3 * Copyright (c) 2002 Fabrice Bellard.
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
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.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
23 #include "bitstream.h"
28 #include "rtp_internal.h"
35 #define RTCP_SR_SIZE 28
36 #define NTP_OFFSET 2208988800ULL
37 #define NTP_OFFSET_US (NTP_OFFSET * 1000000ULL)
39 static uint64_t ntp_time(void)
41 return (av_gettime() / 1000) * 1000 + NTP_OFFSET_US;
44 static int rtp_write_header(AVFormatContext *s1)
46 RTPDemuxContext *s = s1->priv_data;
47 int payload_type, max_packet_size, n;
50 if (s1->nb_streams != 1)
54 payload_type = rtp_get_payload_type(st->codec);
56 payload_type = RTP_PT_PRIVATE; /* private payload type */
57 s->payload_type = payload_type;
59 // following 2 FIXMEs could be set based on the current time, there is normally no info leak, as RTP will likely be transmitted immediately
60 s->base_timestamp = 0; /* FIXME: was random(), what should this be? */
61 s->timestamp = s->base_timestamp;
63 s->ssrc = 0; /* FIXME: was random(), what should this be? */
65 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
67 max_packet_size = url_fget_max_packet_size(s1->pb);
68 if (max_packet_size <= 12)
70 s->max_payload_size = max_packet_size - 12;
72 s->max_frames_per_packet = 0;
74 if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
75 if (st->codec->frame_size == 0) {
76 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
78 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);
81 if (st->codec->codec_type == CODEC_TYPE_VIDEO) {
82 /* FIXME: We should round down here... */
83 s->max_frames_per_packet = av_rescale_q(s1->max_delay, AV_TIME_BASE_Q, st->codec->time_base);
87 av_set_pts_info(st, 32, 1, 90000);
88 switch(st->codec->codec_id) {
91 s->buf_ptr = s->buf + 4;
93 case CODEC_ID_MPEG1VIDEO:
94 case CODEC_ID_MPEG2VIDEO:
96 case CODEC_ID_MPEG2TS:
97 n = s->max_payload_size / TS_PACKET_SIZE;
100 s->max_payload_size = n * TS_PACKET_SIZE;
104 s->read_buf_index = 0;
106 if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
107 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
116 /* send an rtcp sender report packet */
117 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
119 RTPDemuxContext *s = s1->priv_data;
123 printf("RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
126 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = ntp_time;
127 s->last_rtcp_ntp_time = ntp_time;
128 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, AV_TIME_BASE_Q,
129 s1->streams[0]->time_base) + s->base_timestamp;
130 put_byte(s1->pb, (RTP_VERSION << 6));
131 put_byte(s1->pb, 200);
132 put_be16(s1->pb, 6); /* length in words - 1 */
133 put_be32(s1->pb, s->ssrc);
134 put_be32(s1->pb, ntp_time / 1000000);
135 put_be32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
136 put_be32(s1->pb, rtp_ts);
137 put_be32(s1->pb, s->packet_count);
138 put_be32(s1->pb, s->octet_count);
139 put_flush_packet(s1->pb);
142 /* send an rtp packet. sequence number is incremented, but the caller
143 must update the timestamp itself */
144 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
146 RTPDemuxContext *s = s1->priv_data;
149 printf("rtp_send_data size=%d\n", len);
152 /* build the RTP header */
153 put_byte(s1->pb, (RTP_VERSION << 6));
154 put_byte(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
155 put_be16(s1->pb, s->seq);
156 put_be32(s1->pb, s->timestamp);
157 put_be32(s1->pb, s->ssrc);
159 put_buffer(s1->pb, buf1, len);
160 put_flush_packet(s1->pb);
163 s->octet_count += len;
167 /* send an integer number of samples and compute time stamp and fill
168 the rtp send buffer before sending. */
169 static void rtp_send_samples(AVFormatContext *s1,
170 const uint8_t *buf1, int size, int sample_size)
172 RTPDemuxContext *s = s1->priv_data;
173 int len, max_packet_size, n;
175 max_packet_size = (s->max_payload_size / sample_size) * sample_size;
176 /* not needed, but who nows */
177 if ((size % sample_size) != 0)
182 len = FFMIN(max_packet_size, size);
185 memcpy(s->buf_ptr, buf1, len);
189 s->timestamp = s->cur_timestamp + n / sample_size;
190 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
191 n += (s->buf_ptr - s->buf);
195 /* NOTE: we suppose that exactly one frame is given as argument here */
197 static void rtp_send_mpegaudio(AVFormatContext *s1,
198 const uint8_t *buf1, int size)
200 RTPDemuxContext *s = s1->priv_data;
201 int len, count, max_packet_size;
203 max_packet_size = s->max_payload_size;
205 /* test if we must flush because not enough space */
206 len = (s->buf_ptr - s->buf);
207 if ((len + size) > max_packet_size) {
209 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
210 s->buf_ptr = s->buf + 4;
213 if (s->buf_ptr == s->buf + 4) {
214 s->timestamp = s->cur_timestamp;
218 if (size > max_packet_size) {
219 /* big packet: fragment */
222 len = max_packet_size - 4;
225 /* build fragmented packet */
228 s->buf[2] = count >> 8;
230 memcpy(s->buf + 4, buf1, len);
231 ff_rtp_send_data(s1, s->buf, len + 4, 0);
237 if (s->buf_ptr == s->buf + 4) {
238 /* no fragmentation possible */
244 memcpy(s->buf_ptr, buf1, size);
249 static void rtp_send_raw(AVFormatContext *s1,
250 const uint8_t *buf1, int size)
252 RTPDemuxContext *s = s1->priv_data;
253 int len, max_packet_size;
255 max_packet_size = s->max_payload_size;
258 len = max_packet_size;
262 s->timestamp = s->cur_timestamp;
263 ff_rtp_send_data(s1, buf1, len, (len == size));
270 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
271 static void rtp_send_mpegts_raw(AVFormatContext *s1,
272 const uint8_t *buf1, int size)
274 RTPDemuxContext *s = s1->priv_data;
277 while (size >= TS_PACKET_SIZE) {
278 len = s->max_payload_size - (s->buf_ptr - s->buf);
281 memcpy(s->buf_ptr, buf1, len);
286 out_len = s->buf_ptr - s->buf;
287 if (out_len >= s->max_payload_size) {
288 ff_rtp_send_data(s1, s->buf, out_len, 0);
294 /* write an RTP packet. 'buf1' must contain a single specific frame. */
295 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
297 RTPDemuxContext *s = s1->priv_data;
298 AVStream *st = s1->streams[0];
301 uint8_t *buf1= pkt->data;
304 printf("%d: write len=%d\n", pkt->stream_index, size);
307 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
308 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
310 if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
311 (ntp_time() - s->last_rtcp_ntp_time > 5000000))) {
312 rtcp_send_sr(s1, ntp_time());
313 s->last_octet_count = s->octet_count;
316 s->cur_timestamp = s->base_timestamp + pkt->pts;
318 switch(st->codec->codec_id) {
319 case CODEC_ID_PCM_MULAW:
320 case CODEC_ID_PCM_ALAW:
321 case CODEC_ID_PCM_U8:
322 case CODEC_ID_PCM_S8:
323 rtp_send_samples(s1, buf1, size, 1 * st->codec->channels);
325 case CODEC_ID_PCM_U16BE:
326 case CODEC_ID_PCM_U16LE:
327 case CODEC_ID_PCM_S16BE:
328 case CODEC_ID_PCM_S16LE:
329 rtp_send_samples(s1, buf1, size, 2 * st->codec->channels);
333 rtp_send_mpegaudio(s1, buf1, size);
335 case CODEC_ID_MPEG1VIDEO:
336 case CODEC_ID_MPEG2VIDEO:
337 ff_rtp_send_mpegvideo(s1, buf1, size);
340 ff_rtp_send_aac(s1, buf1, size);
342 case CODEC_ID_MPEG2TS:
343 rtp_send_mpegts_raw(s1, buf1, size);
346 ff_rtp_send_h264(s1, buf1, size);
349 /* better than nothing : send the codec raw data */
350 rtp_send_raw(s1, buf1, size);
356 AVOutputFormat rtp_muxer = {
361 sizeof(RTPDemuxContext),