((size % sample_size) != 0) | |
175 av_abort(); | |
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176 n = 0; |
0 | 177 while (size > 0) { |
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178 s->buf_ptr = s->buf; |
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179 len = FFMIN(max_packet_size, size); |
0 | 180 |
181 /* copy data */ | |
182 memcpy(s->buf_ptr, buf1, len); | |
183 s->buf_ptr += len; | |
184 buf1 += len; | |
185 size -= len; | |
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186 s->timestamp = s->cur_timestamp + n / sample_size; |
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187 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); |
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188 n += (s->buf_ptr - s->buf); |
0 | 189 } |
885 | 190 } |
0 | 191 |
192 /* NOTE: we suppose that exactly one frame is given as argument here */ | |
193 /* XXX: test it */ | |
194 static void rtp_send_mpegaudio(AVFormatContext *s1, | |
241 | 195 const uint8_t *buf1, int size) |
0 | 196 { |
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197 RTPDemuxContext *s = s1->priv_data; |
0 | 198 int len, count, max_packet_size; |
199 | |
200 max_packet_size = s->max_payload_size; | |
201 | |
202 /* test if we must flush because not enough space */ | |
203 len = (s->buf_ptr - s->buf); | |
204 if ((len + size) > max_packet_size) { | |
205 if (len > 4) { | |
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206 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); |
0 | 207 s->buf_ptr = s->buf + 4; |
208 } | |
209 } | |
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210 if (s->buf_ptr == s->buf + 4) { |
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211 s->timestamp = s->cur_timestamp; |
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212 } |
0 | 213 |
214 /* add the packet */ | |
215 if (size > max_packet_size) { | |
216 /* big packet: fragment */ | |
217 count = 0; | |
218 while (size > 0) { | |
219 len = max_packet_size - 4; | |
220 if (len > size) | |
221 len = size; | |
222 /* build fragmented packet */ | |
223 s->buf[0] = 0; | |
224 s->buf[1] = 0; | |
225 s->buf[2] = count >> 8; | |
226 s->buf[3] = count; | |
227 memcpy(s->buf + 4, buf1, len); | |
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228 ff_rtp_send_data(s1, s->buf, len + 4, 0); |
0 | 229 size -= len; |
230 buf1 += len; | |
231 count += len; | |
232 } | |
233 } else { | |
234 if (s->buf_ptr == s->buf + 4) { | |
235 /* no fragmentation possible */ | |
236 s->buf[0] = 0; | |
237 s->buf[1] = 0; | |
238 s->buf[2] = 0; | |
239 s->buf[3] = 0; | |
240 } | |
241 memcpy(s->buf_ptr, buf1, size); | |
242 s->buf_ptr += size; | |
243 } | |
244 } | |
245 | |
246 static void rtp_send_raw(AVFormatContext *s1, | |
241 | 247 const uint8_t *buf1, int size) |
0 | 248 { |
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249 RTPDemuxContext *s = s1->priv_data; |
0 | 250 int len, max_packet_size; |
251 | |
252 max_packet_size = s->max_payload_size; | |
253 | |
254 while (size > 0) { | |
255 len = max_packet_size; | |
256 if (len > size) | |
257 len = size; | |
258 | |
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259 s->timestamp = s->cur_timestamp; |
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260 ff_rtp_send_data(s1, buf1, len, (len == size)); |
0 | 261 |
262 buf1 += len; | |
263 size -= len; | |
264 } | |
265 } | |
266 | |
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267 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */ |
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268 static void rtp_send_mpegts_raw(AVFormatContext *s1, |
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269 const uint8_t *buf1, int size) |
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270 { |
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271 RTPDemuxContext *s = s1->priv_data; |
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272 int len, out_len; |
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273 |
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274 while (size >= TS_PACKET_SIZE) { |
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275 len = s->max_payload_size - (s->buf_ptr - s->buf); |
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276 if (len > size) |
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277 len = size; |
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278 memcpy(s->buf_ptr, buf1, len); |
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279 buf1 += len; |
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280 size -= len; |
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281 s->buf_ptr += len; |
885 | 282 |
294
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283 out_len = s->buf_ptr - s->buf; |
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284 if (out_len >= s->max_payload_size) { |
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285 ff_rtp_send_data(s1, s->buf, out_len, 0); |
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286 s->buf_ptr = s->buf; |
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287 } |
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288 } |
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289 } |
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290 |
0 | 291 /* write an RTP packet. 'buf1' must contain a single specific frame. */ |
468 | 292 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) |
0 | 293 { |
294
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294 RTPDemuxContext *s = s1->priv_data; |
0 | 295 AVStream *st = s1->streams[0]; |
296 int rtcp_bytes; | |
468 | 297 int size= pkt->size; |
298 uint8_t *buf1= pkt->data; | |
885 | 299 |
3579 | 300 dprintf(s1, "%d: write len=%d\n", pkt->stream_index, size); |
0 | 301 |
885 | 302 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / |
0 | 303 RTCP_TX_RATIO_DEN; |
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304 if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) && |
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305 (ntp_time() - s->last_rtcp_ntp_time > 5000000))) { |
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306 rtcp_send_sr(s1, ntp_time()); |
0 | 307 s->last_octet_count = s->octet_count; |
308 s->first_packet = 0; | |
309 } | |
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310 s->cur_timestamp = s->base_timestamp + pkt->pts; |
0 | 311 |
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312 switch(st->codec->codec_id) { |
0 | 313 case CODEC_ID_PCM_MULAW: |
314 case CODEC_ID_PCM_ALAW: | |
315 case CODEC_ID_PCM_U8: | |
316 case CODEC_ID_PCM_S8: | |
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317 rtp_send_samples(s1, buf1, size, 1 * st->codec->channels); |
0 | 318 break; |
319 case CODEC_ID_PCM_U16BE: | |
320 case CODEC_ID_PCM_U16LE: | |
321 case CODEC_ID_PCM_S16BE: | |
322 case CODEC_ID_PCM_S16LE: | |
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323 rtp_send_samples(s1, buf1, size, 2 * st->codec->channels); |
0 | 324 break; |
325 case CODEC_ID_MP2: | |
232 | 326 case CODEC_ID_MP3: |
0 | 327 rtp_send_mpegaudio(s1, buf1, size); |
328 break; | |
329 case CODEC_ID_MPEG1VIDEO: | |
2760 | 330 case CODEC_ID_MPEG2VIDEO: |
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331 ff_rtp_send_mpegvideo(s1, buf1, size); |
0 | 332 break; |
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333 case CODEC_ID_AAC: |
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334 ff_rtp_send_aac(s1, buf1, size); |
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335 break; |
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336 case CODEC_ID_MPEG2TS: |
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337 rtp_send_mpegts_raw(s1, buf1, size); |
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338 break; |
2960 | 339 case CODEC_ID_H264: |
340 ff_rtp_send_h264(s1, buf1, size); | |
341 break; | |
0 | 342 default: |
343 /* better than nothing : send the codec raw data */ | |
344 rtp_send_raw(s1, buf1, size); | |
345 break; | |
346 } | |
347 return 0; | |
348 } | |
349 | |
1167 | 350 AVOutputFormat rtp_muxer = { |
0 | 351 "rtp", |
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352 NULL_IF_CONFIG_SMALL("RTP output format"), |
0 | 353 NULL, |
354 NULL, | |
294
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355 sizeof(RTPDemuxContext), |
0 | 356 CODEC_ID_PCM_MULAW, |
357 CODEC_ID_NONE, | |
358 rtp_write_header, | |
359 rtp_write_packet, | |
360 }; |