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ffmpeg / libavformat / rtspenc.c @ d9c0510e

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/*
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 * RTSP muxer
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 * Copyright (c) 2010 Martin Storsjo
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 *
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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 <sys/time.h>
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#if HAVE_POLL_H
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#include <poll.h>
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#endif
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#include "network.h"
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#include "os_support.h"
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#include "rtsp.h"
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#include "internal.h"
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#include "libavutil/intreadwrite.h"
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#include "libavutil/avstring.h"
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#define SDP_MAX_SIZE 16384
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int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
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{
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    RTSPState *rt = s->priv_data;
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    RTSPMessageHeader reply1, *reply = &reply1;
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    int i;
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    char *sdp;
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    AVFormatContext sdp_ctx, *ctx_array[1];
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    s->start_time_realtime = av_gettime();
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    /* Announce the stream */
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    sdp = av_mallocz(SDP_MAX_SIZE);
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    if (sdp == NULL)
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        return AVERROR(ENOMEM);
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    /* We create the SDP based on the RTSP AVFormatContext where we
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     * aren't allowed to change the filename field. (We create the SDP
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     * based on the RTSP context since the contexts for the RTP streams
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     * don't exist yet.) In order to specify a custom URL with the actual
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     * peer IP instead of the originally specified hostname, we create
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     * a temporary copy of the AVFormatContext, where the custom URL is set.
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     *
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     * FIXME: Create the SDP without copying the AVFormatContext.
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     * This either requires setting up the RTP stream AVFormatContexts
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     * already here (complicating things immensely) or getting a more
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     * flexible SDP creation interface.
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     */
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    sdp_ctx = *s;
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    ff_url_join(sdp_ctx.filename, sizeof(sdp_ctx.filename),
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                "rtsp", NULL, addr, -1, NULL);
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    ctx_array[0] = &sdp_ctx;
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    if (avf_sdp_create(ctx_array, 1, sdp, SDP_MAX_SIZE)) {
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        av_free(sdp);
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        return AVERROR_INVALIDDATA;
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    }
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    av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
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    ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri,
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                                  "Content-Type: application/sdp\r\n",
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                                  reply, NULL, sdp, strlen(sdp));
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    av_free(sdp);
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    if (reply->status_code != RTSP_STATUS_OK)
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        return AVERROR_INVALIDDATA;
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    /* Set up the RTSPStreams for each AVStream */
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    for (i = 0; i < s->nb_streams; i++) {
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        RTSPStream *rtsp_st;
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        rtsp_st = av_mallocz(sizeof(RTSPStream));
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        if (!rtsp_st)
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            return AVERROR(ENOMEM);
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        dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
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        rtsp_st->stream_index = i;
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        av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url));
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        /* Note, this must match the relative uri set in the sdp content */
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        av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url),
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                    "/streamid=%d", i);
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    }
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    return 0;
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}
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static int rtsp_write_record(AVFormatContext *s)
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{
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    RTSPState *rt = s->priv_data;
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    RTSPMessageHeader reply1, *reply = &reply1;
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    char cmd[1024];
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    snprintf(cmd, sizeof(cmd),
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             "Range: npt=%0.3f-\r\n",
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             (double) 0);
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    ff_rtsp_send_cmd(s, "RECORD", rt->control_uri, cmd, reply, NULL);
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    if (reply->status_code != RTSP_STATUS_OK)
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        return -1;
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    rt->state = RTSP_STATE_STREAMING;
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    return 0;
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}
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static int rtsp_write_header(AVFormatContext *s)
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{
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    int ret;
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    ret = ff_rtsp_connect(s);
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    if (ret)
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        return ret;
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    if (rtsp_write_record(s) < 0) {
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        ff_rtsp_close_streams(s);
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        ff_rtsp_close_connections(s);
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        return AVERROR_INVALIDDATA;
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    }
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    return 0;
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}
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static int tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
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{
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    RTSPState *rt = s->priv_data;
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    AVFormatContext *rtpctx = rtsp_st->transport_priv;
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    uint8_t *buf, *ptr;
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    int size;
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    uint8_t *interleave_header, *interleaved_packet;
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    size = url_close_dyn_buf(rtpctx->pb, &buf);
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    ptr = buf;
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    while (size > 4) {
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        uint32_t packet_len = AV_RB32(ptr);
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        int id;
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        /* The interleaving header is exactly 4 bytes, which happens to be
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         * the same size as the packet length header from
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         * url_open_dyn_packet_buf. So by writing the interleaving header
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         * over these bytes, we get a consecutive interleaved packet
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         * that can be written in one call. */
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        interleaved_packet = interleave_header = ptr;
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        ptr += 4;
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        size -= 4;
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        if (packet_len > size || packet_len < 2)
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            break;
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        if (ptr[1] >= RTCP_SR && ptr[1] <= RTCP_APP)
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            id = rtsp_st->interleaved_max; /* RTCP */
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        else
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            id = rtsp_st->interleaved_min; /* RTP */
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        interleave_header[0] = '$';
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        interleave_header[1] = id;
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        AV_WB16(interleave_header + 2, packet_len);
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        url_write(rt->rtsp_hd_out, interleaved_packet, 4 + packet_len);
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        ptr += packet_len;
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        size -= packet_len;
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    }
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    av_free(buf);
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    url_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
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    return 0;
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}
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static int rtsp_write_packet(AVFormatContext *s, AVPacket *pkt)
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{
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    RTSPState *rt = s->priv_data;
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    RTSPStream *rtsp_st;
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    int n;
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    struct pollfd p = {url_get_file_handle(rt->rtsp_hd), POLLIN, 0};
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    AVFormatContext *rtpctx;
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    int ret;
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    while (1) {
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        n = poll(&p, 1, 0);
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        if (n <= 0)
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            break;
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        if (p.revents & POLLIN) {
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            RTSPMessageHeader reply;
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            /* Don't let ff_rtsp_read_reply handle interleaved packets,
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             * since it would block and wait for an RTSP reply on the socket
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             * (which may not be coming any time soon) if it handles
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             * interleaved packets internally. */
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            ret = ff_rtsp_read_reply(s, &reply, NULL, 1, NULL);
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            if (ret < 0)
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                return AVERROR(EPIPE);
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            if (ret == 1)
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                ff_rtsp_skip_packet(s);
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            /* XXX: parse message */
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            if (rt->state != RTSP_STATE_STREAMING)
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                return AVERROR(EPIPE);
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        }
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    }
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    if (pkt->stream_index < 0 || pkt->stream_index >= rt->nb_rtsp_streams)
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        return AVERROR_INVALIDDATA;
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    rtsp_st = rt->rtsp_streams[pkt->stream_index];
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    rtpctx = rtsp_st->transport_priv;
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    ret = ff_write_chained(rtpctx, 0, pkt, s);
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    /* ff_write_chained does all the RTP packetization. If using TCP as
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     * transport, rtpctx->pb is only a dyn_packet_buf that queues up the
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     * packets, so we need to send them out on the TCP connection separately.
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     */
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    if (!ret && rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP)
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        ret = tcp_write_packet(s, rtsp_st);
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    return ret;
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}
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static int rtsp_write_close(AVFormatContext *s)
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{
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    RTSPState *rt = s->priv_data;
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    ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
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    ff_rtsp_close_streams(s);
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    ff_rtsp_close_connections(s);
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    ff_network_close();
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    return 0;
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}
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AVOutputFormat ff_rtsp_muxer = {
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    "rtsp",
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    NULL_IF_CONFIG_SMALL("RTSP output format"),
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    NULL,
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    NULL,
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    sizeof(RTSPState),
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    CODEC_ID_AAC,
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    CODEC_ID_MPEG4,
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    rtsp_write_header,
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    rtsp_write_packet,
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    rtsp_write_close,
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    .flags = AVFMT_NOFILE | AVFMT_GLOBALHEADER,
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};
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