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ffmpeg / libavformat / rtpdec.c @ 3abe5fbd

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1 8eb793c4 Luca Abeni
/*
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 * RTP input format
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 * Copyright (c) 2002 Fabrice Bellard.
4
 *
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 * This file is part of FFmpeg.
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 *
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 * FFmpeg is free software; you can redistribute it and/or
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 * modify it under the terms of the GNU Lesser General Public
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 * License as published by the Free Software Foundation; either
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 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * FFmpeg is distributed in the hope that it will be useful,
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 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
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 * Lesser General Public License for more details.
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 *
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 * You should have received a copy of the GNU Lesser General Public
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 * License along with FFmpeg; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */
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#include "avformat.h"
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#include "mpegts.h"
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#include "bitstream.h"
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#include <unistd.h>
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#include "network.h"
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#include "rtp_internal.h"
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#include "rtp_h264.h"
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//#define DEBUG
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/* TODO: - add RTCP statistics reporting (should be optional).
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         - add support for h263/mpeg4 packetized output : IDEA: send a
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         buffer to 'rtp_write_packet' contains all the packets for ONE
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         frame. Each packet should have a four byte header containing
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         the length in big endian format (same trick as
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         'url_open_dyn_packet_buf')
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*/
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/* statistics functions */
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RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
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static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4};
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static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC};
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static void register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
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{
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    handler->next= RTPFirstDynamicPayloadHandler;
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    RTPFirstDynamicPayloadHandler= handler;
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}
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void av_register_rtp_dynamic_payload_handlers(void)
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{
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    register_dynamic_payload_handler(&mp4v_es_handler);
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    register_dynamic_payload_handler(&mpeg4_generic_handler);
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    register_dynamic_payload_handler(&ff_h264_dynamic_handler);
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}
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static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
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{
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    if (buf[1] != 200)
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        return -1;
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    s->last_rtcp_ntp_time = AV_RB64(buf + 8);
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    if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
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        s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
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    s->last_rtcp_timestamp = AV_RB32(buf + 16);
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    return 0;
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}
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#define RTP_SEQ_MOD (1<<16)
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/**
75
* called on parse open packet
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*/
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static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
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{
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    memset(s, 0, sizeof(RTPStatistics));
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    s->max_seq= base_sequence;
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    s->probation= 1;
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}
83
84
/**
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* called whenever there is a large jump in sequence numbers, or when they get out of probation...
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*/
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static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
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{
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    s->max_seq= seq;
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    s->cycles= 0;
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    s->base_seq= seq -1;
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    s->bad_seq= RTP_SEQ_MOD + 1;
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    s->received= 0;
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    s->expected_prior= 0;
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    s->received_prior= 0;
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    s->jitter= 0;
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    s->transit= 0;
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}
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/**
101
* returns 1 if we should handle this packet.
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*/
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static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
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{
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    uint16_t udelta= seq - s->max_seq;
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    const int MAX_DROPOUT= 3000;
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    const int MAX_MISORDER = 100;
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    const int MIN_SEQUENTIAL = 2;
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    /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
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    if(s->probation)
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    {
113
        if(seq==s->max_seq + 1) {
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            s->probation--;
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            s->max_seq= seq;
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            if(s->probation==0) {
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                rtp_init_sequence(s, seq);
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                s->received++;
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                return 1;
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            }
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        } else {
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            s->probation= MIN_SEQUENTIAL - 1;
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            s->max_seq = seq;
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        }
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    } else if (udelta < MAX_DROPOUT) {
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        // in order, with permissible gap
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        if(seq < s->max_seq) {
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            //sequence number wrapped; count antother 64k cycles
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            s->cycles += RTP_SEQ_MOD;
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        }
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        s->max_seq= seq;
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    } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
133
        // sequence made a large jump...
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        if(seq==s->bad_seq) {
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            // two sequential packets-- assume that the other side restarted without telling us; just resync.
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            rtp_init_sequence(s, seq);
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        } else {
138
            s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
139
            return 0;
140
        }
141
    } else {
142
        // duplicate or reordered packet...
143
    }
144
    s->received++;
145
    return 1;
146
}
147
148
#if 0
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/**
150
* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
151
* difference between the arrival and sent timestamp.  As a result, the jitter and transit statistics values
152
* never change.  I left this in in case someone else can see a way. (rdm)
153
*/
154
static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
155
{
156
    uint32_t transit= arrival_timestamp - sent_timestamp;
157
    int d;
158
    s->transit= transit;
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    d= FFABS(transit - s->transit);
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    s->jitter += d - ((s->jitter + 8)>>4);
161
}
162
#endif
163
164
int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
165
{
166
    ByteIOContext *pb;
167
    uint8_t *buf;
168
    int len;
169
    int rtcp_bytes;
170
    RTPStatistics *stats= &s->statistics;
171
    uint32_t lost;
172
    uint32_t extended_max;
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    uint32_t expected_interval;
174
    uint32_t received_interval;
175
    uint32_t lost_interval;
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    uint32_t expected;
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    uint32_t fraction;
178
    uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
179
180
    if (!s->rtp_ctx || (count < 1))
181
        return -1;
182
183
    /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
184
    /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
185
    s->octet_count += count;
186
    rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
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        RTCP_TX_RATIO_DEN;
188
    rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
189
    if (rtcp_bytes < 28)
190
        return -1;
191
    s->last_octet_count = s->octet_count;
192
193
    if (url_open_dyn_buf(&pb) < 0)
194
        return -1;
195
196
    // Receiver Report
197
    put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
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    put_byte(pb, 201);
199
    put_be16(pb, 7); /* length in words - 1 */
200
    put_be32(pb, s->ssrc); // our own SSRC
201
    put_be32(pb, s->ssrc); // XXX: should be the server's here!
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    // some placeholders we should really fill...
203
    // RFC 1889/p64
204
    extended_max= stats->cycles + stats->max_seq;
205
    expected= extended_max - stats->base_seq + 1;
206
    lost= expected - stats->received;
207
    lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
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    expected_interval= expected - stats->expected_prior;
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    stats->expected_prior= expected;
210
    received_interval= stats->received - stats->received_prior;
211
    stats->received_prior= stats->received;
212
    lost_interval= expected_interval - received_interval;
213
    if (expected_interval==0 || lost_interval<=0) fraction= 0;
214
    else fraction = (lost_interval<<8)/expected_interval;
215
216
    fraction= (fraction<<24) | lost;
217
218
    put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
219
    put_be32(pb, extended_max); /* max sequence received */
220
    put_be32(pb, stats->jitter>>4); /* jitter */
221
222
    if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
223
    {
224
        put_be32(pb, 0); /* last SR timestamp */
225
        put_be32(pb, 0); /* delay since last SR */
226
    } else {
227
        uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
228
        uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
229
230
        put_be32(pb, middle_32_bits); /* last SR timestamp */
231
        put_be32(pb, delay_since_last); /* delay since last SR */
232
    }
233
234
    // CNAME
235
    put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
236
    put_byte(pb, 202);
237
    len = strlen(s->hostname);
238
    put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */
239
    put_be32(pb, s->ssrc);
240
    put_byte(pb, 0x01);
241
    put_byte(pb, len);
242
    put_buffer(pb, s->hostname, len);
243
    // padding
244
    for (len = (6 + len) % 4; len % 4; len++) {
245
        put_byte(pb, 0);
246
    }
247
248
    put_flush_packet(pb);
249
    len = url_close_dyn_buf(pb, &buf);
250
    if ((len > 0) && buf) {
251
        int result;
252
#if defined(DEBUG)
253
        printf("sending %d bytes of RR\n", len);
254
#endif
255
        result= url_write(s->rtp_ctx, buf, len);
256
#if defined(DEBUG)
257
        printf("result from url_write: %d\n", result);
258
#endif
259
        av_free(buf);
260
    }
261
    return 0;
262
}
263
264
/**
265
 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
266
 * MPEG2TS streams to indicate that they should be demuxed inside the
267
 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
268
 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
269
 */
270
RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_t *rtp_payload_data)
271
{
272
    RTPDemuxContext *s;
273
274
    s = av_mallocz(sizeof(RTPDemuxContext));
275
    if (!s)
276
        return NULL;
277
    s->payload_type = payload_type;
278
    s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
279
    s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
280
    s->ic = s1;
281
    s->st = st;
282
    s->rtp_payload_data = rtp_payload_data;
283
    rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
284
    if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
285
        s->ts = mpegts_parse_open(s->ic);
286
        if (s->ts == NULL) {
287
            av_free(s);
288
            return NULL;
289
        }
290
    } else {
291
        switch(st->codec->codec_id) {
292
        case CODEC_ID_MPEG1VIDEO:
293
        case CODEC_ID_MPEG2VIDEO:
294
        case CODEC_ID_MP2:
295
        case CODEC_ID_MP3:
296
        case CODEC_ID_MPEG4:
297
        case CODEC_ID_H264:
298
            st->need_parsing = AVSTREAM_PARSE_FULL;
299
            break;
300
        default:
301
            break;
302
        }
303
    }
304
    // needed to send back RTCP RR in RTSP sessions
305
    s->rtp_ctx = rtpc;
306
    gethostname(s->hostname, sizeof(s->hostname));
307
    return s;
308
}
309
310
static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
311
{
312
    int au_headers_length, au_header_size, i;
313
    GetBitContext getbitcontext;
314
    rtp_payload_data_t *infos;
315
316
    infos = s->rtp_payload_data;
317
318
    if (infos == NULL)
319
        return -1;
320
321
    /* decode the first 2 bytes where are stored the AUHeader sections
322
       length in bits */
323
    au_headers_length = AV_RB16(buf);
324
325
    if (au_headers_length > RTP_MAX_PACKET_LENGTH)
326
      return -1;
327
328
    infos->au_headers_length_bytes = (au_headers_length + 7) / 8;
329
330
    /* skip AU headers length section (2 bytes) */
331
    buf += 2;
332
333
    init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);
334
335
    /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
336
    au_header_size = infos->sizelength + infos->indexlength;
337
    if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
338
        return -1;
339
340
    infos->nb_au_headers = au_headers_length / au_header_size;
341
    infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);
342
343
    /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
344
       In my test, the FAAD decoder does not behave correctly when sending each AU one by one
345
       but does when sending the whole as one big packet...  */
346
    infos->au_headers[0].size = 0;
347
    infos->au_headers[0].index = 0;
348
    for (i = 0; i < infos->nb_au_headers; ++i) {
349
        infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
350
        infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
351
    }
352
353
    infos->nb_au_headers = 1;
354
355
    return 0;
356
}
357
358
/**
359
 * This was the second switch in rtp_parse packet.  Normalizes time, if required, sets stream_index, etc.
360
 */
361
static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
362
{
363
    switch(s->st->codec->codec_id) {
364
        case CODEC_ID_MP2:
365
        case CODEC_ID_MPEG1VIDEO:
366
        case CODEC_ID_MPEG2VIDEO:
367
            if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
368
                int64_t addend;
369
370
                int delta_timestamp;
371
                /* XXX: is it really necessary to unify the timestamp base ? */
372
                /* compute pts from timestamp with received ntp_time */
373
                delta_timestamp = timestamp - s->last_rtcp_timestamp;
374
                /* convert to 90 kHz without overflow */
375
                addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14;
376
                addend = (addend * 5625) >> 14;
377
                pkt->pts = addend + delta_timestamp;
378
            }
379
            break;
380
        case CODEC_ID_AAC:
381
        case CODEC_ID_H264:
382
        case CODEC_ID_MPEG4:
383
            pkt->pts = timestamp;
384
            break;
385
        default:
386
            /* no timestamp info yet */
387
            break;
388
    }
389
    pkt->stream_index = s->st->index;
390
}
391
392
/**
393
 * Parse an RTP or RTCP packet directly sent as a buffer.
394
 * @param s RTP parse context.
395
 * @param pkt returned packet
396
 * @param buf input buffer or NULL to read the next packets
397
 * @param len buffer len
398
 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
399
 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
400
 */
401
int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
402
                     const uint8_t *buf, int len)
403
{
404
    unsigned int ssrc, h;
405
    int payload_type, seq, ret;
406
    AVStream *st;
407
    uint32_t timestamp;
408
    int rv= 0;
409
410
    if (!buf) {
411
        /* return the next packets, if any */
412
        if(s->st && s->parse_packet) {
413
            timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
414
            rv= s->parse_packet(s, pkt, &timestamp, NULL, 0);
415
            finalize_packet(s, pkt, timestamp);
416
            return rv;
417
        } else {
418
            // TODO: Move to a dynamic packet handler (like above)
419
            if (s->read_buf_index >= s->read_buf_size)
420
                return -1;
421
            ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
422
                                      s->read_buf_size - s->read_buf_index);
423
            if (ret < 0)
424
                return -1;
425
            s->read_buf_index += ret;
426
            if (s->read_buf_index < s->read_buf_size)
427
                return 1;
428
            else
429
                return 0;
430
        }
431
    }
432
433
    if (len < 12)
434
        return -1;
435
436
    if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
437
        return -1;
438
    if (buf[1] >= 200 && buf[1] <= 204) {
439
        rtcp_parse_packet(s, buf, len);
440
        return -1;
441
    }
442
    payload_type = buf[1] & 0x7f;
443
    seq  = AV_RB16(buf + 2);
444
    timestamp = AV_RB32(buf + 4);
445
    ssrc = AV_RB32(buf + 8);
446
    /* store the ssrc in the RTPDemuxContext */
447
    s->ssrc = ssrc;
448
449
    /* NOTE: we can handle only one payload type */
450
    if (s->payload_type != payload_type)
451
        return -1;
452
453
    st = s->st;
454
    // only do something with this if all the rtp checks pass...
455
    if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
456
    {
457
        av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
458
               payload_type, seq, ((s->seq + 1) & 0xffff));
459
        return -1;
460
    }
461
462
    s->seq = seq;
463
    len -= 12;
464
    buf += 12;
465
466
    if (!st) {
467
        /* specific MPEG2TS demux support */
468
        ret = mpegts_parse_packet(s->ts, pkt, buf, len);
469
        if (ret < 0)
470
            return -1;
471
        if (ret < len) {
472
            s->read_buf_size = len - ret;
473
            memcpy(s->buf, buf + ret, s->read_buf_size);
474
            s->read_buf_index = 0;
475
            return 1;
476
        }
477
    } else {
478
        // at this point, the RTP header has been stripped;  This is ASSUMING that there is only 1 CSRC, which in't wise.
479
        switch(st->codec->codec_id) {
480
        case CODEC_ID_MP2:
481
            /* better than nothing: skip mpeg audio RTP header */
482
            if (len <= 4)
483
                return -1;
484
            h = AV_RB32(buf);
485
            len -= 4;
486
            buf += 4;
487
            av_new_packet(pkt, len);
488
            memcpy(pkt->data, buf, len);
489
            break;
490
        case CODEC_ID_MPEG1VIDEO:
491
        case CODEC_ID_MPEG2VIDEO:
492
            /* better than nothing: skip mpeg video RTP header */
493
            if (len <= 4)
494
                return -1;
495
            h = AV_RB32(buf);
496
            buf += 4;
497
            len -= 4;
498
            if (h & (1 << 26)) {
499
                /* mpeg2 */
500
                if (len <= 4)
501
                    return -1;
502
                buf += 4;
503
                len -= 4;
504
            }
505
            av_new_packet(pkt, len);
506
            memcpy(pkt->data, buf, len);
507
            break;
508
            // moved from below, verbatim.  this is because this section handles packets, and the lower switch handles
509
            // timestamps.
510
            // TODO: Put this into a dynamic packet handler...
511
        case CODEC_ID_AAC:
512
            if (rtp_parse_mp4_au(s, buf))
513
                return -1;
514
            {
515
                rtp_payload_data_t *infos = s->rtp_payload_data;
516
                if (infos == NULL)
517
                    return -1;
518
                buf += infos->au_headers_length_bytes + 2;
519
                len -= infos->au_headers_length_bytes + 2;
520
521
                /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
522
                    one au_header */
523
                av_new_packet(pkt, infos->au_headers[0].size);
524
                memcpy(pkt->data, buf, infos->au_headers[0].size);
525
                buf += infos->au_headers[0].size;
526
                len -= infos->au_headers[0].size;
527
            }
528
            s->read_buf_size = len;
529
            rv= 0;
530
            break;
531
        default:
532
            if(s->parse_packet) {
533
                rv= s->parse_packet(s, pkt, &timestamp, buf, len);
534
            } else {
535
                av_new_packet(pkt, len);
536
                memcpy(pkt->data, buf, len);
537
            }
538
            break;
539
        }
540
541
        // now perform timestamp things....
542
        finalize_packet(s, pkt, timestamp);
543
    }
544
    return rv;
545
}
546
547
void rtp_parse_close(RTPDemuxContext *s)
548
{
549
    // TODO: fold this into the protocol specific data fields.
550
    if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
551
        mpegts_parse_close(s->ts);
552
    }
553
    av_free(s);
554
}