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 rtp_write_header(AVFormatContext *s1)
44 RTPMuxContext *s = s1->priv_data;
45 int payload_type, max_packet_size, n;
48 if (s1->nb_streams != 1)
52 payload_type = ff_rtp_get_payload_type(st->codec);
54 payload_type = RTP_PT_PRIVATE; /* private payload type */
55 s->payload_type = payload_type;
57 // following 2 FIXMEs could be set based on the current time, there is normally no info leak, as RTP will likely be transmitted immediately
58 s->base_timestamp = 0; /* FIXME: was random(), what should this be? */
59 s->timestamp = s->base_timestamp;
61 s->ssrc = 0; /* FIXME: was random(), what should this be? */
63 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
65 max_packet_size = url_fget_max_packet_size(s1->pb);
66 if (max_packet_size <= 12)
68 s->buf = av_malloc(max_packet_size);
70 return AVERROR(ENOMEM);
72 s->max_payload_size = max_packet_size - 12;
74 s->max_frames_per_packet = 0;
76 if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
77 if (st->codec->frame_size == 0) {
78 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
80 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);
83 if (st->codec->codec_type == CODEC_TYPE_VIDEO) {
84 /* FIXME: We should round down here... */
85 s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
89 av_set_pts_info(st, 32, 1, 90000);
90 switch(st->codec->codec_id) {
93 s->buf_ptr = s->buf + 4;
95 case CODEC_ID_MPEG1VIDEO:
96 case CODEC_ID_MPEG2VIDEO:
98 case CODEC_ID_MPEG2TS:
99 n = s->max_payload_size / TS_PACKET_SIZE;
102 s->max_payload_size = n * TS_PACKET_SIZE;
108 if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
109 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
118 /* send an rtcp sender report packet */
119 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
121 RTPMuxContext *s = s1->priv_data;
124 dprintf(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
126 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = ntp_time;
127 s->last_rtcp_ntp_time = ntp_time;
128 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
129 s1->streams[0]->time_base) + s->base_timestamp;
130 put_byte(s1->pb, (RTP_VERSION << 6));
131 put_byte(s1->pb, 200);
132 put_be16(s1->pb, 6); /* length in words - 1 */
133 put_be32(s1->pb, s->ssrc);
134 put_be32(s1->pb, ntp_time / 1000000);
135 put_be32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
136 put_be32(s1->pb, rtp_ts);
137 put_be32(s1->pb, s->packet_count);
138 put_be32(s1->pb, s->octet_count);
139 put_flush_packet(s1->pb);
142 /* send an rtp packet. sequence number is incremented, but the caller
143 must update the timestamp itself */
144 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
146 RTPMuxContext *s = s1->priv_data;
148 dprintf(s1, "rtp_send_data size=%d\n", len);
150 /* build the RTP header */
151 put_byte(s1->pb, (RTP_VERSION << 6));
152 put_byte(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
153 put_be16(s1->pb, s->seq);
154 put_be32(s1->pb, s->timestamp);
155 put_be32(s1->pb, s->ssrc);
157 put_buffer(s1->pb, buf1, len);
158 put_flush_packet(s1->pb);
161 s->octet_count += len;
165 /* send an integer number of samples and compute time stamp and fill
166 the rtp send buffer before sending. */
167 static void rtp_send_samples(AVFormatContext *s1,
168 const uint8_t *buf1, int size, int sample_size)
170 RTPMuxContext *s = s1->priv_data;
171 int len, max_packet_size, n;
173 max_packet_size = (s->max_payload_size / sample_size) * sample_size;
174 /* not needed, but who nows */
175 if ((size % sample_size) != 0)
180 len = FFMIN(max_packet_size, size);
183 memcpy(s->buf_ptr, buf1, len);
187 s->timestamp = s->cur_timestamp + n / sample_size;
188 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
189 n += (s->buf_ptr - s->buf);
193 /* NOTE: we suppose that exactly one frame is given as argument here */
195 static void rtp_send_mpegaudio(AVFormatContext *s1,
196 const uint8_t *buf1, int size)
198 RTPMuxContext *s = s1->priv_data;
199 int len, count, max_packet_size;
201 max_packet_size = s->max_payload_size;
203 /* test if we must flush because not enough space */
204 len = (s->buf_ptr - s->buf);
205 if ((len + size) > max_packet_size) {
207 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
208 s->buf_ptr = s->buf + 4;
211 if (s->buf_ptr == s->buf + 4) {
212 s->timestamp = s->cur_timestamp;
216 if (size > max_packet_size) {
217 /* big packet: fragment */
220 len = max_packet_size - 4;
223 /* build fragmented packet */
226 s->buf[2] = count >> 8;
228 memcpy(s->buf + 4, buf1, len);
229 ff_rtp_send_data(s1, s->buf, len + 4, 0);
235 if (s->buf_ptr == s->buf + 4) {
236 /* no fragmentation possible */
242 memcpy(s->buf_ptr, buf1, size);
247 static void rtp_send_raw(AVFormatContext *s1,
248 const uint8_t *buf1, int size)
250 RTPMuxContext *s = s1->priv_data;
251 int len, max_packet_size;
253 max_packet_size = s->max_payload_size;
256 len = max_packet_size;
260 s->timestamp = s->cur_timestamp;
261 ff_rtp_send_data(s1, buf1, len, (len == size));
268 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
269 static void rtp_send_mpegts_raw(AVFormatContext *s1,
270 const uint8_t *buf1, int size)
272 RTPMuxContext *s = s1->priv_data;
275 while (size >= TS_PACKET_SIZE) {
276 len = s->max_payload_size - (s->buf_ptr - s->buf);
279 memcpy(s->buf_ptr, buf1, len);
284 out_len = s->buf_ptr - s->buf;
285 if (out_len >= s->max_payload_size) {
286 ff_rtp_send_data(s1, s->buf, out_len, 0);
292 /* write an RTP packet. 'buf1' must contain a single specific frame. */
293 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
295 RTPMuxContext *s = s1->priv_data;
296 AVStream *st = s1->streams[0];
299 uint8_t *buf1= pkt->data;
301 dprintf(s1, "%d: write len=%d\n", pkt->stream_index, size);
303 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
305 if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
306 (ntp_time() - s->last_rtcp_ntp_time > 5000000))) {
307 rtcp_send_sr(s1, ntp_time());
308 s->last_octet_count = s->octet_count;
311 s->cur_timestamp = s->base_timestamp + pkt->pts;
313 switch(st->codec->codec_id) {
314 case CODEC_ID_PCM_MULAW:
315 case CODEC_ID_PCM_ALAW:
316 case CODEC_ID_PCM_U8:
317 case CODEC_ID_PCM_S8:
318 rtp_send_samples(s1, buf1, size, 1 * st->codec->channels);
320 case CODEC_ID_PCM_U16BE:
321 case CODEC_ID_PCM_U16LE:
322 case CODEC_ID_PCM_S16BE:
323 case CODEC_ID_PCM_S16LE:
324 rtp_send_samples(s1, buf1, size, 2 * st->codec->channels);
328 rtp_send_mpegaudio(s1, buf1, size);
330 case CODEC_ID_MPEG1VIDEO:
331 case CODEC_ID_MPEG2VIDEO:
332 ff_rtp_send_mpegvideo(s1, buf1, size);
335 ff_rtp_send_aac(s1, buf1, size);
337 case CODEC_ID_MPEG2TS:
338 rtp_send_mpegts_raw(s1, buf1, size);
341 ff_rtp_send_h264(s1, buf1, size);
344 /* better than nothing : send the codec raw data */
345 rtp_send_raw(s1, buf1, size);
351 static int rtp_write_trailer(AVFormatContext *s1)
353 RTPMuxContext *s = s1->priv_data;
360 AVOutputFormat rtp_muxer = {
362 NULL_IF_CONFIG_SMALL("RTP output format"),
365 sizeof(RTPMuxContext),