* 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 "avio_internal.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/avstring.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;
ff_url_join(sdp_ctx.filename, sizeof(sdp_ctx.filename),
"rtsp", NULL, addr, -1, NULL);
ctx_array[0] = &sdp_ctx;
- if (avf_sdp_create(ctx_array, 1, sdp, SDP_MAX_SIZE)) {
+ if (av_sdp_create(ctx_array, 1, sdp, SDP_MAX_SIZE)) {
av_free(sdp);
return AVERROR_INVALIDDATA;
}
/* Set up the RTSPStreams for each AVStream */
for (i = 0; i < s->nb_streams; i++) {
RTSPStream *rtsp_st;
- AVStream *st = s->streams[i];
rtsp_st = av_mallocz(sizeof(RTSPStream));
if (!rtsp_st)
return AVERROR(ENOMEM);
dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
- st->priv_data = rtsp_st;
rtsp_st->stream_index = i;
av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url));
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;
int size;
uint8_t *interleave_header, *interleaved_packet;
- size = url_close_dyn_buf(rtpctx->pb, &buf);
+ size = avio_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
+ * 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;
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);
+ ffio_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
return 0;
}
{
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)
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,
+AVOutputFormat ff_rtsp_muxer = {
+ .name = "rtsp",
+ .long_name = NULL_IF_CONFIG_SMALL("RTSP output format"),
+ .priv_data_size = sizeof(RTSPState),
+ .audio_codec = CODEC_ID_AAC,
+ .video_codec = 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,
};