ffmpeg-2.1.1: move directory

git-svn-id: svn://kolibrios.org@6148 a494cfbc-eb01-0410-851d-a64ba20cac60

contrib/sdk/sources/ffmpeg/ffmpeg-2.1/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);
+    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
+         * 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);
+    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;
+    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,
+};