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
37 static int rtp_write_header(AVFormatContext *s1)
39 RTPDemuxContext *s = s1->priv_data;
40 int payload_type, max_packet_size, n;
43 if (s1->nb_streams != 1)
47 payload_type = rtp_get_payload_type(st->codec);
49 payload_type = RTP_PT_PRIVATE; /* private payload type */
50 s->payload_type = payload_type;
52 // following 2 FIXMEs could be set based on the current time, there is normally no info leak, as RTP will likely be transmitted immediately
53 s->base_timestamp = 0; /* FIXME: was random(), what should this be? */
54 s->timestamp = s->base_timestamp;
56 s->ssrc = 0; /* FIXME: was random(), what should this be? */
58 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
60 max_packet_size = url_fget_max_packet_size(s1->pb);
61 if (max_packet_size <= 12)
63 s->max_payload_size = max_packet_size - 12;
65 s->max_frames_per_packet = 0;
67 if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
68 if (st->codec->frame_size == 0) {
69 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
71 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);
74 if (st->codec->codec_type == CODEC_TYPE_VIDEO) {
75 /* FIXME: We should round down here... */
76 s->max_frames_per_packet = av_rescale_q(s1->max_delay, AV_TIME_BASE_Q, st->codec->time_base);
80 av_set_pts_info(st, 32, 1, 90000);
81 switch(st->codec->codec_id) {
84 s->buf_ptr = s->buf + 4;
86 case CODEC_ID_MPEG1VIDEO:
87 case CODEC_ID_MPEG2VIDEO:
89 case CODEC_ID_MPEG2TS:
90 n = s->max_payload_size / TS_PACKET_SIZE;
93 s->max_payload_size = n * TS_PACKET_SIZE;
97 s->read_buf_index = 0;
99 if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
100 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
109 /* send an rtcp sender report packet */
110 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
112 RTPDemuxContext *s = s1->priv_data;
116 printf("RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
119 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = ntp_time;
120 s->last_rtcp_ntp_time = ntp_time;
121 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, AV_TIME_BASE_Q,
122 s1->streams[0]->time_base) + s->base_timestamp;
123 put_byte(s1->pb, (RTP_VERSION << 6));
124 put_byte(s1->pb, 200);
125 put_be16(s1->pb, 6); /* length in words - 1 */
126 put_be32(s1->pb, s->ssrc);
127 put_be32(s1->pb, ntp_time / 1000000);
128 put_be32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
129 put_be32(s1->pb, rtp_ts);
130 put_be32(s1->pb, s->packet_count);
131 put_be32(s1->pb, s->octet_count);
132 put_flush_packet(s1->pb);
135 /* send an rtp packet. sequence number is incremented, but the caller
136 must update the timestamp itself */
137 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
139 RTPDemuxContext *s = s1->priv_data;
142 printf("rtp_send_data size=%d\n", len);
145 /* build the RTP header */
146 put_byte(s1->pb, (RTP_VERSION << 6));
147 put_byte(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
148 put_be16(s1->pb, s->seq);
149 put_be32(s1->pb, s->timestamp);
150 put_be32(s1->pb, s->ssrc);
152 put_buffer(s1->pb, buf1, len);
153 put_flush_packet(s1->pb);
156 s->octet_count += len;
160 /* send an integer number of samples and compute time stamp and fill
161 the rtp send buffer before sending. */
162 static void rtp_send_samples(AVFormatContext *s1,
163 const uint8_t *buf1, int size, int sample_size)
165 RTPDemuxContext *s = s1->priv_data;
166 int len, max_packet_size, n;
168 max_packet_size = (s->max_payload_size / sample_size) * sample_size;
169 /* not needed, but who nows */
170 if ((size % sample_size) != 0)
175 len = FFMIN(max_packet_size, size);
178 memcpy(s->buf_ptr, buf1, len);
182 s->timestamp = s->cur_timestamp + n / sample_size;
183 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
184 n += (s->buf_ptr - s->buf);
188 /* NOTE: we suppose that exactly one frame is given as argument here */
190 static void rtp_send_mpegaudio(AVFormatContext *s1,
191 const uint8_t *buf1, int size)
193 RTPDemuxContext *s = s1->priv_data;
194 int len, count, max_packet_size;
196 max_packet_size = s->max_payload_size;
198 /* test if we must flush because not enough space */
199 len = (s->buf_ptr - s->buf);
200 if ((len + size) > max_packet_size) {
202 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
203 s->buf_ptr = s->buf + 4;
206 if (s->buf_ptr == s->buf + 4) {
207 s->timestamp = s->cur_timestamp;
211 if (size > max_packet_size) {
212 /* big packet: fragment */
215 len = max_packet_size - 4;
218 /* build fragmented packet */
221 s->buf[2] = count >> 8;
223 memcpy(s->buf + 4, buf1, len);
224 ff_rtp_send_data(s1, s->buf, len + 4, 0);
230 if (s->buf_ptr == s->buf + 4) {
231 /* no fragmentation possible */
237 memcpy(s->buf_ptr, buf1, size);
242 static void rtp_send_raw(AVFormatContext *s1,
243 const uint8_t *buf1, int size)
245 RTPDemuxContext *s = s1->priv_data;
246 int len, max_packet_size;
248 max_packet_size = s->max_payload_size;
251 len = max_packet_size;
255 s->timestamp = s->cur_timestamp;
256 ff_rtp_send_data(s1, buf1, len, (len == size));
263 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
264 static void rtp_send_mpegts_raw(AVFormatContext *s1,
265 const uint8_t *buf1, int size)
267 RTPDemuxContext *s = s1->priv_data;
270 while (size >= TS_PACKET_SIZE) {
271 len = s->max_payload_size - (s->buf_ptr - s->buf);
274 memcpy(s->buf_ptr, buf1, len);
279 out_len = s->buf_ptr - s->buf;
280 if (out_len >= s->max_payload_size) {
281 ff_rtp_send_data(s1, s->buf, out_len, 0);
287 /* write an RTP packet. 'buf1' must contain a single specific frame. */
288 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
290 RTPDemuxContext *s = s1->priv_data;
291 AVStream *st = s1->streams[0];
294 uint8_t *buf1= pkt->data;
297 printf("%d: write len=%d\n", pkt->stream_index, size);
300 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
301 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
303 if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
304 (av_gettime() - s->last_rtcp_ntp_time > 5000000))) {
305 rtcp_send_sr(s1, av_gettime());
306 s->last_octet_count = s->octet_count;
309 s->cur_timestamp = s->base_timestamp + pkt->pts;
311 switch(st->codec->codec_id) {
312 case CODEC_ID_PCM_MULAW:
313 case CODEC_ID_PCM_ALAW:
314 case CODEC_ID_PCM_U8:
315 case CODEC_ID_PCM_S8:
316 rtp_send_samples(s1, buf1, size, 1 * st->codec->channels);
318 case CODEC_ID_PCM_U16BE:
319 case CODEC_ID_PCM_U16LE:
320 case CODEC_ID_PCM_S16BE:
321 case CODEC_ID_PCM_S16LE:
322 rtp_send_samples(s1, buf1, size, 2 * st->codec->channels);
326 rtp_send_mpegaudio(s1, buf1, size);
328 case CODEC_ID_MPEG1VIDEO:
329 case CODEC_ID_MPEG2VIDEO:
330 ff_rtp_send_mpegvideo(s1, buf1, size);
333 ff_rtp_send_aac(s1, buf1, size);
335 case CODEC_ID_MPEG2TS:
336 rtp_send_mpegts_raw(s1, buf1, size);
339 ff_rtp_send_h264(s1, buf1, size);
342 /* better than nothing : send the codec raw data */
343 rtp_send_raw(s1, buf1, size);
349 AVOutputFormat rtp_muxer = {
354 sizeof(RTPDemuxContext),