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path: root/src/plugins/ffmpeg/libavformat/rtpenc.c
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Diffstat (limited to 'src/plugins/ffmpeg/libavformat/rtpenc.c')
-rw-r--r--src/plugins/ffmpeg/libavformat/rtpenc.c360
1 files changed, 0 insertions, 360 deletions
diff --git a/src/plugins/ffmpeg/libavformat/rtpenc.c b/src/plugins/ffmpeg/libavformat/rtpenc.c
deleted file mode 100644
index 2317f5c..0000000
--- a/src/plugins/ffmpeg/libavformat/rtpenc.c
+++ /dev/null
@@ -1,360 +0,0 @@
1/*
2 * RTP output format
3 * Copyright (c) 2002 Fabrice Bellard.
4 *
5 * This file is part of FFmpeg.
6 *
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.
11 *
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.
16 *
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
20 */
21
22#include "libavcodec/bitstream.h"
23#include "avformat.h"
24#include "mpegts.h"
25
26#include <unistd.h>
27#include "network.h"
28
29#include "rtp_internal.h"
30#include "rtp_mpv.h"
31#include "rtp_aac.h"
32#include "rtp_h264.h"
33
34//#define DEBUG
35
36#define RTCP_SR_SIZE 28
37#define NTP_OFFSET 2208988800ULL
38#define NTP_OFFSET_US (NTP_OFFSET * 1000000ULL)
39
40static uint64_t ntp_time(void)
41{
42 return (av_gettime() / 1000) * 1000 + NTP_OFFSET_US;
43}
44
45static int rtp_write_header(AVFormatContext *s1)
46{
47 RTPDemuxContext *s = s1->priv_data;
48 int payload_type, max_packet_size, n;
49 AVStream *st;
50
51 if (s1->nb_streams != 1)
52 return -1;
53 st = s1->streams[0];
54
55 payload_type = rtp_get_payload_type(st->codec);
56 if (payload_type < 0)
57 payload_type = RTP_PT_PRIVATE; /* private payload type */
58 s->payload_type = payload_type;
59
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;
63 s->cur_timestamp = 0;
64 s->ssrc = 0; /* FIXME: was random(), what should this be? */
65 s->first_packet = 1;
66 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
67
68 max_packet_size = url_fget_max_packet_size(s1->pb);
69 if (max_packet_size <= 12)
70 return AVERROR(EIO);
71 s->max_payload_size = max_packet_size - 12;
72
73 s->max_frames_per_packet = 0;
74 if (s1->max_delay) {
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");
78 } else {
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);
80 }
81 }
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);
85 }
86 }
87
88 av_set_pts_info(st, 32, 1, 90000);
89 switch(st->codec->codec_id) {
90 case CODEC_ID_MP2:
91 case CODEC_ID_MP3:
92 s->buf_ptr = s->buf + 4;
93 break;
94 case CODEC_ID_MPEG1VIDEO:
95 case CODEC_ID_MPEG2VIDEO:
96 break;
97 case CODEC_ID_MPEG2TS:
98 n = s->max_payload_size / TS_PACKET_SIZE;
99 if (n < 1)
100 n = 1;
101 s->max_payload_size = n * TS_PACKET_SIZE;
102 s->buf_ptr = s->buf;
103 break;
104 case CODEC_ID_AAC:
105 s->read_buf_index = 0;
106 default:
107 if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
108 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
109 }
110 s->buf_ptr = s->buf;
111 break;
112 }
113
114 return 0;
115}
116
117/* send an rtcp sender report packet */
118static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
119{
120 RTPDemuxContext *s = s1->priv_data;
121 uint32_t rtp_ts;
122
123 dprintf(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
124
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);
139}
140
141/* send an rtp packet. sequence number is incremented, but the caller
142 must update the timestamp itself */
143void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
144{
145 RTPDemuxContext *s = s1->priv_data;
146
147 dprintf(s1, "rtp_send_data size=%d\n", len);
148
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);
155
156 put_buffer(s1->pb, buf1, len);
157 put_flush_packet(s1->pb);
158
159 s->seq++;
160 s->octet_count += len;
161 s->packet_count++;
162}
163
164/* send an integer number of samples and compute time stamp and fill
165 the rtp send buffer before sending. */
166static void rtp_send_samples(AVFormatContext *s1,
167 const uint8_t *buf1, int size, int sample_size)
168{
169 RTPDemuxContext *s = s1->priv_data;
170 int len, max_packet_size, n;
171
172 max_packet_size = (s->max_payload_size / sample_size) * sample_size;
173 /* not needed, but who nows */
174 if ((size % sample_size) != 0)
175 av_abort();
176 n = 0;
177 while (size > 0) {
178 s->buf_ptr = s->buf;
179 len = FFMIN(max_packet_size, size);
180
181 /* copy data */
182 memcpy(s->buf_ptr, buf1, len);
183 s->buf_ptr += len;
184 buf1 += len;
185 size -= 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);
189 }
190}
191
192/* NOTE: we suppose that exactly one frame is given as argument here */
193/* XXX: test it */
194static void rtp_send_mpegaudio(AVFormatContext *s1,
195 const uint8_t *buf1, int size)
196{
197 RTPDemuxContext *s = s1->priv_data;
198 int len, count, max_packet_size;
199
200 max_packet_size = s->max_payload_size;
201
202 /* test if we must flush because not enough space */
203 len = (s->buf_ptr - s->buf);
204 if ((len + size) > max_packet_size) {
205 if (len > 4) {
206 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
207 s->buf_ptr = s->buf + 4;
208 }
209 }
210 if (s->buf_ptr == s->buf + 4) {
211 s->timestamp = s->cur_timestamp;
212 }
213
214 /* add the packet */
215 if (size > max_packet_size) {
216 /* big packet: fragment */
217 count = 0;
218 while (size > 0) {
219 len = max_packet_size - 4;
220 if (len > size)
221 len = size;
222 /* build fragmented packet */
223 s->buf[0] = 0;
224 s->buf[1] = 0;
225 s->buf[2] = count >> 8;
226 s->buf[3] = count;
227 memcpy(s->buf + 4, buf1, len);
228 ff_rtp_send_data(s1, s->buf, len + 4, 0);
229 size -= len;
230 buf1 += len;
231 count += len;
232 }
233 } else {
234 if (s->buf_ptr == s->buf + 4) {
235 /* no fragmentation possible */
236 s->buf[0] = 0;
237 s->buf[1] = 0;
238 s->buf[2] = 0;
239 s->buf[3] = 0;
240 }
241 memcpy(s->buf_ptr, buf1, size);
242 s->buf_ptr += size;
243 }
244}
245
246static void rtp_send_raw(AVFormatContext *s1,
247 const uint8_t *buf1, int size)
248{
249 RTPDemuxContext *s = s1->priv_data;
250 int len, max_packet_size;
251
252 max_packet_size = s->max_payload_size;
253
254 while (size > 0) {
255 len = max_packet_size;
256 if (len > size)
257 len = size;
258
259 s->timestamp = s->cur_timestamp;
260 ff_rtp_send_data(s1, buf1, len, (len == size));
261
262 buf1 += len;
263 size -= len;
264 }
265}
266
267/* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
268static void rtp_send_mpegts_raw(AVFormatContext *s1,
269 const uint8_t *buf1, int size)
270{
271 RTPDemuxContext *s = s1->priv_data;
272 int len, out_len;
273
274 while (size >= TS_PACKET_SIZE) {
275 len = s->max_payload_size - (s->buf_ptr - s->buf);
276 if (len > size)
277 len = size;
278 memcpy(s->buf_ptr, buf1, len);
279 buf1 += len;
280 size -= len;
281 s->buf_ptr += len;
282
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);
286 s->buf_ptr = s->buf;
287 }
288 }
289}
290
291/* write an RTP packet. 'buf1' must contain a single specific frame. */
292static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
293{
294 RTPDemuxContext *s = s1->priv_data;
295 AVStream *st = s1->streams[0];
296 int rtcp_bytes;
297 int size= pkt->size;
298 uint8_t *buf1= pkt->data;
299
300 dprintf(s1, "%d: write len=%d\n", pkt->stream_index, size);
301
302 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
303 RTCP_TX_RATIO_DEN;
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;
308 s->first_packet = 0;
309 }
310 s->cur_timestamp = s->base_timestamp + pkt->pts;
311
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);
318 break;
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);
324 break;
325 case CODEC_ID_MP2:
326 case CODEC_ID_MP3:
327 rtp_send_mpegaudio(s1, buf1, size);
328 break;
329 case CODEC_ID_MPEG1VIDEO:
330 case CODEC_ID_MPEG2VIDEO:
331 ff_rtp_send_mpegvideo(s1, buf1, size);
332 break;
333 case CODEC_ID_AAC:
334 ff_rtp_send_aac(s1, buf1, size);
335 break;
336 case CODEC_ID_MPEG2TS:
337 rtp_send_mpegts_raw(s1, buf1, size);
338 break;
339 case CODEC_ID_H264:
340 ff_rtp_send_h264(s1, buf1, size);
341 break;
342 default:
343 /* better than nothing : send the codec raw data */
344 rtp_send_raw(s1, buf1, size);
345 break;
346 }
347 return 0;
348}
349
350AVOutputFormat rtp_muxer = {
351 "rtp",
352 NULL_IF_CONFIG_SMALL("RTP output format"),
353 NULL,
354 NULL,
355 sizeof(RTPDemuxContext),
356 CODEC_ID_PCM_MULAW,
357 CODEC_ID_NONE,
358 rtp_write_header,
359 rtp_write_packet,
360};
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