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 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 s->payload_type = ff_rtp_get_payload_type(st->codec);
87 if (s->payload_type < 0)
88 s->payload_type = RTP_PT_PRIVATE + (st->codec->codec_type == CODEC_TYPE_AUDIO);
90 // following 2 FIXMEs could be set based on the current time, there is normally no info leak, as RTP will likely be transmitted immediately
91 s->base_timestamp = 0; /* FIXME: was random(), what should this be? */
92 s->timestamp = s->base_timestamp;
94 s->ssrc = 0; /* FIXME: was random(), what should this be? */
96 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
98 max_packet_size = url_fget_max_packet_size(s1->pb);
99 if (max_packet_size <= 12)
101 s->buf = av_malloc(max_packet_size);
102 if (s->buf == NULL) {
103 return AVERROR(ENOMEM);
105 s->max_payload_size = max_packet_size - 12;
107 s->max_frames_per_packet = 0;
109 if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
110 if (st->codec->frame_size == 0) {
111 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
113 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);
116 if (st->codec->codec_type == CODEC_TYPE_VIDEO) {
117 /* FIXME: We should round down here... */
118 s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
122 av_set_pts_info(st, 32, 1, 90000);
123 switch(st->codec->codec_id) {
126 s->buf_ptr = s->buf + 4;
128 case CODEC_ID_MPEG1VIDEO:
129 case CODEC_ID_MPEG2VIDEO:
131 case CODEC_ID_MPEG2TS:
132 n = s->max_payload_size / TS_PACKET_SIZE;
135 s->max_payload_size = n * TS_PACKET_SIZE;
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 == CODEC_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 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = ntp_time;
177 s->last_rtcp_ntp_time = ntp_time;
178 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
179 s1->streams[0]->time_base) + s->base_timestamp;
180 put_byte(s1->pb, (RTP_VERSION << 6));
181 put_byte(s1->pb, 200);
182 put_be16(s1->pb, 6); /* length in words - 1 */
183 put_be32(s1->pb, s->ssrc);
184 put_be32(s1->pb, ntp_time / 1000000);
185 put_be32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
186 put_be32(s1->pb, rtp_ts);
187 put_be32(s1->pb, s->packet_count);
188 put_be32(s1->pb, s->octet_count);
189 put_flush_packet(s1->pb);
192 /* send an rtp packet. sequence number is incremented, but the caller
193 must update the timestamp itself */
194 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
196 RTPMuxContext *s = s1->priv_data;
198 dprintf(s1, "rtp_send_data size=%d\n", len);
200 /* build the RTP header */
201 put_byte(s1->pb, (RTP_VERSION << 6));
202 put_byte(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
203 put_be16(s1->pb, s->seq);
204 put_be32(s1->pb, s->timestamp);
205 put_be32(s1->pb, s->ssrc);
207 put_buffer(s1->pb, buf1, len);
208 put_flush_packet(s1->pb);
211 s->octet_count += len;
215 /* send an integer number of samples and compute time stamp and fill
216 the rtp send buffer before sending. */
217 static void rtp_send_samples(AVFormatContext *s1,
218 const uint8_t *buf1, int size, int sample_size)
220 RTPMuxContext *s = s1->priv_data;
221 int len, max_packet_size, n;
223 max_packet_size = (s->max_payload_size / sample_size) * sample_size;
224 /* not needed, but who nows */
225 if ((size % sample_size) != 0)
230 len = FFMIN(max_packet_size, size);
233 memcpy(s->buf_ptr, buf1, len);
237 s->timestamp = s->cur_timestamp + n / sample_size;
238 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
239 n += (s->buf_ptr - s->buf);
243 static void rtp_send_mpegaudio(AVFormatContext *s1,
244 const uint8_t *buf1, int size)
246 RTPMuxContext *s = s1->priv_data;
247 int len, count, max_packet_size;
249 max_packet_size = s->max_payload_size;
251 /* test if we must flush because not enough space */
252 len = (s->buf_ptr - s->buf);
253 if ((len + size) > max_packet_size) {
255 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
256 s->buf_ptr = s->buf + 4;
259 if (s->buf_ptr == s->buf + 4) {
260 s->timestamp = s->cur_timestamp;
264 if (size > max_packet_size) {
265 /* big packet: fragment */
268 len = max_packet_size - 4;
271 /* build fragmented packet */
274 s->buf[2] = count >> 8;
276 memcpy(s->buf + 4, buf1, len);
277 ff_rtp_send_data(s1, s->buf, len + 4, 0);
283 if (s->buf_ptr == s->buf + 4) {
284 /* no fragmentation possible */
290 memcpy(s->buf_ptr, buf1, size);
295 static void rtp_send_raw(AVFormatContext *s1,
296 const uint8_t *buf1, int size)
298 RTPMuxContext *s = s1->priv_data;
299 int len, max_packet_size;
301 max_packet_size = s->max_payload_size;
304 len = max_packet_size;
308 s->timestamp = s->cur_timestamp;
309 ff_rtp_send_data(s1, buf1, len, (len == size));
316 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
317 static void rtp_send_mpegts_raw(AVFormatContext *s1,
318 const uint8_t *buf1, int size)
320 RTPMuxContext *s = s1->priv_data;
323 while (size >= TS_PACKET_SIZE) {
324 len = s->max_payload_size - (s->buf_ptr - s->buf);
327 memcpy(s->buf_ptr, buf1, len);
332 out_len = s->buf_ptr - s->buf;
333 if (out_len >= s->max_payload_size) {
334 ff_rtp_send_data(s1, s->buf, out_len, 0);
340 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
342 RTPMuxContext *s = s1->priv_data;
343 AVStream *st = s1->streams[0];
347 dprintf(s1, "%d: write len=%d\n", pkt->stream_index, size);
349 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
351 if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
352 (ntp_time() - s->last_rtcp_ntp_time > 5000000))) {
353 rtcp_send_sr(s1, ntp_time());
354 s->last_octet_count = s->octet_count;
357 s->cur_timestamp = s->base_timestamp + pkt->pts;
359 switch(st->codec->codec_id) {
360 case CODEC_ID_PCM_MULAW:
361 case CODEC_ID_PCM_ALAW:
362 case CODEC_ID_PCM_U8:
363 case CODEC_ID_PCM_S8:
364 rtp_send_samples(s1, pkt->data, size, 1 * st->codec->channels);
366 case CODEC_ID_PCM_U16BE:
367 case CODEC_ID_PCM_U16LE:
368 case CODEC_ID_PCM_S16BE:
369 case CODEC_ID_PCM_S16LE:
370 rtp_send_samples(s1, pkt->data, size, 2 * st->codec->channels);
374 rtp_send_mpegaudio(s1, pkt->data, size);
376 case CODEC_ID_MPEG1VIDEO:
377 case CODEC_ID_MPEG2VIDEO:
378 ff_rtp_send_mpegvideo(s1, pkt->data, size);
381 ff_rtp_send_aac(s1, pkt->data, size);
383 case CODEC_ID_AMR_NB:
384 case CODEC_ID_AMR_WB:
385 ff_rtp_send_amr(s1, pkt->data, size);
387 case CODEC_ID_MPEG2TS:
388 rtp_send_mpegts_raw(s1, pkt->data, size);
391 ff_rtp_send_h264(s1, pkt->data, size);
395 ff_rtp_send_h263(s1, pkt->data, size);
398 /* better than nothing : send the codec raw data */
399 rtp_send_raw(s1, pkt->data, size);
405 static int rtp_write_trailer(AVFormatContext *s1)
407 RTPMuxContext *s = s1->priv_data;
414 AVOutputFormat rtp_muxer = {
416 NULL_IF_CONFIG_SMALL("RTP output format"),
419 sizeof(RTPMuxContext),