X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavformat%2Frtspenc.c;h=902076d25dbbf9cfe1613568d37c50b02cb72b3b;hb=681ed0009905a088aae2a5caf2308d89aaa80562;hp=a4c7fd9a104e252d74ed9191e8286cc55475a780;hpb=76d908b3fe8953217c50f491c9b466c5d8a46be7;p=ffmpeg diff --git a/libavformat/rtspenc.c b/libavformat/rtspenc.c index a4c7fd9a104..902076d25db 100644 --- a/libavformat/rtspenc.c +++ b/libavformat/rtspenc.c @@ -2,33 +2,108 @@ * 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 -#if HAVE_SYS_SELECT_H -#include +#if HAVE_POLL_H +#include #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 "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 == NULL) + 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) { @@ -37,8 +112,7 @@ 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; @@ -70,14 +144,14 @@ static int tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st) 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; @@ -85,19 +159,19 @@ static int tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st) 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); + ffio_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE); return 0; } @@ -105,30 +179,23 @@ 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) @@ -166,17 +233,15 @@ static int rtsp_write_close(AVFormatContext *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 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, }; -