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"
31 #define RTCP_SR_SIZE 28
33 static int is_supported(enum CodecID id)
39 case CODEC_ID_MPEG1VIDEO:
40 case CODEC_ID_MPEG2VIDEO:
45 case CODEC_ID_PCM_ALAW:
46 case CODEC_ID_PCM_MULAW:
48 case CODEC_ID_PCM_S16BE:
49 case CODEC_ID_PCM_S16LE:
50 case CODEC_ID_PCM_U16BE:
51 case CODEC_ID_PCM_U16LE:
53 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;
135 /* check for H.264 MP4 syntax */
136 if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
137 s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
140 case CODEC_ID_VORBIS:
141 case CODEC_ID_THEORA:
142 if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
143 s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
144 s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
147 case CODEC_ID_AMR_NB:
148 case CODEC_ID_AMR_WB:
149 if (!s->max_frames_per_packet)
150 s->max_frames_per_packet = 12;
151 if (st->codec->codec_id == CODEC_ID_AMR_NB)
155 /* max_header_toc_size + the largest AMR payload must fit */
156 if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
157 av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
160 if (st->codec->channels != 1) {
161 av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
168 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
169 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
178 /* send an rtcp sender report packet */
179 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
181 RTPMuxContext *s = s1->priv_data;
184 dprintf(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
186 s->last_rtcp_ntp_time = ntp_time;
187 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
188 s1->streams[0]->time_base) + s->base_timestamp;
189 put_byte(s1->pb, (RTP_VERSION << 6));
190 put_byte(s1->pb, 200);
191 put_be16(s1->pb, 6); /* length in words - 1 */
192 put_be32(s1->pb, s->ssrc);
193 put_be32(s1->pb, ntp_time / 1000000);
194 put_be32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
195 put_be32(s1->pb, rtp_ts);
196 put_be32(s1->pb, s->packet_count);
197 put_be32(s1->pb, s->octet_count);
198 put_flush_packet(s1->pb);
201 /* send an rtp packet. sequence number is incremented, but the caller
202 must update the timestamp itself */
203 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
205 RTPMuxContext *s = s1->priv_data;
207 dprintf(s1, "rtp_send_data size=%d\n", len);
209 /* build the RTP header */
210 put_byte(s1->pb, (RTP_VERSION << 6));
211 put_byte(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
212 put_be16(s1->pb, s->seq);
213 put_be32(s1->pb, s->timestamp);
214 put_be32(s1->pb, s->ssrc);
216 put_buffer(s1->pb, buf1, len);
217 put_flush_packet(s1->pb);
220 s->octet_count += len;
224 /* send an integer number of samples and compute time stamp and fill
225 the rtp send buffer before sending. */
226 static void rtp_send_samples(AVFormatContext *s1,
227 const uint8_t *buf1, int size, int sample_size)
229 RTPMuxContext *s = s1->priv_data;
230 int len, max_packet_size, n;
232 max_packet_size = (s->max_payload_size / sample_size) * sample_size;
233 /* not needed, but who nows */
234 if ((size % sample_size) != 0)
239 len = FFMIN(max_packet_size, size);
242 memcpy(s->buf_ptr, buf1, len);
246 s->timestamp = s->cur_timestamp + n / sample_size;
247 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
248 n += (s->buf_ptr - s->buf);
252 static void rtp_send_mpegaudio(AVFormatContext *s1,
253 const uint8_t *buf1, int size)
255 RTPMuxContext *s = s1->priv_data;
256 int len, count, max_packet_size;
258 max_packet_size = s->max_payload_size;
260 /* test if we must flush because not enough space */
261 len = (s->buf_ptr - s->buf);
262 if ((len + size) > max_packet_size) {
264 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
265 s->buf_ptr = s->buf + 4;
268 if (s->buf_ptr == s->buf + 4) {
269 s->timestamp = s->cur_timestamp;
273 if (size > max_packet_size) {
274 /* big packet: fragment */
277 len = max_packet_size - 4;
280 /* build fragmented packet */
283 s->buf[2] = count >> 8;
285 memcpy(s->buf + 4, buf1, len);
286 ff_rtp_send_data(s1, s->buf, len + 4, 0);
292 if (s->buf_ptr == s->buf + 4) {
293 /* no fragmentation possible */
299 memcpy(s->buf_ptr, buf1, size);
304 static void rtp_send_raw(AVFormatContext *s1,
305 const uint8_t *buf1, int size)
307 RTPMuxContext *s = s1->priv_data;
308 int len, max_packet_size;
310 max_packet_size = s->max_payload_size;
313 len = max_packet_size;
317 s->timestamp = s->cur_timestamp;
318 ff_rtp_send_data(s1, buf1, len, (len == size));
325 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
326 static void rtp_send_mpegts_raw(AVFormatContext *s1,
327 const uint8_t *buf1, int size)
329 RTPMuxContext *s = s1->priv_data;
332 while (size >= TS_PACKET_SIZE) {
333 len = s->max_payload_size - (s->buf_ptr - s->buf);
336 memcpy(s->buf_ptr, buf1, len);
341 out_len = s->buf_ptr - s->buf;
342 if (out_len >= s->max_payload_size) {
343 ff_rtp_send_data(s1, s->buf, out_len, 0);
349 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
351 RTPMuxContext *s = s1->priv_data;
352 AVStream *st = s1->streams[0];
356 dprintf(s1, "%d: write len=%d\n", pkt->stream_index, size);
358 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
360 if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
361 (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) {
362 rtcp_send_sr(s1, ff_ntp_time());
363 s->last_octet_count = s->octet_count;
366 s->cur_timestamp = s->base_timestamp + pkt->pts;
368 switch(st->codec->codec_id) {
369 case CODEC_ID_PCM_MULAW:
370 case CODEC_ID_PCM_ALAW:
371 case CODEC_ID_PCM_U8:
372 case CODEC_ID_PCM_S8:
373 rtp_send_samples(s1, pkt->data, size, 1 * st->codec->channels);
375 case CODEC_ID_PCM_U16BE:
376 case CODEC_ID_PCM_U16LE:
377 case CODEC_ID_PCM_S16BE:
378 case CODEC_ID_PCM_S16LE:
379 rtp_send_samples(s1, pkt->data, size, 2 * st->codec->channels);
383 rtp_send_mpegaudio(s1, pkt->data, size);
385 case CODEC_ID_MPEG1VIDEO:
386 case CODEC_ID_MPEG2VIDEO:
387 ff_rtp_send_mpegvideo(s1, pkt->data, size);
390 ff_rtp_send_aac(s1, pkt->data, size);
392 case CODEC_ID_AMR_NB:
393 case CODEC_ID_AMR_WB:
394 ff_rtp_send_amr(s1, pkt->data, size);
396 case CODEC_ID_MPEG2TS:
397 rtp_send_mpegts_raw(s1, pkt->data, size);
400 ff_rtp_send_h264(s1, pkt->data, size);
404 ff_rtp_send_h263(s1, pkt->data, size);
406 case CODEC_ID_VORBIS:
407 case CODEC_ID_THEORA:
408 ff_rtp_send_xiph(s1, pkt->data, size);
411 /* better than nothing : send the codec raw data */
412 rtp_send_raw(s1, pkt->data, size);
418 static int rtp_write_trailer(AVFormatContext *s1)
420 RTPMuxContext *s = s1->priv_data;
427 AVOutputFormat rtp_muxer = {
429 NULL_IF_CONFIG_SMALL("RTP output format"),
432 sizeof(RTPMuxContext),