view rtspenc.c @ 6085:72c7c3d5c4e9 libavformat

matroskaenc: Mux clusters better Start them on keyframes when reasonable, and delay writing audio packets to help ensure that there's audio samples available for the first frame in clusters. Patch by James Zern <jzern at google>

/*
 * RTSP muxer
 * Copyright (c) 2010 Martin Storsjo
 *
 * 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 "avformat.h"

#include <sys/time.h>
#if HAVE_SYS_SELECT_H
#include <sys/select.h>
#endif
#include "network.h"
#include "rtsp.h"
#include "internal.h"
#include <libavutil/intreadwrite.h>

static int rtsp_write_record(AVFormatContext *s)
{
    RTSPState *rt = s->priv_data;
    RTSPMessageHeader reply1, *reply = &reply1;
    char cmd[1024];

    snprintf(cmd, sizeof(cmd),
             "Range: npt=%0.3f-\r\n",
             (double) 0);
    ff_rtsp_send_cmd(s, "RECORD", rt->control_uri, cmd, reply, NULL);
    if (reply->status_code != RTSP_STATUS_OK)
        return -1;
    rt->state = RTSP_STATE_STREAMING;
    return 0;
}

static int rtsp_write_header(AVFormatContext *s)
{
    RTSPState *rt = s->priv_data;
    int ret;

    ret = ff_rtsp_connect(s);
    if (ret)
        return ret;

    if (rtsp_write_record(s) < 0) {
        ff_rtsp_close_streams(s);
        url_close(rt->rtsp_hd);
        return AVERROR_INVALIDDATA;
    }
    return 0;
}

static int tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
{
    RTSPState *rt = s->priv_data;
    AVFormatContext *rtpctx = rtsp_st->transport_priv;
    uint8_t *buf, *ptr;
    int size;
    uint8_t *interleave_header, *interleaved_packet;

    size = url_close_dyn_buf(rtpctx->pb, &buf);
    ptr = buf;
    while (size > 4) {
        uint32_t packet_len = AV_RB32(ptr);
        int id;
        /* The interleaving header is exactly 4 bytes, which happens to be
         * the same size as the packet length header from
         * url_open_dyn_packet_buf. So by writing the interleaving header
         * over these bytes, we get a consecutive interleaved packet
         * that can be written in one call. */
        interleaved_packet = interleave_header = ptr;
        ptr += 4;
        size -= 4;
        if (packet_len > size || packet_len < 2)
            break;
        if (ptr[1] >= 200 && ptr[1] <= 204)
            id = rtsp_st->interleaved_max; /* RTCP */
        else
            id = rtsp_st->interleaved_min; /* RTP */
        interleave_header[0] = '$';
        interleave_header[1] = id;
        AV_WB16(interleave_header + 2, packet_len);
        url_write(rt->rtsp_hd, interleaved_packet, 4 + packet_len);
        ptr += packet_len;
        size -= packet_len;
    }
    av_free(buf);
    url_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
    return 0;
}

static int rtsp_write_packet(AVFormatContext *s, AVPacket *pkt)
{
    RTSPState *rt = s->priv_data;
    RTSPStream *rtsp_st;
    fd_set rfds;
    int n, tcp_fd;
    struct timeval tv;
    AVFormatContext *rtpctx;
    int ret;

    tcp_fd = url_get_file_handle(rt->rtsp_hd);

    while (1) {
        FD_ZERO(&rfds);
        FD_SET(tcp_fd, &rfds);
        tv.tv_sec = 0;
        tv.tv_usec = 0;
        n = select(tcp_fd + 1, &rfds, NULL, NULL, &tv);
        if (n <= 0)
            break;
        if (FD_ISSET(tcp_fd, &rfds)) {
            RTSPMessageHeader reply;

            /* Don't let ff_rtsp_read_reply handle interleaved packets,
             * since it would block and wait for an RTSP reply on the socket
             * (which may not be coming any time soon) if it handles
             * interleaved packets internally. */
            ret = ff_rtsp_read_reply(s, &reply, NULL, 1);
            if (ret < 0)
                return AVERROR(EPIPE);
            if (ret == 1)
                ff_rtsp_skip_packet(s);
            /* XXX: parse message */
            if (rt->state != RTSP_STATE_STREAMING)
                return AVERROR(EPIPE);
        }
    }

    if (pkt->stream_index < 0 || pkt->stream_index >= rt->nb_rtsp_streams)
        return AVERROR_INVALIDDATA;
    rtsp_st = rt->rtsp_streams[pkt->stream_index];
    rtpctx = rtsp_st->transport_priv;

    ret = ff_write_chained(rtpctx, 0, pkt, s);
    /* ff_write_chained does all the RTP packetization. If using TCP as
     * transport, rtpctx->pb is only a dyn_packet_buf that queues up the
     * packets, so we need to send them out on the TCP connection separately.
     */
    if (!ret && rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP)
        ret = tcp_write_packet(s, rtsp_st);
    return ret;
}

static int rtsp_write_close(AVFormatContext *s)
{
    RTSPState *rt = s->priv_data;

    ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);

    ff_rtsp_close_streams(s);
    url_close(rt->rtsp_hd);
    ff_network_close();
    return 0;
}

AVOutputFormat rtsp_muxer = {
    "rtsp",
    NULL_IF_CONFIG_SMALL("RTSP output format"),
    NULL,
    NULL,
    sizeof(RTSPState),
    CODEC_ID_AAC,
    CODEC_ID_MPEG4,
    rtsp_write_header,
    rtsp_write_packet,
    rtsp_write_close,
    .flags = AVFMT_NOFILE | AVFMT_GLOBALHEADER,
};