summaryrefslogtreecommitdiffstats
path: root/libavformat/rtpenc.c
blob: ce42e3e9aa0035405f52fad4d69d4e33e6b0d6f6 (plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
/*
 * RTP output format
 * Copyright (c) 2002 Fabrice Bellard
 *
 * This file is part of Libav.
 *
 * Libav is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * Libav is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with Libav; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

#include "avformat.h"
#include "mpegts.h"
#include "internal.h"
#include "libavutil/mathematics.h"
#include "libavutil/random_seed.h"
#include "libavutil/opt.h"

#include "rtpenc.h"

//#define DEBUG

static const AVOption options[] = {
    FF_RTP_FLAG_OPTS(RTPMuxContext, flags),
    { "payload_type", "Specify RTP payload type", offsetof(RTPMuxContext, payload_type), FF_OPT_TYPE_INT, {.dbl = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM },
    { NULL },
};

static const AVClass rtp_muxer_class = {
    .class_name = "RTP muxer",
    .item_name  = av_default_item_name,
    .option     = options,
    .version    = LIBAVUTIL_VERSION_INT,
};

#define RTCP_SR_SIZE 28

static int is_supported(enum CodecID id)
{
    switch(id) {
    case CODEC_ID_H263:
    case CODEC_ID_H263P:
    case CODEC_ID_H264:
    case CODEC_ID_MPEG1VIDEO:
    case CODEC_ID_MPEG2VIDEO:
    case CODEC_ID_MPEG4:
    case CODEC_ID_AAC:
    case CODEC_ID_MP2:
    case CODEC_ID_MP3:
    case CODEC_ID_PCM_ALAW:
    case CODEC_ID_PCM_MULAW:
    case CODEC_ID_PCM_S8:
    case CODEC_ID_PCM_S16BE:
    case CODEC_ID_PCM_S16LE:
    case CODEC_ID_PCM_U16BE:
    case CODEC_ID_PCM_U16LE:
    case CODEC_ID_PCM_U8:
    case CODEC_ID_MPEG2TS:
    case CODEC_ID_AMR_NB:
    case CODEC_ID_AMR_WB:
    case CODEC_ID_VORBIS:
    case CODEC_ID_THEORA:
    case CODEC_ID_VP8:
    case CODEC_ID_ADPCM_G722:
        return 1;
    default:
        return 0;
    }
}

static int rtp_write_header(AVFormatContext *s1)
{
    RTPMuxContext *s = s1->priv_data;
    int max_packet_size, n;
    AVStream *st;

    if (s1->nb_streams != 1)
        return -1;
    st = s1->streams[0];
    if (!is_supported(st->codec->codec_id)) {
        av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", st->codec->codec_id);

        return -1;
    }

    if (s->payload_type < 0)
        s->payload_type = ff_rtp_get_payload_type(s1, st->codec);
    s->base_timestamp = av_get_random_seed();
    s->timestamp = s->base_timestamp;
    s->cur_timestamp = 0;
    s->ssrc = av_get_random_seed();
    s->first_packet = 1;
    s->first_rtcp_ntp_time = ff_ntp_time();
    if (s1->start_time_realtime)
        /* Round the NTP time to whole milliseconds. */
        s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
                                 NTP_OFFSET_US;

    max_packet_size = s1->pb->max_packet_size;
    if (max_packet_size <= 12)
        return AVERROR(EIO);
    s->buf = av_malloc(max_packet_size);
    if (s->buf == NULL) {
        return AVERROR(ENOMEM);
    }
    s->max_payload_size = max_packet_size - 12;

    s->max_frames_per_packet = 0;
    if (s1->max_delay) {
        if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
            if (st->codec->frame_size == 0) {
                av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
            } else {
                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);
            }
        }
        if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
            /* FIXME: We should round down here... */
            s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
        }
    }

    av_set_pts_info(st, 32, 1, 90000);
    switch(st->codec->codec_id) {
    case CODEC_ID_MP2:
    case CODEC_ID_MP3:
        s->buf_ptr = s->buf + 4;
        break;
    case CODEC_ID_MPEG1VIDEO:
    case CODEC_ID_MPEG2VIDEO:
        break;
    case CODEC_ID_MPEG2TS:
        n = s->max_payload_size / TS_PACKET_SIZE;
        if (n < 1)
            n = 1;
        s->max_payload_size = n * TS_PACKET_SIZE;
        s->buf_ptr = s->buf;
        break;
    case CODEC_ID_H264:
        /* check for H.264 MP4 syntax */
        if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
            s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
        }
        break;
    case CODEC_ID_VORBIS:
    case CODEC_ID_THEORA:
        if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
        s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
        s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
        s->num_frames = 0;
        goto defaultcase;
    case CODEC_ID_VP8:
        av_log(s1, AV_LOG_ERROR, "RTP VP8 payload implementation is "
                                 "incompatible with the latest spec drafts.\n");
        break;
    case CODEC_ID_ADPCM_G722:
        /* Due to a historical error, the clock rate for G722 in RTP is
         * 8000, even if the sample rate is 16000. See RFC 3551. */
        av_set_pts_info(st, 32, 1, 8000);
        break;
    case CODEC_ID_AMR_NB:
    case CODEC_ID_AMR_WB:
        if (!s->max_frames_per_packet)
            s->max_frames_per_packet = 12;
        if (st->codec->codec_id == CODEC_ID_AMR_NB)
            n = 31;
        else
            n = 61;
        /* max_header_toc_size + the largest AMR payload must fit */
        if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
            av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
            return -1;
        }
        if (st->codec->channels != 1) {
            av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
            return -1;
        }
    case CODEC_ID_AAC:
        s->num_frames = 0;
    default:
defaultcase:
        if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
            av_set_pts_info(st, 32, 1, st->codec->sample_rate);
        }
        s->buf_ptr = s->buf;
        break;
    }

    return 0;
}

/* send an rtcp sender report packet */
static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
{
    RTPMuxContext *s = s1->priv_data;
    uint32_t rtp_ts;

    av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);

    s->last_rtcp_ntp_time = ntp_time;
    rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
                          s1->streams[0]->time_base) + s->base_timestamp;
    avio_w8(s1->pb, (RTP_VERSION << 6));
    avio_w8(s1->pb, RTCP_SR);
    avio_wb16(s1->pb, 6); /* length in words - 1 */
    avio_wb32(s1->pb, s->ssrc);
    avio_wb32(s1->pb, ntp_time / 1000000);
    avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
    avio_wb32(s1->pb, rtp_ts);
    avio_wb32(s1->pb, s->packet_count);
    avio_wb32(s1->pb, s->octet_count);
    avio_flush(s1->pb);
}

/* send an rtp packet. sequence number is incremented, but the caller
   must update the timestamp itself */
void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
{
    RTPMuxContext *s = s1->priv_data;

    av_dlog(s1, "rtp_send_data size=%d\n", len);

    /* build the RTP header */
    avio_w8(s1->pb, (RTP_VERSION << 6));
    avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
    avio_wb16(s1->pb, s->seq);
    avio_wb32(s1->pb, s->timestamp);
    avio_wb32(s1->pb, s->ssrc);

    avio_write(s1->pb, buf1, len);
    avio_flush(s1->pb);

    s->seq++;
    s->octet_count += len;
    s->packet_count++;
}

/* send an integer number of samples and compute time stamp and fill
   the rtp send buffer before sending. */
static void rtp_send_samples(AVFormatContext *s1,
                             const uint8_t *buf1, int size, int sample_size)
{
    RTPMuxContext *s = s1->priv_data;
    int len, max_packet_size, n;

    max_packet_size = (s->max_payload_size / sample_size) * sample_size;
    /* not needed, but who nows */
    if ((size % sample_size) != 0)
        av_abort();
    n = 0;
    while (size > 0) {
        s->buf_ptr = s->buf;
        len = FFMIN(max_packet_size, size);

        /* copy data */
        memcpy(s->buf_ptr, buf1, len);
        s->buf_ptr += len;
        buf1 += len;
        size -= len;
        s->timestamp = s->cur_timestamp + n / sample_size;
        ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
        n += (s->buf_ptr - s->buf);
    }
}

static void rtp_send_mpegaudio(AVFormatContext *s1,
                               const uint8_t *buf1, int size)
{
    RTPMuxContext *s = s1->priv_data;
    int len, count, max_packet_size;

    max_packet_size = s->max_payload_size;

    /* test if we must flush because not enough space */
    len = (s->buf_ptr - s->buf);
    if ((len + size) > max_packet_size) {
        if (len > 4) {
            ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
            s->buf_ptr = s->buf + 4;
        }
    }
    if (s->buf_ptr == s->buf + 4) {
        s->timestamp = s->cur_timestamp;
    }

    /* add the packet */
    if (size > max_packet_size) {
        /* big packet: fragment */
        count = 0;
        while (size > 0) {
            len = max_packet_size - 4;
            if (len > size)
                len = size;
            /* build fragmented packet */
            s->buf[0] = 0;
            s->buf[1] = 0;
            s->buf[2] = count >> 8;
            s->buf[3] = count;
            memcpy(s->buf + 4, buf1, len);
            ff_rtp_send_data(s1, s->buf, len + 4, 0);
            size -= len;
            buf1 += len;
            count += len;
        }
    } else {
        if (s->buf_ptr == s->buf + 4) {
            /* no fragmentation possible */
            s->buf[0] = 0;
            s->buf[1] = 0;
            s->buf[2] = 0;
            s->buf[3] = 0;
        }
        memcpy(s->buf_ptr, buf1, size);
        s->buf_ptr += size;
    }
}

static void rtp_send_raw(AVFormatContext *s1,
                         const uint8_t *buf1, int size)
{
    RTPMuxContext *s = s1->priv_data;
    int len, max_packet_size;

    max_packet_size = s->max_payload_size;

    while (size > 0) {
        len = max_packet_size;
        if (len > size)
            len = size;

        s->timestamp = s->cur_timestamp;
        ff_rtp_send_data(s1, buf1, len, (len == size));

        buf1 += len;
        size -= len;
    }
}

/* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
static void rtp_send_mpegts_raw(AVFormatContext *s1,
                                const uint8_t *buf1, int size)
{
    RTPMuxContext *s = s1->priv_data;
    int len, out_len;

    while (size >= TS_PACKET_SIZE) {
        len = s->max_payload_size - (s->buf_ptr - s->buf);
        if (len > size)
            len = size;
        memcpy(s->buf_ptr, buf1, len);
        buf1 += len;
        size -= len;
        s->buf_ptr += len;

        out_len = s->buf_ptr - s->buf;
        if (out_len >= s->max_payload_size) {
            ff_rtp_send_data(s1, s->buf, out_len, 0);
            s->buf_ptr = s->buf;
        }
    }
}

static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
{
    RTPMuxContext *s = s1->priv_data;
    AVStream *st = s1->streams[0];
    int rtcp_bytes;
    int size= pkt->size;

    av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size);

    rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
        RTCP_TX_RATIO_DEN;
    if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
                           (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) {
        rtcp_send_sr(s1, ff_ntp_time());
        s->last_octet_count = s->octet_count;
        s->first_packet = 0;
    }
    s->cur_timestamp = s->base_timestamp + pkt->pts;

    switch(st->codec->codec_id) {
    case CODEC_ID_PCM_MULAW:
    case CODEC_ID_PCM_ALAW:
    case CODEC_ID_PCM_U8:
    case CODEC_ID_PCM_S8:
        rtp_send_samples(s1, pkt->data, size, 1 * st->codec->channels);
        break;
    case CODEC_ID_PCM_U16BE:
    case CODEC_ID_PCM_U16LE:
    case CODEC_ID_PCM_S16BE:
    case CODEC_ID_PCM_S16LE:
        rtp_send_samples(s1, pkt->data, size, 2 * st->codec->channels);
        break;
    case CODEC_ID_ADPCM_G722:
        /* The actual sample size is half a byte per sample, but since the
         * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
         * the correct parameter for send_samples is 1 byte per stream clock. */
        rtp_send_samples(s1, pkt->data, size, 1 * st->codec->channels);
        break;
    case CODEC_ID_MP2:
    case CODEC_ID_MP3:
        rtp_send_mpegaudio(s1, pkt->data, size);
        break;
    case CODEC_ID_MPEG1VIDEO:
    case CODEC_ID_MPEG2VIDEO:
        ff_rtp_send_mpegvideo(s1, pkt->data, size);
        break;
    case CODEC_ID_AAC:
        if (s->flags & FF_RTP_FLAG_MP4A_LATM)
            ff_rtp_send_latm(s1, pkt->data, size);
        else
            ff_rtp_send_aac(s1, pkt->data, size);
        break;
    case CODEC_ID_AMR_NB:
    case CODEC_ID_AMR_WB:
        ff_rtp_send_amr(s1, pkt->data, size);
        break;
    case CODEC_ID_MPEG2TS:
        rtp_send_mpegts_raw(s1, pkt->data, size);
        break;
    case CODEC_ID_H264:
        ff_rtp_send_h264(s1, pkt->data, size);
        break;
    case CODEC_ID_H263:
    case CODEC_ID_H263P:
        ff_rtp_send_h263(s1, pkt->data, size);
        break;
    case CODEC_ID_VORBIS:
    case CODEC_ID_THEORA:
        ff_rtp_send_xiph(s1, pkt->data, size);
        break;
    case CODEC_ID_VP8:
        ff_rtp_send_vp8(s1, pkt->data, size);
        break;
    default:
        /* better than nothing : send the codec raw data */
        rtp_send_raw(s1, pkt->data, size);
        break;
    }
    return 0;
}

static int rtp_write_trailer(AVFormatContext *s1)
{
    RTPMuxContext *s = s1->priv_data;

    av_freep(&s->buf);

    return 0;
}

AVOutputFormat ff_rtp_muxer = {
    .name              = "rtp",
    .long_name         = NULL_IF_CONFIG_SMALL("RTP output format"),
    .priv_data_size    = sizeof(RTPMuxContext),
    .audio_codec       = CODEC_ID_PCM_MULAW,
    .video_codec       = CODEC_ID_NONE,
    .write_header      = rtp_write_header,
    .write_packet      = rtp_write_packet,
    .write_trailer     = rtp_write_trailer,
    .priv_class = &rtp_muxer_class,
};
OpenPOWER on IntegriCloud