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Diffstat (limited to 'ffmpeg1/libavformat/rtspenc.c')
-rw-r--r--ffmpeg1/libavformat/rtspenc.c247
1 files changed, 0 insertions, 247 deletions
diff --git a/ffmpeg1/libavformat/rtspenc.c b/ffmpeg1/libavformat/rtspenc.c
deleted file mode 100644
index bad6fbd..0000000
--- a/ffmpeg1/libavformat/rtspenc.c
+++ /dev/null
@@ -1,247 +0,0 @@
-/*
- * 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,
-};
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