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"
29 #include "rtp_internal.h"
36 #define RTCP_SR_SIZE 28
37 #define NTP_OFFSET 2208988800ULL
38 #define NTP_OFFSET_US (NTP_OFFSET * 1000000ULL)
40 static uint64_t ntp_time(void)
42 return (av_gettime() / 1000) * 1000 + NTP_OFFSET_US
;
45 static int rtp_write_header(AVFormatContext
*s1
)
47 RTPDemuxContext
*s
= s1
->priv_data
;
48 int payload_type
, max_packet_size
, n
;
51 if (s1
->nb_streams
!= 1)
55 payload_type
= rtp_get_payload_type(st
->codec
);
57 payload_type
= RTP_PT_PRIVATE
; /* private payload type */
58 s
->payload_type
= payload_type
;
60 // following 2 FIXMEs could be set based on the current time, there is normally no info leak, as RTP will likely be transmitted immediately
61 s
->base_timestamp
= 0; /* FIXME: was random(), what should this be? */
62 s
->timestamp
= s
->base_timestamp
;
64 s
->ssrc
= 0; /* FIXME: was random(), what should this be? */
66 s
->first_rtcp_ntp_time
= AV_NOPTS_VALUE
;
68 max_packet_size
= url_fget_max_packet_size(s1
->pb
);
69 if (max_packet_size
<= 12)
71 s
->max_payload_size
= max_packet_size
- 12;
73 s
->max_frames_per_packet
= 0;
75 if (st
->codec
->codec_type
== CODEC_TYPE_AUDIO
) {
76 if (st
->codec
->frame_size
== 0) {
77 av_log(s1
, AV_LOG_ERROR
, "Cannot respect max delay: frame size = 0\n");
79 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
);
82 if (st
->codec
->codec_type
== CODEC_TYPE_VIDEO
) {
83 /* FIXME: We should round down here... */
84 s
->max_frames_per_packet
= av_rescale_q(s1
->max_delay
, (AVRational
){1, 1000000}, st
->codec
->time_base
);
88 av_set_pts_info(st
, 32, 1, 90000);
89 switch(st
->codec
->codec_id
) {
92 s
->buf_ptr
= s
->buf
+ 4;
94 case CODEC_ID_MPEG1VIDEO
:
95 case CODEC_ID_MPEG2VIDEO
:
97 case CODEC_ID_MPEG2TS
:
98 n
= s
->max_payload_size
/ TS_PACKET_SIZE
;
101 s
->max_payload_size
= n
* TS_PACKET_SIZE
;
105 s
->read_buf_index
= 0;
107 if (st
->codec
->codec_type
== CODEC_TYPE_AUDIO
) {
108 av_set_pts_info(st
, 32, 1, st
->codec
->sample_rate
);
117 /* send an rtcp sender report packet */
118 static void rtcp_send_sr(AVFormatContext
*s1
, int64_t ntp_time
)
120 RTPDemuxContext
*s
= s1
->priv_data
;
123 dprintf(s1
, "RTCP: %02x %"PRIx64
" %x\n", s
->payload_type
, ntp_time
, s
->timestamp
);
125 if (s
->first_rtcp_ntp_time
== AV_NOPTS_VALUE
) s
->first_rtcp_ntp_time
= ntp_time
;
126 s
->last_rtcp_ntp_time
= ntp_time
;
127 rtp_ts
= av_rescale_q(ntp_time
- s
->first_rtcp_ntp_time
, (AVRational
){1, 1000000},
128 s1
->streams
[0]->time_base
) + s
->base_timestamp
;
129 put_byte(s1
->pb
, (RTP_VERSION
<< 6));
130 put_byte(s1
->pb
, 200);
131 put_be16(s1
->pb
, 6); /* length in words - 1 */
132 put_be32(s1
->pb
, s
->ssrc
);
133 put_be32(s1
->pb
, ntp_time
/ 1000000);
134 put_be32(s1
->pb
, ((ntp_time
% 1000000) << 32) / 1000000);
135 put_be32(s1
->pb
, rtp_ts
);
136 put_be32(s1
->pb
, s
->packet_count
);
137 put_be32(s1
->pb
, s
->octet_count
);
138 put_flush_packet(s1
->pb
);
141 /* send an rtp packet. sequence number is incremented, but the caller
142 must update the timestamp itself */
143 void ff_rtp_send_data(AVFormatContext
*s1
, const uint8_t *buf1
, int len
, int m
)
145 RTPDemuxContext
*s
= s1
->priv_data
;
147 dprintf(s1
, "rtp_send_data size=%d\n", len
);
149 /* build the RTP header */
150 put_byte(s1
->pb
, (RTP_VERSION
<< 6));
151 put_byte(s1
->pb
, (s
->payload_type
& 0x7f) | ((m
& 0x01) << 7));
152 put_be16(s1
->pb
, s
->seq
);
153 put_be32(s1
->pb
, s
->timestamp
);
154 put_be32(s1
->pb
, s
->ssrc
);
156 put_buffer(s1
->pb
, buf1
, len
);
157 put_flush_packet(s1
->pb
);
160 s
->octet_count
+= len
;
164 /* send an integer number of samples and compute time stamp and fill
165 the rtp send buffer before sending. */
166 static void rtp_send_samples(AVFormatContext
*s1
,
167 const uint8_t *buf1
, int size
, int sample_size
)
169 RTPDemuxContext
*s
= s1
->priv_data
;
170 int len
, max_packet_size
, n
;
172 max_packet_size
= (s
->max_payload_size
/ sample_size
) * sample_size
;
173 /* not needed, but who nows */
174 if ((size
% sample_size
) != 0)
179 len
= FFMIN(max_packet_size
, size
);
182 memcpy(s
->buf_ptr
, buf1
, len
);
186 s
->timestamp
= s
->cur_timestamp
+ n
/ sample_size
;
187 ff_rtp_send_data(s1
, s
->buf
, s
->buf_ptr
- s
->buf
, 0);
188 n
+= (s
->buf_ptr
- s
->buf
);
192 /* NOTE: we suppose that exactly one frame is given as argument here */
194 static void rtp_send_mpegaudio(AVFormatContext
*s1
,
195 const uint8_t *buf1
, int size
)
197 RTPDemuxContext
*s
= s1
->priv_data
;
198 int len
, count
, max_packet_size
;
200 max_packet_size
= s
->max_payload_size
;
202 /* test if we must flush because not enough space */
203 len
= (s
->buf_ptr
- s
->buf
);
204 if ((len
+ size
) > max_packet_size
) {
206 ff_rtp_send_data(s1
, s
->buf
, s
->buf_ptr
- s
->buf
, 0);
207 s
->buf_ptr
= s
->buf
+ 4;
210 if (s
->buf_ptr
== s
->buf
+ 4) {
211 s
->timestamp
= s
->cur_timestamp
;
215 if (size
> max_packet_size
) {
216 /* big packet: fragment */
219 len
= max_packet_size
- 4;
222 /* build fragmented packet */
225 s
->buf
[2] = count
>> 8;
227 memcpy(s
->buf
+ 4, buf1
, len
);
228 ff_rtp_send_data(s1
, s
->buf
, len
+ 4, 0);
234 if (s
->buf_ptr
== s
->buf
+ 4) {
235 /* no fragmentation possible */
241 memcpy(s
->buf_ptr
, buf1
, size
);
246 static void rtp_send_raw(AVFormatContext
*s1
,
247 const uint8_t *buf1
, int size
)
249 RTPDemuxContext
*s
= s1
->priv_data
;
250 int len
, max_packet_size
;
252 max_packet_size
= s
->max_payload_size
;
255 len
= max_packet_size
;
259 s
->timestamp
= s
->cur_timestamp
;
260 ff_rtp_send_data(s1
, buf1
, len
, (len
== size
));
267 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
268 static void rtp_send_mpegts_raw(AVFormatContext
*s1
,
269 const uint8_t *buf1
, int size
)
271 RTPDemuxContext
*s
= s1
->priv_data
;
274 while (size
>= TS_PACKET_SIZE
) {
275 len
= s
->max_payload_size
- (s
->buf_ptr
- s
->buf
);
278 memcpy(s
->buf_ptr
, buf1
, len
);
283 out_len
= s
->buf_ptr
- s
->buf
;
284 if (out_len
>= s
->max_payload_size
) {
285 ff_rtp_send_data(s1
, s
->buf
, out_len
, 0);
291 /* write an RTP packet. 'buf1' must contain a single specific frame. */
292 static int rtp_write_packet(AVFormatContext
*s1
, AVPacket
*pkt
)
294 RTPDemuxContext
*s
= s1
->priv_data
;
295 AVStream
*st
= s1
->streams
[0];
298 uint8_t *buf1
= pkt
->data
;
300 dprintf(s1
, "%d: write len=%d\n", pkt
->stream_index
, size
);
302 rtcp_bytes
= ((s
->octet_count
- s
->last_octet_count
) * RTCP_TX_RATIO_NUM
) /
304 if (s
->first_packet
|| ((rtcp_bytes
>= RTCP_SR_SIZE
) &&
305 (ntp_time() - s
->last_rtcp_ntp_time
> 5000000))) {
306 rtcp_send_sr(s1
, ntp_time());
307 s
->last_octet_count
= s
->octet_count
;
310 s
->cur_timestamp
= s
->base_timestamp
+ pkt
->pts
;
312 switch(st
->codec
->codec_id
) {
313 case CODEC_ID_PCM_MULAW
:
314 case CODEC_ID_PCM_ALAW
:
315 case CODEC_ID_PCM_U8
:
316 case CODEC_ID_PCM_S8
:
317 rtp_send_samples(s1
, buf1
, size
, 1 * st
->codec
->channels
);
319 case CODEC_ID_PCM_U16BE
:
320 case CODEC_ID_PCM_U16LE
:
321 case CODEC_ID_PCM_S16BE
:
322 case CODEC_ID_PCM_S16LE
:
323 rtp_send_samples(s1
, buf1
, size
, 2 * st
->codec
->channels
);
327 rtp_send_mpegaudio(s1
, buf1
, size
);
329 case CODEC_ID_MPEG1VIDEO
:
330 case CODEC_ID_MPEG2VIDEO
:
331 ff_rtp_send_mpegvideo(s1
, buf1
, size
);
334 ff_rtp_send_aac(s1
, buf1
, size
);
336 case CODEC_ID_MPEG2TS
:
337 rtp_send_mpegts_raw(s1
, buf1
, size
);
340 ff_rtp_send_h264(s1
, buf1
, size
);
343 /* better than nothing : send the codec raw data */
344 rtp_send_raw(s1
, buf1
, size
);
350 AVOutputFormat rtp_muxer
= {
352 NULL_IF_CONFIG_SMALL("RTP output format"),
355 sizeof(RTPDemuxContext
),