view rtpproto.c @ 4723:a2390c6a35e6 libavformat

Fix index generation in the way that it was supposed to be used. See the discussion in the ML thread "[PATCH] rmdec.c: merge old/new packet reading code". Over time, this code broke somewhat, e.g. seq was never actually written into (and was thus always 1, therefore the seq condition was always true), whereas it was supposed to be set to the sequence number of the video slice in case the video frame is divided over multiple RM packets (slices). The problem of this is that packets other than those containing the beginning of a video frame would be indexed as well. Secondly, flags&2 is supposed to be true for video keyframes and for these audio packets containing the start of a block. For some codecs (e.g. AAC), that is every single packet, whereas for others (e.g. cook), that is the packet containing the first of a series of scrambled packets that are to be descrambled together. Indexing any of the following would lead to incomplete and thus useless frames. Problem here is that flags would be reset to 2 to indicate that the first packet is ready to be returned, and in addition if no data was left to be returned (which is always true for the first packet), then we wouldn't actually write the index entry anyway. All in all, the idea was good and it probably worked at some point, but that is long ago. This patch should at the very least make it likely for this code to be executed again at the right times, i.e. the way it was originally intended to be used.

/*
 * RTP network protocol
 * 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
 */

/**
 * @file libavformat/rtpproto.c
 * RTP protocol
 */

#include "libavutil/avstring.h"
#include "avformat.h"

#include <unistd.h>
#include <stdarg.h>
#include "network.h"
#include "os_support.h"
#include <fcntl.h>
#if HAVE_SYS_SELECT_H
#include <sys/select.h>
#endif

#define RTP_TX_BUF_SIZE  (64 * 1024)
#define RTP_RX_BUF_SIZE  (128 * 1024)

typedef struct RTPContext {
    URLContext *rtp_hd, *rtcp_hd;
    int rtp_fd, rtcp_fd;
} RTPContext;

/**
 * If no filename is given to av_open_input_file because you want to
 * get the local port first, then you must call this function to set
 * the remote server address.
 *
 * @param s1 media file context
 * @param uri of the remote server
 * @return zero if no error.
 */

int rtp_set_remote_url(URLContext *h, const char *uri)
{
    RTPContext *s = h->priv_data;
    char hostname[256];
    int port;

    char buf[1024];
    char path[1024];

    url_split(NULL, 0, NULL, 0, hostname, sizeof(hostname), &port,
              path, sizeof(path), uri);

    snprintf(buf, sizeof(buf), "udp://%s:%d%s", hostname, port, path);
    udp_set_remote_url(s->rtp_hd, buf);

    snprintf(buf, sizeof(buf), "udp://%s:%d%s", hostname, port + 1, path);
    udp_set_remote_url(s->rtcp_hd, buf);
    return 0;
}


/**
 * add option to url of the form:
 * "http://host:port/path?option1=val1&option2=val2...
 */

static void url_add_option(char *buf, int buf_size, const char *fmt, ...)
{
    char buf1[1024];
    va_list ap;

    va_start(ap, fmt);
    if (strchr(buf, '?'))
        av_strlcat(buf, "&", buf_size);
    else
        av_strlcat(buf, "?", buf_size);
    vsnprintf(buf1, sizeof(buf1), fmt, ap);
    av_strlcat(buf, buf1, buf_size);
    va_end(ap);
}

static void build_udp_url(char *buf, int buf_size,
                          const char *hostname, int port,
                          int local_port, int ttl,
                          int max_packet_size)
{
    snprintf(buf, buf_size, "udp://%s:%d", hostname, port);
    if (local_port >= 0)
        url_add_option(buf, buf_size, "localport=%d", local_port);
    if (ttl >= 0)
        url_add_option(buf, buf_size, "ttl=%d", ttl);
    if (max_packet_size >=0)
        url_add_option(buf, buf_size, "pkt_size=%d", max_packet_size);
}

/**
 * url syntax: rtp://host:port[?option=val...]
 * option: 'ttl=n'       : set the ttl value (for multicast only)
 *         'localport=n' : set the local port to n
 *         'pkt_size=n'  : set max packet size
 *
 */

static int rtp_open(URLContext *h, const char *uri, int flags)
{
    RTPContext *s;
    int port, is_output, ttl, local_port, max_packet_size;
    char hostname[256];
    char buf[1024];
    char path[1024];
    const char *p;

    is_output = (flags & URL_WRONLY);

    s = av_mallocz(sizeof(RTPContext));
    if (!s)
        return AVERROR(ENOMEM);
    h->priv_data = s;

    url_split(NULL, 0, NULL, 0, hostname, sizeof(hostname), &port,
              path, sizeof(path), uri);
    /* extract parameters */
    ttl = -1;
    local_port = -1;
    max_packet_size = -1;

    p = strchr(uri, '?');
    if (p) {
        if (find_info_tag(buf, sizeof(buf), "ttl", p)) {
            ttl = strtol(buf, NULL, 10);
        }
        if (find_info_tag(buf, sizeof(buf), "localport", p)) {
            local_port = strtol(buf, NULL, 10);
        }
        if (find_info_tag(buf, sizeof(buf), "pkt_size", p)) {
            max_packet_size = strtol(buf, NULL, 10);
        }
    }

    build_udp_url(buf, sizeof(buf),
                  hostname, port, local_port, ttl, max_packet_size);
    if (url_open(&s->rtp_hd, buf, flags) < 0)
        goto fail;
    local_port = udp_get_local_port(s->rtp_hd);
    /* XXX: need to open another connection if the port is not even */

    /* well, should suppress localport in path */

    build_udp_url(buf, sizeof(buf),
                  hostname, port + 1, local_port + 1, ttl, max_packet_size);
    if (url_open(&s->rtcp_hd, buf, flags) < 0)
        goto fail;

    /* just to ease handle access. XXX: need to suppress direct handle
       access */
    s->rtp_fd = url_get_file_handle(s->rtp_hd);
    s->rtcp_fd = url_get_file_handle(s->rtcp_hd);

    h->max_packet_size = url_get_max_packet_size(s->rtp_hd);
    h->is_streamed = 1;
    return 0;

 fail:
    if (s->rtp_hd)
        url_close(s->rtp_hd);
    if (s->rtcp_hd)
        url_close(s->rtcp_hd);
    av_free(s);
    return AVERROR(EIO);
}

static int rtp_read(URLContext *h, uint8_t *buf, int size)
{
    RTPContext *s = h->priv_data;
    struct sockaddr_in from;
    socklen_t from_len;
    int len, fd_max, n;
    fd_set rfds;
#if 0
    for(;;) {
        from_len = sizeof(from);
        len = recvfrom (s->rtp_fd, buf, size, 0,
                        (struct sockaddr *)&from, &from_len);
        if (len < 0) {
            if (ff_neterrno() == FF_NETERROR(EAGAIN) ||
                ff_neterrno() == FF_NETERROR(EINTR))
                continue;
            return AVERROR(EIO);
        }
        break;
    }
#else
    for(;;) {
        /* build fdset to listen to RTP and RTCP packets */
        FD_ZERO(&rfds);
        fd_max = s->rtp_fd;
        FD_SET(s->rtp_fd, &rfds);
        if (s->rtcp_fd > fd_max)
            fd_max = s->rtcp_fd;
        FD_SET(s->rtcp_fd, &rfds);
        n = select(fd_max + 1, &rfds, NULL, NULL, NULL);
        if (n > 0) {
            /* first try RTCP */
            if (FD_ISSET(s->rtcp_fd, &rfds)) {
                from_len = sizeof(from);
                len = recvfrom (s->rtcp_fd, buf, size, 0,
                                (struct sockaddr *)&from, &from_len);
                if (len < 0) {
                    if (ff_neterrno() == FF_NETERROR(EAGAIN) ||
                        ff_neterrno() == FF_NETERROR(EINTR))
                        continue;
                    return AVERROR(EIO);
                }
                break;
            }
            /* then RTP */
            if (FD_ISSET(s->rtp_fd, &rfds)) {
                from_len = sizeof(from);
                len = recvfrom (s->rtp_fd, buf, size, 0,
                                (struct sockaddr *)&from, &from_len);
                if (len < 0) {
                    if (ff_neterrno() == FF_NETERROR(EAGAIN) ||
                        ff_neterrno() == FF_NETERROR(EINTR))
                        continue;
                    return AVERROR(EIO);
                }
                break;
            }
        }
    }
#endif
    return len;
}

static int rtp_write(URLContext *h, uint8_t *buf, int size)
{
    RTPContext *s = h->priv_data;
    int ret;
    URLContext *hd;

    if (buf[1] >= 200 && buf[1] <= 204) {
        /* RTCP payload type */
        hd = s->rtcp_hd;
    } else {
        /* RTP payload type */
        hd = s->rtp_hd;
    }

    ret = url_write(hd, buf, size);
#if 0
    {
        struct timespec ts;
        ts.tv_sec = 0;
        ts.tv_nsec = 10 * 1000000;
        nanosleep(&ts, NULL);
    }
#endif
    return ret;
}

static int rtp_close(URLContext *h)
{
    RTPContext *s = h->priv_data;

    url_close(s->rtp_hd);
    url_close(s->rtcp_hd);
    av_free(s);
    return 0;
}

/**
 * Return the local port used by the RTP connection
 * @param s1 media file context
 * @return the local port number
 */

int rtp_get_local_port(URLContext *h)
{
    RTPContext *s = h->priv_data;
    return udp_get_local_port(s->rtp_hd);
}

#if (LIBAVFORMAT_VERSION_MAJOR <= 52)
/**
 * Return the rtp and rtcp file handles for select() usage to wait for
 * several RTP streams at the same time.
 * @param h media file context
 */

void rtp_get_file_handles(URLContext *h, int *prtp_fd, int *prtcp_fd)
{
    RTPContext *s = h->priv_data;

    *prtp_fd = s->rtp_fd;
    *prtcp_fd = s->rtcp_fd;
}
#endif

static int rtp_get_file_handle(URLContext *h)
{
    RTPContext *s = h->priv_data;
    return s->rtp_fd;
}

URLProtocol rtp_protocol = {
    "rtp",
    rtp_open,
    rtp_read,
    rtp_write,
    NULL, /* seek */
    rtp_close,
    .url_get_file_handle = rtp_get_file_handle,
};