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/get_bits.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 is_supported(enum CodecID id)
48 case CODEC_ID_MPEG1VIDEO:
49 case CODEC_ID_MPEG2VIDEO:
54 case CODEC_ID_PCM_ALAW:
55 case CODEC_ID_PCM_MULAW:
57 case CODEC_ID_PCM_S16BE:
58 case CODEC_ID_PCM_S16LE:
59 case CODEC_ID_PCM_U16BE:
60 case CODEC_ID_PCM_U16LE:
62 case CODEC_ID_MPEG2TS:
71 static int rtp_write_header(AVFormatContext *s1)
73 RTPMuxContext *s = s1->priv_data;
74 int payload_type, max_packet_size, n;
77 if (s1->nb_streams != 1)
80 if (!is_supported(st->codec->codec_id)) {
81 av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", st->codec->codec_id);
86 payload_type = ff_rtp_get_payload_type(st->codec);
88 payload_type = RTP_PT_PRIVATE; /* private payload type */
89 s->payload_type = payload_type;
91 // following 2 FIXMEs could be set based on the current time, there is normally no info leak, as RTP will likely be transmitted immediately
92 s->base_timestamp = 0; /* FIXME: was random(), what should this be? */
93 s->timestamp = s->base_timestamp;
95 s->ssrc = 0; /* FIXME: was random(), what should this be? */
97 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
99 max_packet_size = url_fget_max_packet_size(s1->pb);
100 if (max_packet_size <= 12)
102 s->buf = av_malloc(max_packet_size);
103 if (s->buf == NULL) {
104 return AVERROR(ENOMEM);
106 s->max_payload_size = max_packet_size - 12;
108 s->max_frames_per_packet = 0;
110 if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
111 if (st->codec->frame_size == 0) {
112 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
114 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);
117 if (st->codec->codec_type == CODEC_TYPE_VIDEO) {
118 /* FIXME: We should round down here... */
119 s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
123 av_set_pts_info(st, 32, 1, 90000);
124 switch(st->codec->codec_id) {
127 s->buf_ptr = s->buf + 4;
129 case CODEC_ID_MPEG1VIDEO:
130 case CODEC_ID_MPEG2VIDEO:
132 case CODEC_ID_MPEG2TS:
133 n = s->max_payload_size / TS_PACKET_SIZE;
136 s->max_payload_size = n * TS_PACKET_SIZE;
139 case CODEC_ID_AMR_NB:
140 case CODEC_ID_AMR_WB:
141 if (!s->max_frames_per_packet)
142 s->max_frames_per_packet = 12;
143 if (st->codec->codec_id == CODEC_ID_AMR_NB)
147 /* max_header_toc_size + the largest AMR payload must fit */
148 if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
149 av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
152 if (st->codec->channels != 1) {
153 av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
159 if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
160 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
169 /* send an rtcp sender report packet */
170 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
172 RTPMuxContext *s = s1->priv_data;
175 dprintf(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
177 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = ntp_time;
178 s->last_rtcp_ntp_time = ntp_time;
179 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
180 s1->streams[0]->time_base) + s->base_timestamp;
181 put_byte(s1->pb, (RTP_VERSION << 6));
182 put_byte(s1->pb, 200);
183 put_be16(s1->pb, 6); /* length in words - 1 */
184 put_be32(s1->pb, s->ssrc);
185 put_be32(s1->pb, ntp_time / 1000000);
186 put_be32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
187 put_be32(s1->pb, rtp_ts);
188 put_be32(s1->pb, s->packet_count);
189 put_be32(s1->pb, s->octet_count);
190 put_flush_packet(s1->pb);
193 /* send an rtp packet. sequence number is incremented, but the caller
194 must update the timestamp itself */
195 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
197 RTPMuxContext *s = s1->priv_data;
199 dprintf(s1, "rtp_send_data size=%d\n", len);
201 /* build the RTP header */
202 put_byte(s1->pb, (RTP_VERSION << 6));
203 put_byte(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
204 put_be16(s1->pb, s->seq);
205 put_be32(s1->pb, s->timestamp);
206 put_be32(s1->pb, s->ssrc);
208 put_buffer(s1->pb, buf1, len);
209 put_flush_packet(s1->pb);
212 s->octet_count += len;
216 /* send an integer number of samples and compute time stamp and fill
217 the rtp send buffer before sending. */
218 static void rtp_send_samples(AVFormatContext *s1,
219 const uint8_t *buf1, int size, int sample_size)
221 RTPMuxContext *s = s1->priv_data;
222 int len, max_packet_size, n;
224 max_packet_size = (s->max_payload_size / sample_size) * sample_size;
225 /* not needed, but who nows */
226 if ((size % sample_size) != 0)
231 len = FFMIN(max_packet_size, size);
234 memcpy(s->buf_ptr, buf1, len);
238 s->timestamp = s->cur_timestamp + n / sample_size;
239 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
240 n += (s->buf_ptr - s->buf);
244 /* NOTE: we suppose that exactly one frame is given as argument here */
246 static void rtp_send_mpegaudio(AVFormatContext *s1,
247 const uint8_t *buf1, int size)
249 RTPMuxContext *s = s1->priv_data;
250 int len, count, max_packet_size;
252 max_packet_size = s->max_payload_size;
254 /* test if we must flush because not enough space */
255 len = (s->buf_ptr - s->buf);
256 if ((len + size) > max_packet_size) {
258 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
259 s->buf_ptr = s->buf + 4;
262 if (s->buf_ptr == s->buf + 4) {
263 s->timestamp = s->cur_timestamp;
267 if (size > max_packet_size) {
268 /* big packet: fragment */
271 len = max_packet_size - 4;
274 /* build fragmented packet */
277 s->buf[2] = count >> 8;
279 memcpy(s->buf + 4, buf1, len);
280 ff_rtp_send_data(s1, s->buf, len + 4, 0);
286 if (s->buf_ptr == s->buf + 4) {
287 /* no fragmentation possible */
293 memcpy(s->buf_ptr, buf1, size);
298 static void rtp_send_raw(AVFormatContext *s1,
299 const uint8_t *buf1, int size)
301 RTPMuxContext *s = s1->priv_data;
302 int len, max_packet_size;
304 max_packet_size = s->max_payload_size;
307 len = max_packet_size;
311 s->timestamp = s->cur_timestamp;
312 ff_rtp_send_data(s1, buf1, len, (len == size));
319 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
320 static void rtp_send_mpegts_raw(AVFormatContext *s1,
321 const uint8_t *buf1, int size)
323 RTPMuxContext *s = s1->priv_data;
326 while (size >= TS_PACKET_SIZE) {
327 len = s->max_payload_size - (s->buf_ptr - s->buf);
330 memcpy(s->buf_ptr, buf1, len);
335 out_len = s->buf_ptr - s->buf;
336 if (out_len >= s->max_payload_size) {
337 ff_rtp_send_data(s1, s->buf, out_len, 0);
343 /* write an RTP packet. 'buf1' must contain a single specific frame. */
344 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
346 RTPMuxContext *s = s1->priv_data;
347 AVStream *st = s1->streams[0];
350 uint8_t *buf1= pkt->data;
352 dprintf(s1, "%d: write len=%d\n", pkt->stream_index, size);
354 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
356 if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
357 (ntp_time() - s->last_rtcp_ntp_time > 5000000))) {
358 rtcp_send_sr(s1, ntp_time());
359 s->last_octet_count = s->octet_count;
362 s->cur_timestamp = s->base_timestamp + pkt->pts;
364 switch(st->codec->codec_id) {
365 case CODEC_ID_PCM_MULAW:
366 case CODEC_ID_PCM_ALAW:
367 case CODEC_ID_PCM_U8:
368 case CODEC_ID_PCM_S8:
369 rtp_send_samples(s1, buf1, size, 1 * st->codec->channels);
371 case CODEC_ID_PCM_U16BE:
372 case CODEC_ID_PCM_U16LE:
373 case CODEC_ID_PCM_S16BE:
374 case CODEC_ID_PCM_S16LE:
375 rtp_send_samples(s1, buf1, size, 2 * st->codec->channels);
379 rtp_send_mpegaudio(s1, buf1, size);
381 case CODEC_ID_MPEG1VIDEO:
382 case CODEC_ID_MPEG2VIDEO:
383 ff_rtp_send_mpegvideo(s1, buf1, size);
386 ff_rtp_send_aac(s1, buf1, size);
388 case CODEC_ID_AMR_NB:
389 case CODEC_ID_AMR_WB:
390 ff_rtp_send_amr(s1, buf1, size);
392 case CODEC_ID_MPEG2TS:
393 rtp_send_mpegts_raw(s1, buf1, size);
396 ff_rtp_send_h264(s1, buf1, size);
400 ff_rtp_send_h263(s1, buf1, size);
403 /* better than nothing : send the codec raw data */
404 rtp_send_raw(s1, buf1, size);
410 static int rtp_write_trailer(AVFormatContext *s1)
412 RTPMuxContext *s = s1->priv_data;
419 AVOutputFormat rtp_muxer = {
421 NULL_IF_CONFIG_SMALL("RTP output format"),
424 sizeof(RTPMuxContext),