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
25 #include "libavutil/random_seed.h"
33 #define RTCP_SR_SIZE 28
35 static int is_supported(enum CodecID id)
41 case CODEC_ID_MPEG1VIDEO:
42 case CODEC_ID_MPEG2VIDEO:
47 case CODEC_ID_PCM_ALAW:
48 case CODEC_ID_PCM_MULAW:
50 case CODEC_ID_PCM_S16BE:
51 case CODEC_ID_PCM_S16LE:
52 case CODEC_ID_PCM_U16BE:
53 case CODEC_ID_PCM_U16LE:
55 case CODEC_ID_MPEG2TS:
64 static int rtp_write_header(AVFormatContext *s1)
66 RTPMuxContext *s = s1->priv_data;
67 int max_packet_size, n;
70 if (s1->nb_streams != 1)
73 if (!is_supported(st->codec->codec_id)) {
74 av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", st->codec->codec_id);
79 s->payload_type = ff_rtp_get_payload_type(st->codec);
80 if (s->payload_type < 0)
81 s->payload_type = RTP_PT_PRIVATE + (st->codec->codec_type == AVMEDIA_TYPE_AUDIO);
83 s->base_timestamp = av_get_random_seed();
84 s->timestamp = s->base_timestamp;
86 s->ssrc = av_get_random_seed();
88 s->first_rtcp_ntp_time = ff_ntp_time();
89 if (s1->start_time_realtime)
90 /* Round the NTP time to whole milliseconds. */
91 s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
94 max_packet_size = url_fget_max_packet_size(s1->pb);
95 if (max_packet_size <= 12)
97 s->buf = av_malloc(max_packet_size);
99 return AVERROR(ENOMEM);
101 s->max_payload_size = max_packet_size - 12;
103 s->max_frames_per_packet = 0;
105 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
106 if (st->codec->frame_size == 0) {
107 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
109 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);
112 if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
113 /* FIXME: We should round down here... */
114 s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
118 av_set_pts_info(st, 32, 1, 90000);
119 switch(st->codec->codec_id) {
122 s->buf_ptr = s->buf + 4;
124 case CODEC_ID_MPEG1VIDEO:
125 case CODEC_ID_MPEG2VIDEO:
127 case CODEC_ID_MPEG2TS:
128 n = s->max_payload_size / TS_PACKET_SIZE;
131 s->max_payload_size = n * TS_PACKET_SIZE;
134 case CODEC_ID_AMR_NB:
135 case CODEC_ID_AMR_WB:
136 if (!s->max_frames_per_packet)
137 s->max_frames_per_packet = 12;
138 if (st->codec->codec_id == CODEC_ID_AMR_NB)
142 /* max_header_toc_size + the largest AMR payload must fit */
143 if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
144 av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
147 if (st->codec->channels != 1) {
148 av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
154 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
155 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
164 /* send an rtcp sender report packet */
165 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
167 RTPMuxContext *s = s1->priv_data;
170 dprintf(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
172 s->last_rtcp_ntp_time = ntp_time;
173 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
174 s1->streams[0]->time_base) + s->base_timestamp;
175 put_byte(s1->pb, (RTP_VERSION << 6));
176 put_byte(s1->pb, 200);
177 put_be16(s1->pb, 6); /* length in words - 1 */
178 put_be32(s1->pb, s->ssrc);
179 put_be32(s1->pb, ntp_time / 1000000);
180 put_be32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
181 put_be32(s1->pb, rtp_ts);
182 put_be32(s1->pb, s->packet_count);
183 put_be32(s1->pb, s->octet_count);
184 put_flush_packet(s1->pb);
187 /* send an rtp packet. sequence number is incremented, but the caller
188 must update the timestamp itself */
189 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
191 RTPMuxContext *s = s1->priv_data;
193 dprintf(s1, "rtp_send_data size=%d\n", len);
195 /* build the RTP header */
196 put_byte(s1->pb, (RTP_VERSION << 6));
197 put_byte(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
198 put_be16(s1->pb, s->seq);
199 put_be32(s1->pb, s->timestamp);
200 put_be32(s1->pb, s->ssrc);
202 put_buffer(s1->pb, buf1, len);
203 put_flush_packet(s1->pb);
206 s->octet_count += len;
210 /* send an integer number of samples and compute time stamp and fill
211 the rtp send buffer before sending. */
212 static void rtp_send_samples(AVFormatContext *s1,
213 const uint8_t *buf1, int size, int sample_size)
215 RTPMuxContext *s = s1->priv_data;
216 int len, max_packet_size, n;
218 max_packet_size = (s->max_payload_size / sample_size) * sample_size;
219 /* not needed, but who nows */
220 if ((size % sample_size) != 0)
225 len = FFMIN(max_packet_size, size);
228 memcpy(s->buf_ptr, buf1, len);
232 s->timestamp = s->cur_timestamp + n / sample_size;
233 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
234 n += (s->buf_ptr - s->buf);
238 static void rtp_send_mpegaudio(AVFormatContext *s1,
239 const uint8_t *buf1, int size)
241 RTPMuxContext *s = s1->priv_data;
242 int len, count, max_packet_size;
244 max_packet_size = s->max_payload_size;
246 /* test if we must flush because not enough space */
247 len = (s->buf_ptr - s->buf);
248 if ((len + size) > max_packet_size) {
250 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
251 s->buf_ptr = s->buf + 4;
254 if (s->buf_ptr == s->buf + 4) {
255 s->timestamp = s->cur_timestamp;
259 if (size > max_packet_size) {
260 /* big packet: fragment */
263 len = max_packet_size - 4;
266 /* build fragmented packet */
269 s->buf[2] = count >> 8;
271 memcpy(s->buf + 4, buf1, len);
272 ff_rtp_send_data(s1, s->buf, len + 4, 0);
278 if (s->buf_ptr == s->buf + 4) {
279 /* no fragmentation possible */
285 memcpy(s->buf_ptr, buf1, size);
290 static void rtp_send_raw(AVFormatContext *s1,
291 const uint8_t *buf1, int size)
293 RTPMuxContext *s = s1->priv_data;
294 int len, max_packet_size;
296 max_packet_size = s->max_payload_size;
299 len = max_packet_size;
303 s->timestamp = s->cur_timestamp;
304 ff_rtp_send_data(s1, buf1, len, (len == size));
311 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
312 static void rtp_send_mpegts_raw(AVFormatContext *s1,
313 const uint8_t *buf1, int size)
315 RTPMuxContext *s = s1->priv_data;
318 while (size >= TS_PACKET_SIZE) {
319 len = s->max_payload_size - (s->buf_ptr - s->buf);
322 memcpy(s->buf_ptr, buf1, len);
327 out_len = s->buf_ptr - s->buf;
328 if (out_len >= s->max_payload_size) {
329 ff_rtp_send_data(s1, s->buf, out_len, 0);
335 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
337 RTPMuxContext *s = s1->priv_data;
338 AVStream *st = s1->streams[0];
342 dprintf(s1, "%d: write len=%d\n", pkt->stream_index, size);
344 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
346 if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
347 (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) {
348 rtcp_send_sr(s1, ff_ntp_time());
349 s->last_octet_count = s->octet_count;
352 s->cur_timestamp = s->base_timestamp + pkt->pts;
354 switch(st->codec->codec_id) {
355 case CODEC_ID_PCM_MULAW:
356 case CODEC_ID_PCM_ALAW:
357 case CODEC_ID_PCM_U8:
358 case CODEC_ID_PCM_S8:
359 rtp_send_samples(s1, pkt->data, size, 1 * st->codec->channels);
361 case CODEC_ID_PCM_U16BE:
362 case CODEC_ID_PCM_U16LE:
363 case CODEC_ID_PCM_S16BE:
364 case CODEC_ID_PCM_S16LE:
365 rtp_send_samples(s1, pkt->data, size, 2 * st->codec->channels);
369 rtp_send_mpegaudio(s1, pkt->data, size);
371 case CODEC_ID_MPEG1VIDEO:
372 case CODEC_ID_MPEG2VIDEO:
373 ff_rtp_send_mpegvideo(s1, pkt->data, size);
376 ff_rtp_send_aac(s1, pkt->data, size);
378 case CODEC_ID_AMR_NB:
379 case CODEC_ID_AMR_WB:
380 ff_rtp_send_amr(s1, pkt->data, size);
382 case CODEC_ID_MPEG2TS:
383 rtp_send_mpegts_raw(s1, pkt->data, size);
386 ff_rtp_send_h264(s1, pkt->data, size);
390 ff_rtp_send_h263(s1, pkt->data, size);
393 /* better than nothing : send the codec raw data */
394 rtp_send_raw(s1, pkt->data, size);
400 static int rtp_write_trailer(AVFormatContext *s1)
402 RTPMuxContext *s = s1->priv_data;
409 AVOutputFormat rtp_muxer = {
411 NULL_IF_CONFIG_SMALL("RTP output format"),
414 sizeof(RTPMuxContext),