rtpdec.c

ffmpeg移植到symbian的全部源代码

/*
 * RTP input format
 * Copyright (c) 2002 Fabrice Bellard.
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg 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.
 *
 * FFmpeg 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 FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

#include "libavcodec/bitstream.h"
#include "avformat.h"
#include "mpegts.h"

#include <unistd.h>
#include "network.h"

#include "rtp_internal.h"
#include "rtp_h264.h"

//#define DEBUG

/* TODO: - add RTCP statistics reporting (should be optional).

         - add support for h263/mpeg4 packetized output : IDEA: send a
         buffer to 'rtp_write_packet' contains all the packets for ONE
         frame. Each packet should have a four byte header containing
         the length in big endian format (same trick as
         'url_open_dyn_packet_buf')
*/

/* statistics functions */
RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;

static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4};
static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC};

static void register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
{
    handler->next= RTPFirstDynamicPayloadHandler;
    RTPFirstDynamicPayloadHandler= handler;
}

void av_register_rtp_dynamic_payload_handlers(void)
{
    register_dynamic_payload_handler(&mp4v_es_handler);
    register_dynamic_payload_handler(&mpeg4_generic_handler);
    register_dynamic_payload_handler(&ff_h264_dynamic_handler);
}

static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
{
    if (buf[1] != 200)
        return -1;
    s->last_rtcp_ntp_time = AV_RB64(buf + 8);
    if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
        s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
    s->last_rtcp_timestamp = AV_RB32(buf + 16);
    return 0;
}

#define RTP_SEQ_MOD (1<<16)

/**
* called on parse open packet
*/
static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
{
    memset(s, 0, sizeof(RTPStatistics));
    s->max_seq= base_sequence;
    s->probation= 1;
}

/**
* called whenever there is a large jump in sequence numbers, or when they get out of probation...
*/
static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
{
    s->max_seq= seq;
    s->cycles= 0;
    s->base_seq= seq -1;
    s->bad_seq= RTP_SEQ_MOD + 1;
    s->received= 0;
    s->expected_prior= 0;
    s->received_prior= 0;
    s->jitter= 0;
    s->transit= 0;
}

/**
* returns 1 if we should handle this packet.
*/
static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
{
    uint16_t udelta= seq - s->max_seq;
    const int MAX_DROPOUT= 3000;
    const int MAX_MISORDER = 100;
    const int MIN_SEQUENTIAL = 2;

    /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
    if(s->probation)
    {
        if(seq==s->max_seq + 1) {
            s->probation--;
            s->max_seq= seq;
            if(s->probation==0) {
                rtp_init_sequence(s, seq);
                s->received++;
                return 1;
            }
        } else {
            s->probation= MIN_SEQUENTIAL - 1;
            s->max_seq = seq;
        }
    } else if (udelta < MAX_DROPOUT) {
        // in order, with permissible gap
        if(seq < s->max_seq) {
            //sequence number wrapped; count antother 64k cycles
            s->cycles += RTP_SEQ_MOD;
        }
        s->max_seq= seq;
    } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
        // sequence made a large jump...
        if(seq==s->bad_seq) {
            // two sequential packets-- assume that the other side restarted without telling us; just resync.
            rtp_init_sequence(s, seq);
        } else {
            s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
            return 0;
        }
    } else {
        // duplicate or reordered packet...
    }
    s->received++;
    return 1;
}

#if 0
/**
* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
* difference between the arrival and sent timestamp.  As a result, the jitter and transit statistics values
* never change.  I left this in in case someone else can see a way. (rdm)
*/
static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
{
    uint32_t transit= arrival_timestamp - sent_timestamp;
    int d;
    s->transit= transit;
    d= FFABS(transit - s->transit);
    s->jitter += d - ((s->jitter + 8)>>4);
}
#endif

int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
{
    ByteIOContext *pb;
    uint8_t *buf;
    int len;
    int rtcp_bytes;
    RTPStatistics *stats= &s->statistics;
    uint32_t lost;
    uint32_t extended_max;
    uint32_t expected_interval;
    uint32_t received_interval;
    uint32_t lost_interval;
    uint32_t expected;
    uint32_t fraction;
    uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?

    if (!s->rtp_ctx || (count < 1))
        return -1;

    /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
    /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
    s->octet_count += count;
    rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
        RTCP_TX_RATIO_DEN;
    rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
    if (rtcp_bytes < 28)
        return -1;
    s->last_octet_count = s->octet_count;

    if (url_open_dyn_buf(&pb) < 0)
        return -1;

    // Receiver Report
    put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
    put_byte(pb, 201);
    put_be16(pb, 7); /* length in words - 1 */
    put_be32(pb, s->ssrc); // our own SSRC
    put_be32(pb, s->ssrc); // XXX: should be the server's here!
    // some placeholders we should really fill...
    // RFC 1889/p64
    extended_max= stats->cycles + stats->max_seq;
    expected= extended_max - stats->base_seq + 1;
    lost= expected - stats->received;
    lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
    expected_interval= expected - stats->expected_prior;
    stats->expected_prior= expected;
    received_interval= stats->received - stats->received_prior;
    stats->received_prior= stats->received;
    lost_interval= expected_interval - received_interval;
    if (expected_interval==0 || lost_interval<=0) fraction= 0;
    else fraction = (lost_interval<<8)/expected_interval;

    fraction= (fraction<<24) | lost;

    put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
    put_be32(pb, extended_max); /* max sequence received */
    put_be32(pb, stats->jitter>>4); /* jitter */

    if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
    {
        put_be32(pb, 0); /* last SR timestamp */
        put_be32(pb, 0); /* delay since last SR */
    } else {
        uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
        uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;

        put_be32(pb, middle_32_bits); /* last SR timestamp */
        put_be32(pb, delay_since_last); /* delay since last SR */
    }

    // CNAME
    put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
    put_byte(pb, 202);
    len = strlen(s->hostname);
    put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */
    put_be32(pb, s->ssrc);
    put_byte(pb, 0x01);
    put_byte(pb, len);
#ifdef __CW32__
    put_buffer(pb, (const unsigned char*)s->hostname, len);
#else
    put_buffer(pb, s->hostname, len);
#endif
    // padding
    for (len = (6 + len) % 4; len % 4; len++) {
        put_byte(pb, 0);
    }

    put_flush_packet(pb);
    len = url_close_dyn_buf(pb, &buf);
    if ((len > 0) && buf) {
        int result;
        dprintf(s->ic, "sending %d bytes of RR\n", len);
        result= url_write(s->rtp_ctx, buf, len);
        dprintf(s->ic, "result from url_write: %d\n", result);
        av_free(buf);
    }
    return 0;
}

/**
 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
 * MPEG2TS streams to indicate that they should be demuxed inside the
 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
 */
RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_t *rtp_payload_data)