1/*
2 * RTSP muxer
3 * Copyright (c) 2010 Martin Storsjo
4 *
5 * This file is part of Libav.
6 *
7 * Libav is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * Libav is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
15 * Lesser General Public License for more details.
16 *
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with Libav; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21
22#include "avformat.h"
23
24#include <sys/time.h>
25#if HAVE_POLL_H
26#include <poll.h>
27#endif
28#include "network.h"
29#include "os_support.h"
30#include "rtsp.h"
31#include "internal.h"
32#include "avio_internal.h"
33#include "libavutil/intreadwrite.h"
34#include "libavutil/avstring.h"
35#include "url.h"
36
37#define SDP_MAX_SIZE 16384
38
39static const AVClass rtsp_muxer_class = {
40    .class_name = "RTSP muxer",
41    .item_name  = av_default_item_name,
42    .option     = ff_rtsp_options,
43    .version    = LIBAVUTIL_VERSION_INT,
44};
45
46int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
47{
48    RTSPState *rt = s->priv_data;
49    RTSPMessageHeader reply1, *reply = &reply1;
50    int i;
51    char *sdp;
52    AVFormatContext sdp_ctx, *ctx_array[1];
53
54    s->start_time_realtime = av_gettime();
55
56    /* Announce the stream */
57    sdp = av_mallocz(SDP_MAX_SIZE);
58    if (sdp == NULL)
59        return AVERROR(ENOMEM);
60    /* We create the SDP based on the RTSP AVFormatContext where we
61     * aren't allowed to change the filename field. (We create the SDP
62     * based on the RTSP context since the contexts for the RTP streams
63     * don't exist yet.) In order to specify a custom URL with the actual
64     * peer IP instead of the originally specified hostname, we create
65     * a temporary copy of the AVFormatContext, where the custom URL is set.
66     *
67     * FIXME: Create the SDP without copying the AVFormatContext.
68     * This either requires setting up the RTP stream AVFormatContexts
69     * already here (complicating things immensely) or getting a more
70     * flexible SDP creation interface.
71     */
72    sdp_ctx = *s;
73    ff_url_join(sdp_ctx.filename, sizeof(sdp_ctx.filename),
74                "rtsp", NULL, addr, -1, NULL);
75    ctx_array[0] = &sdp_ctx;
76    if (av_sdp_create(ctx_array, 1, sdp, SDP_MAX_SIZE)) {
77        av_free(sdp);
78        return AVERROR_INVALIDDATA;
79    }
80    av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
81    ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri,
82                                  "Content-Type: application/sdp\r\n",
83                                  reply, NULL, sdp, strlen(sdp));
84    av_free(sdp);
85    if (reply->status_code != RTSP_STATUS_OK)
86        return AVERROR_INVALIDDATA;
87
88    /* Set up the RTSPStreams for each AVStream */
89    for (i = 0; i < s->nb_streams; i++) {
90        RTSPStream *rtsp_st;
91
92        rtsp_st = av_mallocz(sizeof(RTSPStream));
93        if (!rtsp_st)
94            return AVERROR(ENOMEM);
95        dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
96
97        rtsp_st->stream_index = i;
98
99        av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url));
100        /* Note, this must match the relative uri set in the sdp content */
101        av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url),
102                    "/streamid=%d", i);
103    }
104
105    return 0;
106}
107
108static int rtsp_write_record(AVFormatContext *s)
109{
110    RTSPState *rt = s->priv_data;
111    RTSPMessageHeader reply1, *reply = &reply1;
112    char cmd[1024];
113
114    snprintf(cmd, sizeof(cmd),
115             "Range: npt=0.000-\r\n");
116    ff_rtsp_send_cmd(s, "RECORD", rt->control_uri, cmd, reply, NULL);
117    if (reply->status_code != RTSP_STATUS_OK)
118        return -1;
119    rt->state = RTSP_STATE_STREAMING;
120    return 0;
121}
122
123static int rtsp_write_header(AVFormatContext *s)
124{
125    int ret;
126
127    ret = ff_rtsp_connect(s);
128    if (ret)
129        return ret;
130
131    if (rtsp_write_record(s) < 0) {
132        ff_rtsp_close_streams(s);
133        ff_rtsp_close_connections(s);
134        return AVERROR_INVALIDDATA;
135    }
136    return 0;
137}
138
139static int tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
140{
141    RTSPState *rt = s->priv_data;
142    AVFormatContext *rtpctx = rtsp_st->transport_priv;
143    uint8_t *buf, *ptr;
144    int size;
145    uint8_t *interleave_header, *interleaved_packet;
146
147    size = avio_close_dyn_buf(rtpctx->pb, &buf);
148    ptr = buf;
149    while (size > 4) {
150        uint32_t packet_len = AV_RB32(ptr);
151        int id;
152        /* The interleaving header is exactly 4 bytes, which happens to be
153         * the same size as the packet length header from
154         * ffio_open_dyn_packet_buf. So by writing the interleaving header
155         * over these bytes, we get a consecutive interleaved packet
156         * that can be written in one call. */
157        interleaved_packet = interleave_header = ptr;
158        ptr += 4;
159        size -= 4;
160        if (packet_len > size || packet_len < 2)
161            break;
162        if (ptr[1] >= RTCP_SR && ptr[1] <= RTCP_APP)
163            id = rtsp_st->interleaved_max; /* RTCP */
164        else
165            id = rtsp_st->interleaved_min; /* RTP */
166        interleave_header[0] = '$';
167        interleave_header[1] = id;
168        AV_WB16(interleave_header + 2, packet_len);
169        ffurl_write(rt->rtsp_hd_out, interleaved_packet, 4 + packet_len);
170        ptr += packet_len;
171        size -= packet_len;
172    }
173    av_free(buf);
174    ffio_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
175    return 0;
176}
177
178static int rtsp_write_packet(AVFormatContext *s, AVPacket *pkt)
179{
180    RTSPState *rt = s->priv_data;
181    RTSPStream *rtsp_st;
182    int n;
183    struct pollfd p = {ffurl_get_file_handle(rt->rtsp_hd), POLLIN, 0};
184    AVFormatContext *rtpctx;
185    int ret;
186
187    while (1) {
188        n = poll(&p, 1, 0);
189        if (n <= 0)
190            break;
191        if (p.revents & POLLIN) {
192            RTSPMessageHeader reply;
193
194            /* Don't let ff_rtsp_read_reply handle interleaved packets,
195             * since it would block and wait for an RTSP reply on the socket
196             * (which may not be coming any time soon) if it handles
197             * interleaved packets internally. */
198            ret = ff_rtsp_read_reply(s, &reply, NULL, 1, NULL);
199            if (ret < 0)
200                return AVERROR(EPIPE);
201            if (ret == 1)
202                ff_rtsp_skip_packet(s);
203            /* XXX: parse message */
204            if (rt->state != RTSP_STATE_STREAMING)
205                return AVERROR(EPIPE);
206        }
207    }
208
209    if (pkt->stream_index < 0 || pkt->stream_index >= rt->nb_rtsp_streams)
210        return AVERROR_INVALIDDATA;
211    rtsp_st = rt->rtsp_streams[pkt->stream_index];
212    rtpctx = rtsp_st->transport_priv;
213
214    ret = ff_write_chained(rtpctx, 0, pkt, s);
215    /* ff_write_chained does all the RTP packetization. If using TCP as
216     * transport, rtpctx->pb is only a dyn_packet_buf that queues up the
217     * packets, so we need to send them out on the TCP connection separately.
218     */
219    if (!ret && rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP)
220        ret = tcp_write_packet(s, rtsp_st);
221    return ret;
222}
223
224static int rtsp_write_close(AVFormatContext *s)
225{
226    RTSPState *rt = s->priv_data;
227
228    ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
229
230    ff_rtsp_close_streams(s);
231    ff_rtsp_close_connections(s);
232    ff_network_close();
233    return 0;
234}
235
236AVOutputFormat ff_rtsp_muxer = {
237    .name              = "rtsp",
238    .long_name         = NULL_IF_CONFIG_SMALL("RTSP output format"),
239    .priv_data_size    = sizeof(RTSPState),
240    .audio_codec       = CODEC_ID_AAC,
241    .video_codec       = CODEC_ID_MPEG4,
242    .write_header      = rtsp_write_header,
243    .write_packet      = rtsp_write_packet,
244    .write_trailer     = rtsp_write_close,
245    .flags = AVFMT_NOFILE | AVFMT_GLOBALHEADER,
246    .priv_class = &rtsp_muxer_class,
247};
248
249