diff options
Diffstat (limited to 'ffmpeg/libavformat/rtspenc.c')
| -rw-r--r-- | ffmpeg/libavformat/rtspenc.c | 247 |
1 files changed, 247 insertions, 0 deletions
diff --git a/ffmpeg/libavformat/rtspenc.c b/ffmpeg/libavformat/rtspenc.c new file mode 100644 index 0000000..bad6fbd --- /dev/null +++ b/ffmpeg/libavformat/rtspenc.c @@ -0,0 +1,247 @@ +/* + * RTSP muxer + * Copyright (c) 2010 Martin Storsjo + * + * This file is part of FFmpeg. + * + * FFmpeg 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, + * 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 + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include "avformat.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 "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) +{ + RTSPState *rt = s->priv_data; + RTSPMessageHeader reply1, *reply = &reply1; + char cmd[1024]; + + snprintf(cmd, sizeof(cmd), + "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; + rt->state = RTSP_STATE_STREAMING; + return 0; +} + +static int rtsp_write_header(AVFormatContext *s) +{ + int ret; + + ret = ff_rtsp_connect(s); + if (ret) + return ret; + + if (rtsp_write_record(s) < 0) { + ff_rtsp_close_streams(s); + ff_rtsp_close_connections(s); + return AVERROR_INVALIDDATA; + } + return 0; +} + +static int tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st) +{ + RTSPState *rt = s->priv_data; + AVFormatContext *rtpctx = rtsp_st->transport_priv; + uint8_t *buf, *ptr; + int size; + uint8_t *interleave_header, *interleaved_packet; + + 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 + * 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; + ptr += 4; + size -= 4; + if (packet_len > size || packet_len < 2) + break; + 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); + ffurl_write(rt->rtsp_hd_out, interleaved_packet, 4 + packet_len); + ptr += packet_len; + size -= packet_len; + } + av_free(buf); + ffio_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE); + return 0; +} + +static int rtsp_write_packet(AVFormatContext *s, AVPacket *pkt) +{ + RTSPState *rt = s->priv_data; + RTSPStream *rtsp_st; + int n; + struct pollfd p = {ffurl_get_file_handle(rt->rtsp_hd), POLLIN, 0}; + AVFormatContext *rtpctx; + int ret; + + while (1) { + n = poll(&p, 1, 0); + if (n <= 0) + break; + 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, NULL); + if (ret < 0) + return AVERROR(EPIPE); + if (ret == 1) + ff_rtsp_skip_packet(s); + /* XXX: parse message */ + if (rt->state != RTSP_STATE_STREAMING) + return AVERROR(EPIPE); + } + } + + if (pkt->stream_index < 0 || pkt->stream_index >= rt->nb_rtsp_streams) + return AVERROR_INVALIDDATA; + rtsp_st = rt->rtsp_streams[pkt->stream_index]; + rtpctx = rtsp_st->transport_priv; + + ret = ff_write_chained(rtpctx, 0, pkt, s); + /* ff_write_chained does all the RTP packetization. If using TCP as + * transport, rtpctx->pb is only a dyn_packet_buf that queues up the + * 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); + return ret; +} + +static int rtsp_write_close(AVFormatContext *s) +{ + RTSPState *rt = s->priv_data; + + ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL); + + ff_rtsp_close_streams(s); + ff_rtsp_close_connections(s); + ff_network_close(); + return 0; +} + +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, +}; |