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:
62 static int rtp_write_header(AVFormatContext *s1)
64 RTPMuxContext *s = s1->priv_data;
65 int max_packet_size, n;
68 if (s1->nb_streams != 1)
71 if (!is_supported(st->codec->codec_id)) {
72 av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", st->codec->codec_id);
77 s->payload_type = ff_rtp_get_payload_type(st->codec);
78 if (s->payload_type < 0)
79 s->payload_type = RTP_PT_PRIVATE + (st->codec->codec_type == AVMEDIA_TYPE_AUDIO);
81 s->base_timestamp = av_get_random_seed();
82 s->timestamp = s->base_timestamp;
84 s->ssrc = av_get_random_seed();
86 s->first_rtcp_ntp_time = ff_ntp_time();
87 if (s1->start_time_realtime)
88 /* Round the NTP time to whole milliseconds. */
89 s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
92 max_packet_size = url_fget_max_packet_size(s1->pb);
93 if (max_packet_size <= 12)
95 s->buf = av_malloc(max_packet_size);
97 return AVERROR(ENOMEM);
99 s->max_payload_size = max_packet_size - 12;
101 s->max_frames_per_packet = 0;
103 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
104 if (st->codec->frame_size == 0) {
105 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
107 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);
110 if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
111 /* FIXME: We should round down here... */
112 s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
116 av_set_pts_info(st, 32, 1, 90000);
117 switch(st->codec->codec_id) {
120 s->buf_ptr = s->buf + 4;
122 case CODEC_ID_MPEG1VIDEO:
123 case CODEC_ID_MPEG2VIDEO:
125 case CODEC_ID_MPEG2TS:
126 n = s->max_payload_size / TS_PACKET_SIZE;
129 s->max_payload_size = n * TS_PACKET_SIZE;
133 /* check for H.264 MP4 syntax */
134 if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
135 s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
138 case CODEC_ID_AMR_NB:
139 case CODEC_ID_AMR_WB:
140 if (!s->max_frames_per_packet)
141 s->max_frames_per_packet = 12;
142 if (st->codec->codec_id == CODEC_ID_AMR_NB)
146 /* max_header_toc_size + the largest AMR payload must fit */
147 if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
148 av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
151 if (st->codec->channels != 1) {
152 av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
158 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
159 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
168 /* send an rtcp sender report packet */
169 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
171 RTPMuxContext *s = s1->priv_data;
174 dprintf(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
176 s->last_rtcp_ntp_time = ntp_time;
177 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
178 s1->streams[0]->time_base) + s->base_timestamp;
179 put_byte(s1->pb, (RTP_VERSION << 6));
180 put_byte(s1->pb, 200);
181 put_be16(s1->pb, 6); /* length in words - 1 */
182 put_be32(s1->pb, s->ssrc);
183 put_be32(s1->pb, ntp_time / 1000000);
184 put_be32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
185 put_be32(s1->pb, rtp_ts);
186 put_be32(s1->pb, s->packet_count);
187 put_be32(s1->pb, s->octet_count);
188 put_flush_packet(s1->pb);
191 /* send an rtp packet. sequence number is incremented, but the caller
192 must update the timestamp itself */
193 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
195 RTPMuxContext *s = s1->priv_data;
197 dprintf(s1, "rtp_send_data size=%d\n", len);
199 /* build the RTP header */
200 put_byte(s1->pb, (RTP_VERSION << 6));
201 put_byte(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
202 put_be16(s1->pb, s->seq);
203 put_be32(s1->pb, s->timestamp);
204 put_be32(s1->pb, s->ssrc);
206 put_buffer(s1->pb, buf1, len);
207 put_flush_packet(s1->pb);
210 s->octet_count += len;
214 /* send an integer number of samples and compute time stamp and fill
215 the rtp send buffer before sending. */
216 static void rtp_send_samples(AVFormatContext *s1,
217 const uint8_t *buf1, int size, int sample_size)
219 RTPMuxContext *s = s1->priv_data;
220 int len, max_packet_size, n;
222 max_packet_size = (s->max_payload_size / sample_size) * sample_size;
223 /* not needed, but who nows */
224 if ((size % sample_size) != 0)
229 len = FFMIN(max_packet_size, size);
232 memcpy(s->buf_ptr, buf1, len);
236 s->timestamp = s->cur_timestamp + n / sample_size;
237 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
238 n += (s->buf_ptr - s->buf);
242 static void rtp_send_mpegaudio(AVFormatContext *s1,
243 const uint8_t *buf1, int size)
245 RTPMuxContext *s = s1->priv_data;
246 int len, count, max_packet_size;
248 max_packet_size = s->max_payload_size;
250 /* test if we must flush because not enough space */
251 len = (s->buf_ptr - s->buf);
252 if ((len + size) > max_packet_size) {
254 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
255 s->buf_ptr = s->buf + 4;
258 if (s->buf_ptr == s->buf + 4) {
259 s->timestamp = s->cur_timestamp;
263 if (size > max_packet_size) {
264 /* big packet: fragment */
267 len = max_packet_size - 4;
270 /* build fragmented packet */
273 s->buf[2] = count >> 8;
275 memcpy(s->buf + 4, buf1, len);
276 ff_rtp_send_data(s1, s->buf, len + 4, 0);
282 if (s->buf_ptr == s->buf + 4) {
283 /* no fragmentation possible */
289 memcpy(s->buf_ptr, buf1, size);
294 static void rtp_send_raw(AVFormatContext *s1,
295 const uint8_t *buf1, int size)
297 RTPMuxContext *s = s1->priv_data;
298 int len, max_packet_size;
300 max_packet_size = s->max_payload_size;
303 len = max_packet_size;
307 s->timestamp = s->cur_timestamp;
308 ff_rtp_send_data(s1, buf1, len, (len == size));
315 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
316 static void rtp_send_mpegts_raw(AVFormatContext *s1,
317 const uint8_t *buf1, int size)
319 RTPMuxContext *s = s1->priv_data;
322 while (size >= TS_PACKET_SIZE) {
323 len = s->max_payload_size - (s->buf_ptr - s->buf);
326 memcpy(s->buf_ptr, buf1, len);
331 out_len = s->buf_ptr - s->buf;
332 if (out_len >= s->max_payload_size) {
333 ff_rtp_send_data(s1, s->buf, out_len, 0);
339 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
341 RTPMuxContext *s = s1->priv_data;
342 AVStream *st = s1->streams[0];
346 dprintf(s1, "%d: write len=%d\n", pkt->stream_index, size);
348 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
350 if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
351 (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) {
352 rtcp_send_sr(s1, ff_ntp_time());
353 s->last_octet_count = s->octet_count;
356 s->cur_timestamp = s->base_timestamp + pkt->pts;
358 switch(st->codec->codec_id) {
359 case CODEC_ID_PCM_MULAW:
360 case CODEC_ID_PCM_ALAW:
361 case CODEC_ID_PCM_U8:
362 case CODEC_ID_PCM_S8:
363 rtp_send_samples(s1, pkt->data, size, 1 * st->codec->channels);
365 case CODEC_ID_PCM_U16BE:
366 case CODEC_ID_PCM_U16LE:
367 case CODEC_ID_PCM_S16BE:
368 case CODEC_ID_PCM_S16LE:
369 rtp_send_samples(s1, pkt->data, size, 2 * st->codec->channels);
373 rtp_send_mpegaudio(s1, pkt->data, size);
375 case CODEC_ID_MPEG1VIDEO:
376 case CODEC_ID_MPEG2VIDEO:
377 ff_rtp_send_mpegvideo(s1, pkt->data, size);
380 ff_rtp_send_aac(s1, pkt->data, size);
382 case CODEC_ID_AMR_NB:
383 case CODEC_ID_AMR_WB:
384 ff_rtp_send_amr(s1, pkt->data, size);
386 case CODEC_ID_MPEG2TS:
387 rtp_send_mpegts_raw(s1, pkt->data, size);
390 ff_rtp_send_h264(s1, pkt->data, size);
394 ff_rtp_send_h263(s1, pkt->data, size);
397 /* better than nothing : send the codec raw data */
398 rtp_send_raw(s1, pkt->data, size);
404 static int rtp_write_trailer(AVFormatContext *s1)
406 RTPMuxContext *s = s1->priv_data;
413 AVOutputFormat rtp_muxer = {
415 NULL_IF_CONFIG_SMALL("RTP output format"),
418 sizeof(RTPMuxContext),