* RTSP muxer
* Copyright (c) 2010 Martin Storsjo
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav 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,
+ * Libav 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
+ * License along with Libav; 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>
+#if HAVE_POLL_H
+#include <poll.h>
#endif
#include "network.h"
+#include "os_support.h"
#include "rtsp.h"
#include "internal.h"
-#include <libavutil/intreadwrite.h>
+#include "avio_internal.h"
+#include "libavutil/intreadwrite.h"
+#include "libavutil/avstring.h"
+#include "libavutil/time.h"
+#include "url.h"
+
+#define SDP_MAX_SIZE 16384
+
+static const AVClass rtsp_muxer_class = {
+ .class_name = "RTSP muxer",
+ .item_name = av_default_item_name,
+ .option = ff_rtsp_options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
+
+int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
+{
+ RTSPState *rt = s->priv_data;
+ RTSPMessageHeader reply1, *reply = &reply1;
+ int i;
+ char *sdp;
+ AVFormatContext sdp_ctx, *ctx_array[1];
+
+ s->start_time_realtime = av_gettime();
+
+ /* Announce the stream */
+ sdp = av_mallocz(SDP_MAX_SIZE);
+ if (!sdp)
+ return AVERROR(ENOMEM);
+ /* We create the SDP based on the RTSP AVFormatContext where we
+ * aren't allowed to change the filename field. (We create the SDP
+ * based on the RTSP context since the contexts for the RTP streams
+ * don't exist yet.) In order to specify a custom URL with the actual
+ * peer IP instead of the originally specified hostname, we create
+ * a temporary copy of the AVFormatContext, where the custom URL is set.
+ *
+ * FIXME: Create the SDP without copying the AVFormatContext.
+ * This either requires setting up the RTP stream AVFormatContexts
+ * already here (complicating things immensely) or getting a more
+ * flexible SDP creation interface.
+ */
+ sdp_ctx = *s;
+ ff_url_join(sdp_ctx.filename, sizeof(sdp_ctx.filename),
+ "rtsp", NULL, addr, -1, NULL);
+ ctx_array[0] = &sdp_ctx;
+ if (av_sdp_create(ctx_array, 1, sdp, SDP_MAX_SIZE)) {
+ av_free(sdp);
+ return AVERROR_INVALIDDATA;
+ }
+ av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
+ ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri,
+ "Content-Type: application/sdp\r\n",
+ reply, NULL, sdp, strlen(sdp));
+ av_free(sdp);
+ if (reply->status_code != RTSP_STATUS_OK)
+ return AVERROR_INVALIDDATA;
+
+ /* Set up the RTSPStreams for each AVStream */
+ for (i = 0; i < s->nb_streams; i++) {
+ RTSPStream *rtsp_st;
+
+ rtsp_st = av_mallocz(sizeof(RTSPStream));
+ if (!rtsp_st)
+ return AVERROR(ENOMEM);
+ dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
+
+ rtsp_st->stream_index = i;
+
+ av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url));
+ /* Note, this must match the relative uri set in the sdp content */
+ av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url),
+ "/streamid=%d", i);
+ }
+
+ return 0;
+}
static int rtsp_write_record(AVFormatContext *s)
{
char cmd[1024];
snprintf(cmd, sizeof(cmd),
- "Range: npt=%0.3f-\r\n",
- (double) 0);
+ "Range: npt=0.000-\r\n");
ff_rtsp_send_cmd(s, "RECORD", rt->control_uri, cmd, reply, NULL);
if (reply->status_code != RTSP_STATUS_OK)
return -1;
return 0;
}
-static int tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
+int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
{
RTSPState *rt = s->priv_data;
AVFormatContext *rtpctx = rtsp_st->transport_priv;
int size;
uint8_t *interleave_header, *interleaved_packet;
- size = url_close_dyn_buf(rtpctx->pb, &buf);
+ size = avio_close_dyn_buf(rtpctx->pb, &buf);
+ rtpctx->pb = NULL;
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
+ * ffio_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;
size -= 4;
if (packet_len > size || packet_len < 2)
break;
- if (ptr[1] >= 200 && ptr[1] <= 204)
+ if (RTP_PT_IS_RTCP(ptr[1]))
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_out, interleaved_packet, 4 + packet_len);
+ ffurl_write(rt->rtsp_hd_out, 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;
+ return ffio_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
}
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;
+ int n;
+ struct pollfd p = {ffurl_get_file_handle(rt->rtsp_hd), POLLIN, 0};
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);
+ n = poll(&p, 1, 0);
if (n <= 0)
break;
- if (FD_ISSET(tcp_fd, &rfds)) {
+ if (p.revents & POLLIN) {
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);
+ ret = ff_rtsp_read_reply(s, &reply, NULL, 1, NULL);
if (ret < 0)
return AVERROR(EPIPE);
if (ret == 1)
* 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);
+ ret = ff_rtsp_tcp_write_packet(s, rtsp_st);
return ret;
}
{
RTSPState *rt = s->priv_data;
+ // If we want to send RTCP_BYE packets, these are sent by av_write_trailer.
+ // Thus call this on all streams before doing the teardown. This is
+ // done within ff_rtsp_undo_setup.
+ ff_rtsp_undo_setup(s, 1);
+
ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
ff_rtsp_close_streams(s);
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,
+AVOutputFormat ff_rtsp_muxer = {
+ .name = "rtsp",
+ .long_name = NULL_IF_CONFIG_SMALL("RTSP output"),
+ .priv_data_size = sizeof(RTSPState),
+ .audio_codec = AV_CODEC_ID_AAC,
+ .video_codec = AV_CODEC_ID_MPEG4,
+ .write_header = rtsp_write_header,
+ .write_packet = rtsp_write_packet,
+ .write_trailer = rtsp_write_close,
+ .flags = AVFMT_NOFILE | AVFMT_GLOBALHEADER,
+ .priv_class = &rtsp_muxer_class,
};
-
projects / ffmpeg / blobdiff