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
22 #include "libavcodec/bitstream.h"
33 #define RTCP_SR_SIZE 28
34 #define NTP_OFFSET 2208988800ULL
35 #define NTP_OFFSET_US (NTP_OFFSET * 1000000ULL)
37 static uint64_t ntp_time(void)
39 return (av_gettime() / 1000) * 1000 + NTP_OFFSET_US;
42 static int is_supported(enum CodecID id)
48 case CODEC_ID_MPEG1VIDEO:
49 case CODEC_ID_MPEG2VIDEO:
54 case CODEC_ID_PCM_ALAW:
55 case CODEC_ID_PCM_MULAW:
57 case CODEC_ID_PCM_S16BE:
58 case CODEC_ID_PCM_S16LE:
59 case CODEC_ID_PCM_U16BE:
60 case CODEC_ID_PCM_U16LE:
62 case CODEC_ID_MPEG2TS:
69 static int rtp_write_header(AVFormatContext *s1)
71 RTPMuxContext *s = s1->priv_data;
72 int payload_type, max_packet_size, n;
75 if (s1->nb_streams != 1)
78 if (!is_supported(st->codec->codec_id)) {
79 av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", st->codec->codec_id);
84 payload_type = ff_rtp_get_payload_type(st->codec);
86 payload_type = RTP_PT_PRIVATE; /* private payload type */
87 s->payload_type = payload_type;
89 // following 2 FIXMEs could be set based on the current time, there is normally no info leak, as RTP will likely be transmitted immediately
90 s->base_timestamp = 0; /* FIXME: was random(), what should this be? */
91 s->timestamp = s->base_timestamp;
93 s->ssrc = 0; /* FIXME: was random(), what should this be? */
95 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
97 max_packet_size = url_fget_max_packet_size(s1->pb);
98 if (max_packet_size <= 12)
100 s->buf = av_malloc(max_packet_size);
101 if (s->buf == NULL) {
102 return AVERROR(ENOMEM);
104 s->max_payload_size = max_packet_size - 12;
106 s->max_frames_per_packet = 0;
108 if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
109 if (st->codec->frame_size == 0) {
110 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
112 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);
115 if (st->codec->codec_type == CODEC_TYPE_VIDEO) {
116 /* FIXME: We should round down here... */
117 s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
121 av_set_pts_info(st, 32, 1, 90000);
122 switch(st->codec->codec_id) {
125 s->buf_ptr = s->buf + 4;
127 case CODEC_ID_MPEG1VIDEO:
128 case CODEC_ID_MPEG2VIDEO:
130 case CODEC_ID_MPEG2TS:
131 n = s->max_payload_size / TS_PACKET_SIZE;
134 s->max_payload_size = n * TS_PACKET_SIZE;
140 if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
141 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
150 /* send an rtcp sender report packet */
151 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
153 RTPMuxContext *s = s1->priv_data;
156 dprintf(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
158 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = ntp_time;
159 s->last_rtcp_ntp_time = ntp_time;
160 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
161 s1->streams[0]->time_base) + s->base_timestamp;
162 put_byte(s1->pb, (RTP_VERSION << 6));
163 put_byte(s1->pb, 200);
164 put_be16(s1->pb, 6); /* length in words - 1 */
165 put_be32(s1->pb, s->ssrc);
166 put_be32(s1->pb, ntp_time / 1000000);
167 put_be32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
168 put_be32(s1->pb, rtp_ts);
169 put_be32(s1->pb, s->packet_count);
170 put_be32(s1->pb, s->octet_count);
171 put_flush_packet(s1->pb);
174 /* send an rtp packet. sequence number is incremented, but the caller
175 must update the timestamp itself */
176 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
178 RTPMuxContext *s = s1->priv_data;
180 dprintf(s1, "rtp_send_data size=%d\n", len);
182 /* build the RTP header */
183 put_byte(s1->pb, (RTP_VERSION << 6));
184 put_byte(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
185 put_be16(s1->pb, s->seq);
186 put_be32(s1->pb, s->timestamp);
187 put_be32(s1->pb, s->ssrc);
189 put_buffer(s1->pb, buf1, len);
190 put_flush_packet(s1->pb);
193 s->octet_count += len;
197 /* send an integer number of samples and compute time stamp and fill
198 the rtp send buffer before sending. */
199 static void rtp_send_samples(AVFormatContext *s1,
200 const uint8_t *buf1, int size, int sample_size)
202 RTPMuxContext *s = s1->priv_data;
203 int len, max_packet_size, n;
205 max_packet_size = (s->max_payload_size / sample_size) * sample_size;
206 /* not needed, but who nows */
207 if ((size % sample_size) != 0)
212 len = FFMIN(max_packet_size, size);
215 memcpy(s->buf_ptr, buf1, len);
219 s->timestamp = s->cur_timestamp + n / sample_size;
220 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
221 n += (s->buf_ptr - s->buf);
225 /* NOTE: we suppose that exactly one frame is given as argument here */
227 static void rtp_send_mpegaudio(AVFormatContext *s1,
228 const uint8_t *buf1, int size)
230 RTPMuxContext *s = s1->priv_data;
231 int len, count, max_packet_size;
233 max_packet_size = s->max_payload_size;
235 /* test if we must flush because not enough space */
236 len = (s->buf_ptr - s->buf);
237 if ((len + size) > max_packet_size) {
239 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
240 s->buf_ptr = s->buf + 4;
243 if (s->buf_ptr == s->buf + 4) {
244 s->timestamp = s->cur_timestamp;
248 if (size > max_packet_size) {
249 /* big packet: fragment */
252 len = max_packet_size - 4;
255 /* build fragmented packet */
258 s->buf[2] = count >> 8;
260 memcpy(s->buf + 4, buf1, len);
261 ff_rtp_send_data(s1, s->buf, len + 4, 0);
267 if (s->buf_ptr == s->buf + 4) {
268 /* no fragmentation possible */
274 memcpy(s->buf_ptr, buf1, size);
279 static void rtp_send_raw(AVFormatContext *s1,
280 const uint8_t *buf1, int size)
282 RTPMuxContext *s = s1->priv_data;
283 int len, max_packet_size;
285 max_packet_size = s->max_payload_size;
288 len = max_packet_size;
292 s->timestamp = s->cur_timestamp;
293 ff_rtp_send_data(s1, buf1, len, (len == size));
300 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
301 static void rtp_send_mpegts_raw(AVFormatContext *s1,
302 const uint8_t *buf1, int size)
304 RTPMuxContext *s = s1->priv_data;
307 while (size >= TS_PACKET_SIZE) {
308 len = s->max_payload_size - (s->buf_ptr - s->buf);
311 memcpy(s->buf_ptr, buf1, len);
316 out_len = s->buf_ptr - s->buf;
317 if (out_len >= s->max_payload_size) {
318 ff_rtp_send_data(s1, s->buf, out_len, 0);
324 /* write an RTP packet. 'buf1' must contain a single specific frame. */
325 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
327 RTPMuxContext *s = s1->priv_data;
328 AVStream *st = s1->streams[0];
331 uint8_t *buf1= pkt->data;
333 dprintf(s1, "%d: write len=%d\n", pkt->stream_index, size);
335 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
337 if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
338 (ntp_time() - s->last_rtcp_ntp_time > 5000000))) {
339 rtcp_send_sr(s1, ntp_time());
340 s->last_octet_count = s->octet_count;
343 s->cur_timestamp = s->base_timestamp + pkt->pts;
345 switch(st->codec->codec_id) {
346 case CODEC_ID_PCM_MULAW:
347 case CODEC_ID_PCM_ALAW:
348 case CODEC_ID_PCM_U8:
349 case CODEC_ID_PCM_S8:
350 rtp_send_samples(s1, buf1, size, 1 * st->codec->channels);
352 case CODEC_ID_PCM_U16BE:
353 case CODEC_ID_PCM_U16LE:
354 case CODEC_ID_PCM_S16BE:
355 case CODEC_ID_PCM_S16LE:
356 rtp_send_samples(s1, buf1, size, 2 * st->codec->channels);
360 rtp_send_mpegaudio(s1, buf1, size);
362 case CODEC_ID_MPEG1VIDEO:
363 case CODEC_ID_MPEG2VIDEO:
364 ff_rtp_send_mpegvideo(s1, buf1, size);
367 ff_rtp_send_aac(s1, buf1, size);
369 case CODEC_ID_MPEG2TS:
370 rtp_send_mpegts_raw(s1, buf1, size);
373 ff_rtp_send_h264(s1, buf1, size);
377 ff_rtp_send_h263(s1, buf1, size);
380 /* better than nothing : send the codec raw data */
381 rtp_send_raw(s1, buf1, size);
387 static int rtp_write_trailer(AVFormatContext *s1)
389 RTPMuxContext *s = s1->priv_data;
396 AVOutputFormat rtp_muxer = {
398 NULL_IF_CONFIG_SMALL("RTP output format"),
401 sizeof(RTPMuxContext),