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https://git.ffmpeg.org/ffmpeg.git
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8c1ebdcea2
* qatar/master: shorten: Use separate pointers for the allocated memory for decoded samples. atrac3: Fix crash in tonal component decoding. ws_snd1: Fix wrong samples counts. movenc: Don't set a default sample duration when creating ismv rtp: Factorize the check for distinguishing RTCP packets from RTP golomb: avoid infinite loop on all-zero input (or end of buffer). bethsoftvid: synchronize video timestamps with audio sample rate bethsoftvid: add audio stream only after getting the first audio packet bethsoftvid: Set video packet duration instead of accumulating pts. bethsoftvid: set packet key frame flag for audio and I-frame video packets. bethsoftvid: fix read_packet() return codes. bethsoftvid: pass palette in side data instead of in a separate packet. sdp: Ignore RTCP packets when autodetecting RTP streams proresenc: initialise 'sign' variable mpegaudio: replace memcpy by SIMD code vc1: prevent using last_frame as a reference for I/P first frame. Conflicts: libavcodec/atrac3.c libavcodec/golomb.h libavcodec/shorten.c libavcodec/ws-snd1.c tests/ref/fate/bethsoft-vid Merged-by: Michael Niedermayer <michaelni@gmx.at>
336 lines
9.7 KiB
C
336 lines
9.7 KiB
C
/*
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* RTP network protocol
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* Copyright (c) 2002 Fabrice Bellard
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* RTP protocol
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*/
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#include "libavutil/parseutils.h"
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#include "libavutil/avstring.h"
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#include "avformat.h"
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#include "avio_internal.h"
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#include "rtpdec.h"
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#include "url.h"
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#include <unistd.h>
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#include <stdarg.h>
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#include "internal.h"
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#include "network.h"
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#include "os_support.h"
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#include <fcntl.h>
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#if HAVE_POLL_H
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#include <sys/poll.h>
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#endif
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#include <sys/time.h>
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#define RTP_TX_BUF_SIZE (64 * 1024)
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#define RTP_RX_BUF_SIZE (128 * 1024)
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typedef struct RTPContext {
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URLContext *rtp_hd, *rtcp_hd;
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int rtp_fd, rtcp_fd;
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} RTPContext;
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/**
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* If no filename is given to av_open_input_file because you want to
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* get the local port first, then you must call this function to set
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* the remote server address.
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*
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* @param h media file context
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* @param uri of the remote server
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* @return zero if no error.
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*/
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int ff_rtp_set_remote_url(URLContext *h, const char *uri)
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{
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RTPContext *s = h->priv_data;
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char hostname[256];
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int port;
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char buf[1024];
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char path[1024];
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av_url_split(NULL, 0, NULL, 0, hostname, sizeof(hostname), &port,
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path, sizeof(path), uri);
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ff_url_join(buf, sizeof(buf), "udp", NULL, hostname, port, "%s", path);
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ff_udp_set_remote_url(s->rtp_hd, buf);
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ff_url_join(buf, sizeof(buf), "udp", NULL, hostname, port + 1, "%s", path);
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ff_udp_set_remote_url(s->rtcp_hd, buf);
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return 0;
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}
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/**
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* add option to url of the form:
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* "http://host:port/path?option1=val1&option2=val2...
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*/
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static av_printf_format(3, 4) void url_add_option(char *buf, int buf_size, const char *fmt, ...)
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{
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char buf1[1024];
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va_list ap;
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va_start(ap, fmt);
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if (strchr(buf, '?'))
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av_strlcat(buf, "&", buf_size);
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else
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av_strlcat(buf, "?", buf_size);
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vsnprintf(buf1, sizeof(buf1), fmt, ap);
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av_strlcat(buf, buf1, buf_size);
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va_end(ap);
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}
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static void build_udp_url(char *buf, int buf_size,
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const char *hostname, int port,
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int local_port, int ttl,
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int max_packet_size, int connect)
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{
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ff_url_join(buf, buf_size, "udp", NULL, hostname, port, NULL);
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if (local_port >= 0)
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url_add_option(buf, buf_size, "localport=%d", local_port);
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if (ttl >= 0)
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url_add_option(buf, buf_size, "ttl=%d", ttl);
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if (max_packet_size >=0)
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url_add_option(buf, buf_size, "pkt_size=%d", max_packet_size);
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if (connect)
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url_add_option(buf, buf_size, "connect=1");
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url_add_option(buf, buf_size, "fifo_size=0");
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}
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/**
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* url syntax: rtp://host:port[?option=val...]
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* option: 'ttl=n' : set the ttl value (for multicast only)
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* 'rtcpport=n' : set the remote rtcp port to n
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* 'localrtpport=n' : set the local rtp port to n
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* 'localrtcpport=n' : set the local rtcp port to n
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* 'pkt_size=n' : set max packet size
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* 'connect=0/1' : do a connect() on the UDP socket
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* deprecated option:
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* 'localport=n' : set the local port to n
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*
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* if rtcpport isn't set the rtcp port will be the rtp port + 1
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* if local rtp port isn't set any available port will be used for the local
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* rtp and rtcp ports
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* if the local rtcp port is not set it will be the local rtp port + 1
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*/
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static int rtp_open(URLContext *h, const char *uri, int flags)
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{
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RTPContext *s = h->priv_data;
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int rtp_port, rtcp_port,
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ttl, connect,
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local_rtp_port, local_rtcp_port, max_packet_size;
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char hostname[256];
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char buf[1024];
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char path[1024];
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const char *p;
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av_url_split(NULL, 0, NULL, 0, hostname, sizeof(hostname), &rtp_port,
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path, sizeof(path), uri);
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/* extract parameters */
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ttl = -1;
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rtcp_port = rtp_port+1;
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local_rtp_port = -1;
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local_rtcp_port = -1;
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max_packet_size = -1;
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connect = 0;
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p = strchr(uri, '?');
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if (p) {
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if (av_find_info_tag(buf, sizeof(buf), "ttl", p)) {
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ttl = strtol(buf, NULL, 10);
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}
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if (av_find_info_tag(buf, sizeof(buf), "rtcpport", p)) {
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rtcp_port = strtol(buf, NULL, 10);
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}
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if (av_find_info_tag(buf, sizeof(buf), "localport", p)) {
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local_rtp_port = strtol(buf, NULL, 10);
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}
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if (av_find_info_tag(buf, sizeof(buf), "localrtpport", p)) {
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local_rtp_port = strtol(buf, NULL, 10);
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}
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if (av_find_info_tag(buf, sizeof(buf), "localrtcpport", p)) {
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local_rtcp_port = strtol(buf, NULL, 10);
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}
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if (av_find_info_tag(buf, sizeof(buf), "pkt_size", p)) {
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max_packet_size = strtol(buf, NULL, 10);
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}
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if (av_find_info_tag(buf, sizeof(buf), "connect", p)) {
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connect = strtol(buf, NULL, 10);
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}
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}
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build_udp_url(buf, sizeof(buf),
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hostname, rtp_port, local_rtp_port, ttl, max_packet_size,
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connect);
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if (ffurl_open(&s->rtp_hd, buf, flags, &h->interrupt_callback, NULL) < 0)
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goto fail;
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if (local_rtp_port>=0 && local_rtcp_port<0)
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local_rtcp_port = ff_udp_get_local_port(s->rtp_hd) + 1;
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build_udp_url(buf, sizeof(buf),
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hostname, rtcp_port, local_rtcp_port, ttl, max_packet_size,
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connect);
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if (ffurl_open(&s->rtcp_hd, buf, flags, &h->interrupt_callback, NULL) < 0)
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goto fail;
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/* just to ease handle access. XXX: need to suppress direct handle
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access */
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s->rtp_fd = ffurl_get_file_handle(s->rtp_hd);
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s->rtcp_fd = ffurl_get_file_handle(s->rtcp_hd);
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h->max_packet_size = s->rtp_hd->max_packet_size;
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h->is_streamed = 1;
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return 0;
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fail:
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if (s->rtp_hd)
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ffurl_close(s->rtp_hd);
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if (s->rtcp_hd)
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ffurl_close(s->rtcp_hd);
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return AVERROR(EIO);
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}
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static int rtp_read(URLContext *h, uint8_t *buf, int size)
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{
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RTPContext *s = h->priv_data;
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struct sockaddr_storage from;
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socklen_t from_len;
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int len, n;
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struct pollfd p[2] = {{s->rtp_fd, POLLIN, 0}, {s->rtcp_fd, POLLIN, 0}};
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for(;;) {
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if (ff_check_interrupt(&h->interrupt_callback))
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return AVERROR_EXIT;
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/* build fdset to listen to RTP and RTCP packets */
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n = poll(p, 2, 100);
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if (n > 0) {
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/* first try RTCP */
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if (p[1].revents & POLLIN) {
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from_len = sizeof(from);
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len = recvfrom (s->rtcp_fd, buf, size, 0,
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(struct sockaddr *)&from, &from_len);
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if (len < 0) {
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if (ff_neterrno() == AVERROR(EAGAIN) ||
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ff_neterrno() == AVERROR(EINTR))
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continue;
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return AVERROR(EIO);
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}
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break;
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}
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/* then RTP */
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if (p[0].revents & POLLIN) {
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from_len = sizeof(from);
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len = recvfrom (s->rtp_fd, buf, size, 0,
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(struct sockaddr *)&from, &from_len);
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if (len < 0) {
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if (ff_neterrno() == AVERROR(EAGAIN) ||
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ff_neterrno() == AVERROR(EINTR))
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continue;
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return AVERROR(EIO);
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}
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break;
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}
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} else if (n < 0) {
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if (ff_neterrno() == AVERROR(EINTR))
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continue;
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return AVERROR(EIO);
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}
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}
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return len;
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}
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static int rtp_write(URLContext *h, const uint8_t *buf, int size)
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{
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RTPContext *s = h->priv_data;
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int ret;
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URLContext *hd;
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if (RTP_PT_IS_RTCP(buf[1])) {
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/* RTCP payload type */
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hd = s->rtcp_hd;
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} else {
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/* RTP payload type */
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hd = s->rtp_hd;
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}
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ret = ffurl_write(hd, buf, size);
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return ret;
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}
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static int rtp_close(URLContext *h)
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{
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RTPContext *s = h->priv_data;
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ffurl_close(s->rtp_hd);
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ffurl_close(s->rtcp_hd);
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return 0;
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}
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/**
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* Return the local rtp port used by the RTP connection
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* @param h media file context
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* @return the local port number
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*/
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int ff_rtp_get_local_rtp_port(URLContext *h)
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{
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RTPContext *s = h->priv_data;
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return ff_udp_get_local_port(s->rtp_hd);
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}
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/**
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* Return the local rtcp port used by the RTP connection
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* @param h media file context
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* @return the local port number
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*/
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int ff_rtp_get_local_rtcp_port(URLContext *h)
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{
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RTPContext *s = h->priv_data;
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return ff_udp_get_local_port(s->rtcp_hd);
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}
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static int rtp_get_file_handle(URLContext *h)
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{
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RTPContext *s = h->priv_data;
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return s->rtp_fd;
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}
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int ff_rtp_get_rtcp_file_handle(URLContext *h) {
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RTPContext *s = h->priv_data;
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return s->rtcp_fd;
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}
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URLProtocol ff_rtp_protocol = {
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.name = "rtp",
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.url_open = rtp_open,
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.url_read = rtp_read,
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.url_write = rtp_write,
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.url_close = rtp_close,
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.url_get_file_handle = rtp_get_file_handle,
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.priv_data_size = sizeof(RTPContext),
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.flags = URL_PROTOCOL_FLAG_NETWORK,
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};
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