* RTSP/SDP client
* Copyright (c) 2002 Fabrice Bellard
*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * 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.
*
- * Libav is distributed in the hope that it will be useful,
+ * 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 Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
+#include "libavutil/avassert.h"
#include "libavutil/base64.h"
+#include "libavutil/bprint.h"
#include "libavutil/avstring.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/mathematics.h"
#include "rtpenc.h"
#include "mpegts.h"
-/* Timeout values for socket poll, in ms,
- * and read_packet(), in seconds */
-#define POLL_TIMEOUT_MS 100
+/* Default timeout values for read packet in seconds */
#define READ_PACKET_TIMEOUT_S 10
-#define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
-#define SDP_MAX_SIZE 16384
#define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
#define DEFAULT_REORDERING_DELAY 100000
#define RTSP_FLAG_OPTS(name, longname) \
{ name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
- { "filter_src", "Only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }
+ { "filter_src", "only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }
#define RTSP_MEDIATYPE_OPTS(name, longname) \
- { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { .i64 = (1 << (AVMEDIA_TYPE_DATA+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
+ { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { .i64 = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
{ "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
{ "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
- { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
+ { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }, \
+ { "subtitle", "Subtitle", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_SUBTITLE}, 0, 0, DEC, "allowed_media_types" }
#define COMMON_OPTS() \
- { "reorder_queue_size", "Number of packets to buffer for handling of reordered packets", OFFSET(reordering_queue_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC }, \
+ { "reorder_queue_size", "set number of packets to buffer for handling of reordered packets", OFFSET(reordering_queue_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC }, \
{ "buffer_size", "Underlying protocol send/receive buffer size", OFFSET(buffer_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC|ENC }, \
{ "pkt_size", "Underlying protocol send packet size", OFFSET(pkt_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, ENC } \
const AVOption ff_rtsp_options[] = {
- { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC },
+ { "initial_pause", "do not start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, DEC },
FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
- { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
+ { "rtsp_transport", "set RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
{ "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
{ "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
{ "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
{ "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
- RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
- { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },
- RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
- { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
- { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
- { "timeout", "Maximum timeout (in seconds) to wait for incoming connections. -1 is infinite. Implies flag listen", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },
+ { "https", "HTTPS tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTPS )}, 0, 0, DEC, "rtsp_transport" },
+ RTSP_FLAG_OPTS("rtsp_flags", "set RTSP flags"),
+ { "listen", "wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },
+ { "prefer_tcp", "try RTP via TCP first, if available", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_PREFER_TCP}, 0, 0, DEC|ENC, "rtsp_flags" },
+ { "satip_raw", "export raw MPEG-TS stream instead of demuxing", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_SATIP_RAW}, 0, 0, DEC, "rtsp_flags" },
+ RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
+ { "min_port", "set minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
+ { "max_port", "set maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
+ { "listen_timeout", "set maximum timeout (in seconds) to wait for incoming connections (-1 is infinite, imply flag listen)", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },
+ { "timeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
COMMON_OPTS(),
+ { "user_agent", "override User-Agent header", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC },
{ NULL },
};
static const AVOption sdp_options[] = {
RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
- { "custom_io", "Use custom IO", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
- { "rtcp_to_source", "Send RTCP packets to the source address of received packets", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_RTCP_TO_SOURCE}, 0, 0, DEC, "rtsp_flags" },
- RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
+ { "custom_io", "use custom I/O", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
+ { "rtcp_to_source", "send RTCP packets to the source address of received packets", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_RTCP_TO_SOURCE}, 0, 0, DEC, "rtsp_flags" },
+ { "listen_timeout", "set maximum timeout (in seconds) to wait for incoming connections", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = READ_PACKET_TIMEOUT_S}, INT_MIN, INT_MAX, DEC },
+ RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
COMMON_OPTS(),
{ NULL },
};
static const AVOption rtp_options[] = {
- RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
+ RTSP_FLAG_OPTS("rtp_flags", "set RTP flags"),
+ { "listen_timeout", "set maximum timeout (in seconds) to wait for incoming connections", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = READ_PACKET_TIMEOUT_S}, INT_MIN, INT_MAX, DEC },
+ RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
COMMON_OPTS(),
{ NULL },
};
if (*p == '-') {
p++;
get_word_sep(buf, sizeof(buf), "-", &p);
- av_parse_time(end, buf, 1);
+ if (av_parse_time(end, buf, 1) < 0)
+ av_log(NULL, AV_LOG_DEBUG, "Failed to parse interval end specification '%s'\n", buf);
}
}
}
#if CONFIG_RTPDEC
-static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
+static void init_rtp_handler(const RTPDynamicProtocolHandler *handler,
RTSPStream *rtsp_st, AVStream *st)
{
AVCodecParameters *par = st ? st->codecpar : NULL;
}
}
+static int init_satip_stream(AVFormatContext *s)
+{
+ RTSPState *rt = s->priv_data;
+ RTSPStream *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->sdp_payload_type = 33; // MP2T
+ av_strlcpy(rtsp_st->control_url,
+ rt->control_uri, sizeof(rtsp_st->control_url));
+
+ if (rt->rtsp_flags & RTSP_FLAG_SATIP_RAW) {
+ AVStream *st = avformat_new_stream(s, NULL);
+ if (!st)
+ return AVERROR(ENOMEM);
+ st->id = rt->nb_rtsp_streams - 1;
+ rtsp_st->stream_index = st->index;
+ st->codecpar->codec_type = AVMEDIA_TYPE_DATA;
+ st->codecpar->codec_id = AV_CODEC_ID_MPEG2TS;
+ } else {
+ rtsp_st->stream_index = -1;
+ init_rtp_handler(&ff_mpegts_dynamic_handler, rtsp_st, NULL);
+ finalize_rtp_handler_init(s, rtsp_st, NULL);
+ }
+ return 0;
+}
+
/* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
static int sdp_parse_rtpmap(AVFormatContext *s,
AVStream *st, RTSPStream *rtsp_st,
}
if (par->codec_id == AV_CODEC_ID_NONE) {
- RTPDynamicProtocolHandler *handler =
+ const RTPDynamicProtocolHandler *handler =
ff_rtp_handler_find_by_name(buf, par->codec_type);
init_rtp_handler(handler, rtsp_st, st);
/* If no dynamic handler was found, check with the list of standard
codec_type = AVMEDIA_TYPE_AUDIO;
} else if (!strcmp(st_type, "video")) {
codec_type = AVMEDIA_TYPE_VIDEO;
- } else if (!strcmp(st_type, "application") || !strcmp(st_type, "text")) {
+ } else if (!strcmp(st_type, "application")) {
codec_type = AVMEDIA_TYPE_DATA;
+ } else if (!strcmp(st_type, "text")) {
+ codec_type = AVMEDIA_TYPE_SUBTITLE;
}
- if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
+ if (codec_type == AVMEDIA_TYPE_UNKNOWN ||
+ !(rt->media_type_mask & (1 << codec_type)) ||
+ rt->nb_rtsp_streams >= s->max_streams
+ ) {
s1->skip_media = 1;
return;
}
/* no corresponding stream */
if (rt->transport == RTSP_TRANSPORT_RAW) {
if (CONFIG_RTPDEC && !rt->ts)
- rt->ts = ff_mpegts_parse_open(s);
+ rt->ts = avpriv_mpegts_parse_open(s);
} else {
- RTPDynamicProtocolHandler *handler;
+ const RTPDynamicProtocolHandler *handler;
handler = ff_rtp_handler_find_by_id(
rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
init_rtp_handler(handler, rtsp_st, NULL);
rtsp_st->stream_index = st->index;
st->codecpar->codec_type = codec_type;
if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
- RTPDynamicProtocolHandler *handler;
+ const RTPDynamicProtocolHandler *handler;
/* if standard payload type, we can find the codec right now */
ff_rtp_get_codec_info(st->codecpar, rtsp_st->sdp_payload_type);
if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO &&
break;
case 'a':
if (av_strstart(p, "control:", &p)) {
- if (s->nb_streams == 0) {
+ if (rt->nb_rtsp_streams == 0) {
if (!strncmp(p, "rtsp://", 7))
av_strlcpy(rt->control_uri, p,
sizeof(rt->control_uri));
{
const char *p;
int letter, i;
- /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
- * contain long SDP lines containing complete ASF Headers (several
- * kB) or arrays of MDPR (RM stream descriptor) headers plus
- * "rulebooks" describing their properties. Therefore, the SDP line
- * buffer is large.
- *
- * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
- * in rtpdec_xiph.c. */
- char buf[16384], *q;
+ char buf[SDP_MAX_SIZE], *q;
SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
p = content;
}
for (i = 0; i < s1->nb_default_include_source_addrs; i++)
- av_free(s1->default_include_source_addrs[i]);
+ av_freep(&s1->default_include_source_addrs[i]);
av_freep(&s1->default_include_source_addrs);
for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
- av_free(s1->default_exclude_source_addrs[i]);
+ av_freep(&s1->default_exclude_source_addrs[i]);
av_freep(&s1->default_exclude_source_addrs);
return 0;
ff_rtsp_tcp_write_packet(s, rtsp_st);
ffio_free_dyn_buf(&rtpctx->pb);
} else {
- avio_close(rtpctx->pb);
+ avio_closep(&rtpctx->pb);
}
avformat_free_context(rtpctx);
} else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
ff_rtp_parse_close(rtsp_st->transport_priv);
}
rtsp_st->transport_priv = NULL;
- if (rtsp_st->rtp_handle)
- ffurl_close(rtsp_st->rtp_handle);
- rtsp_st->rtp_handle = NULL;
+ ffurl_closep(&rtsp_st->rtp_handle);
}
}
av_free(rtsp_st->dynamic_protocol_context);
}
for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
- av_free(rtsp_st->include_source_addrs[j]);
+ av_freep(&rtsp_st->include_source_addrs[j]);
av_freep(&rtsp_st->include_source_addrs);
for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
- av_free(rtsp_st->exclude_source_addrs[j]);
+ av_freep(&rtsp_st->exclude_source_addrs[j]);
av_freep(&rtsp_st->exclude_source_addrs);
- av_free(rtsp_st);
+ av_freep(&rtsp_st);
}
}
- av_free(rt->rtsp_streams);
+ av_freep(&rt->rtsp_streams);
if (rt->asf_ctx) {
avformat_close_input(&rt->asf_ctx);
}
if (CONFIG_RTPDEC && rt->ts)
- ff_mpegts_parse_close(rt->ts);
- av_freep(&rt->protocols);
- av_free(rt->p);
- av_free(rt->recvbuf);
+ avpriv_mpegts_parse_close(rt->ts);
+ av_freep(&rt->p);
+ av_freep(&rt->recvbuf);
}
int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
if (!st)
s->ctx_flags |= AVFMTCTX_NOHEADER;
- if (CONFIG_RTSP_MUXER && s->oformat) {
+ if (CONFIG_RTSP_MUXER && s->oformat && st) {
int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
s, st, rtsp_st->rtp_handle,
RTSP_TCP_MAX_PACKET_SIZE,
st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
} else if (rt->transport == RTSP_TRANSPORT_RAW) {
return 0; // Don't need to open any parser here
- } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
+ } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT && st)
rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
rtsp_st->dynamic_protocol_context,
rtsp_st->dynamic_handler);
static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
{
int read = 0;
- char key[20], value[1024], url[1024] = "";
+ char key[20], value[MAX_URL_SIZE], url[MAX_URL_SIZE] = "";
uint32_t seq = 0, rtptime = 0;
for (;;) {
} else if (av_stristart(p, "Content-Type:", &p)) {
p += strspn(p, SPACE_CHARS);
av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
+ } else if (av_stristart(p, "com.ses.streamID:", &p)) {
+ p += strspn(p, SPACE_CHARS);
+ av_strlcpy(reply->stream_id, p, sizeof(reply->stream_id));
}
}
{
RTSPState *rt = s->priv_data;
int ret, len, len1;
- uint8_t buf[1024];
+ uint8_t buf[MAX_URL_SIZE];
ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
if (ret != 3)
int return_on_interleaved_data, const char *method)
{
RTSPState *rt = s->priv_data;
- char buf[4096], buf1[1024], *q;
+ char buf[MAX_URL_SIZE], buf1[MAX_URL_SIZE], *q;
unsigned char ch;
const char *p;
int ret, content_length, line_count = 0, request = 0;
content = av_malloc(content_length + 1);
if (!content)
return AVERROR(ENOMEM);
- ffurl_read_complete(rt->rtsp_hd, content, content_length);
+ if (ffurl_read_complete(rt->rtsp_hd, content, content_length) != content_length)
+ return AVERROR(EIO);
content[content_length] = '\0';
}
if (content_ptr)
*content_ptr = content;
else
- av_free(content);
+ av_freep(&content);
if (request) {
- char buf[1024];
+ char buf[MAX_URL_SIZE];
char base64buf[AV_BASE64_SIZE(sizeof(buf))];
const char* ptr = buf;
int send_content_length)
{
RTSPState *rt = s->priv_data;
- char buf[4096], *out_buf;
+ char buf[MAX_URL_SIZE], *out_buf;
char base64buf[AV_BASE64_SIZE(sizeof(buf))];
+ if (!rt->rtsp_hd_out)
+ return AVERROR(ENOTCONN);
+
/* Add in RTSP headers */
out_buf = buf;
rt->seq++;
if (headers)
av_strlcat(buf, headers, sizeof(buf));
av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
- av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", LIBAVFORMAT_IDENT);
+ av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
if (rt->session_id[0] != '\0' && (!headers ||
!strstr(headers, "\nIf-Match:"))) {
av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
int rtx = 0, j, i, err, interleave = 0, port_off;
RTSPStream *rtsp_st;
RTSPMessageHeader reply1, *reply = &reply1;
- char cmd[2048];
+ char cmd[MAX_URL_SIZE];
const char *trans_pref;
if (rt->transport == RTSP_TRANSPORT_RDT)
/* default timeout: 1 minute */
rt->timeout = 60;
- /* for each stream, make the setup request */
- /* XXX: we assume the same server is used for the control of each
- * RTSP stream */
-
/* Choose a random starting offset within the first half of the
* port range, to allow for a number of ports to try even if the offset
* happens to be at the end of the random range. */
port_off -= port_off & 0x01;
for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
- char transport[2048];
+ char transport[MAX_URL_SIZE];
/*
* WMS serves all UDP data over a single connection, the RTX, which
"?localport=%d", j);
/* we will use two ports per rtp stream (rtp and rtcp) */
j += 2;
- err = ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
- &s->interrupt_callback, &opts, rt->protocols, NULL);
+ err = ffurl_open_whitelist(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
+ &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
av_dict_free(&opts);
if (!err)
goto rtp_opened;
}
-
av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
err = AVERROR(EIO);
goto fail;
rtp_opened:
port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
have_port:
- snprintf(transport, sizeof(transport) - 1,
- "%s/UDP;", trans_pref);
+ av_strlcpy(transport, trans_pref, sizeof(transport));
+ av_strlcat(transport,
+ rt->server_type == RTSP_SERVER_SATIP ? ";" : "/UDP;",
+ sizeof(transport));
if (rt->server_type != RTSP_SERVER_REAL)
av_strlcat(transport, "unicast;", sizeof(transport));
av_strlcatf(transport, sizeof(transport),
goto fail;
} else if (reply->status_code != RTSP_STATUS_OK ||
reply->nb_transports != 1) {
- err = AVERROR_INVALIDDATA;
+ err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
goto fail;
}
+ if (rt->server_type == RTSP_SERVER_SATIP && reply->stream_id[0]) {
+ char proto[128], host[128], path[512], auth[128];
+ int port;
+ av_url_split(proto, sizeof(proto), auth, sizeof(auth), host, sizeof(host),
+ &port, path, sizeof(path), rt->control_uri);
+ ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL, host,
+ port, "/stream=%s", reply->stream_id);
+ }
+
/* XXX: same protocol for all streams is required */
if (i > 0) {
if (reply->transports[0].lower_transport != rt->lower_transport ||
break;
case RTSP_LOWER_TRANSPORT_UDP: {
- char url[1024], options[30] = "";
+ char url[MAX_URL_SIZE], options[30] = "";
const char *peer = host;
if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
break;
}
case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
- char url[1024], namebuf[50], optbuf[20] = "";
+ char url[MAX_URL_SIZE], namebuf[50], optbuf[20] = "";
struct sockaddr_storage addr;
int port, ttl;
+ AVDictionary *opts = map_to_opts(rt);
if (reply->transports[0].destination.ss_family) {
addr = reply->transports[0].destination;
namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
port, "%s", optbuf);
- if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
- &s->interrupt_callback, NULL, rt->protocols, NULL) < 0) {
+ err = ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
+ &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
+ av_dict_free(&opts);
+
+ if (err < 0) {
err = AVERROR_INVALIDDATA;
goto fail;
}
void ff_rtsp_close_connections(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
- if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
- ffurl_close(rt->rtsp_hd);
- rt->rtsp_hd = rt->rtsp_hd_out = NULL;
+ if (rt->rtsp_hd_out != rt->rtsp_hd)
+ ffurl_closep(&rt->rtsp_hd_out);
+ rt->rtsp_hd_out = NULL;
+ ffurl_closep(&rt->rtsp_hd);
}
int ff_rtsp_connect(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
char proto[128], host[1024], path[1024];
- char tcpname[1024], cmd[2048], auth[128];
+ char tcpname[1024], cmd[MAX_URL_SIZE], auth[128];
const char *lower_rtsp_proto = "tcp";
int port, err, tcp_fd;
- RTSPMessageHeader reply1 = {0}, *reply = &reply1;
+ RTSPMessageHeader reply1, *reply = &reply1;
int lower_transport_mask = 0;
int default_port = RTSP_DEFAULT_PORT;
+ int https_tunnel = 0;
char real_challenge[64] = "";
struct sockaddr_storage peer;
socklen_t peer_len = sizeof(peer);
if (!ff_network_init())
return AVERROR(EIO);
- if (!rt->protocols) {
- rt->protocols = ffurl_get_protocols(s->protocol_whitelist,
- s->protocol_blacklist);
- if (!rt->protocols)
- return AVERROR(ENOMEM);
- }
-
if (s->max_delay < 0) /* Not set by the caller */
s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
rt->control_transport = RTSP_MODE_PLAIN;
- if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
+ if (rt->lower_transport_mask & ((1 << RTSP_LOWER_TRANSPORT_HTTP) |
+ (1 << RTSP_LOWER_TRANSPORT_HTTPS))) {
+ https_tunnel = !!(rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTPS));
rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
rt->control_transport = RTSP_MODE_TUNNEL;
}
rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
redirect:
+ memset(&reply1, 0, sizeof(reply1));
/* extract hostname and port */
av_url_split(proto, sizeof(proto), auth, sizeof(auth),
- host, sizeof(host), &port, path, sizeof(path), s->filename);
+ host, sizeof(host), &port, path, sizeof(path), s->url);
if (!strcmp(proto, "rtsps")) {
lower_rtsp_proto = "tls";
default_port = RTSPS_DEFAULT_PORT;
rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
+ } else if (!strcmp(proto, "satip")) {
+ av_strlcpy(proto, "rtsp", sizeof(proto));
+ rt->server_type = RTSP_SERVER_SATIP;
}
if (*auth) {
}
}
- /* Construct the URI used in request; this is similar to s->filename,
+ /* Construct the URI used in request; this is similar to s->url,
* but with authentication credentials removed and RTSP specific options
* stripped out. */
ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
char httpname[1024];
char sessioncookie[17];
char headers[1024];
+ AVDictionary *options = NULL;
+
+ av_dict_set_int(&options, "timeout", rt->stimeout, 0);
- ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
+ ff_url_join(httpname, sizeof(httpname), https_tunnel ? "https" : "http", auth, host, port, "%s", path);
snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
av_get_random_seed(), av_get_random_seed());
/* GET requests */
if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
- &s->interrupt_callback, rt->protocols) < 0) {
+ &s->interrupt_callback) < 0) {
err = AVERROR(EIO);
goto fail;
}
sessioncookie);
av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
+ if (!rt->rtsp_hd->protocol_whitelist && s->protocol_whitelist) {
+ rt->rtsp_hd->protocol_whitelist = av_strdup(s->protocol_whitelist);
+ if (!rt->rtsp_hd->protocol_whitelist) {
+ err = AVERROR(ENOMEM);
+ goto fail;
+ }
+ }
+
/* complete the connection */
- if (ffurl_connect(rt->rtsp_hd, NULL)) {
+ if (ffurl_connect(rt->rtsp_hd, &options)) {
+ av_dict_free(&options);
err = AVERROR(EIO);
goto fail;
}
/* POST requests */
if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
- &s->interrupt_callback, rt->protocols) < 0 ) {
+ &s->interrupt_callback) < 0 ) {
err = AVERROR(EIO);
goto fail;
}
sessioncookie);
av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
+ av_opt_set(rt->rtsp_hd_out->priv_data, "send_expect_100", "0", 0);
/* Initialize the authentication state for the POST session. The HTTP
* protocol implementation doesn't properly handle multi-pass
ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
/* complete the connection */
- if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
+ if (ffurl_connect(rt->rtsp_hd_out, &options)) {
+ av_dict_free(&options);
err = AVERROR(EIO);
goto fail;
}
+ av_dict_free(&options);
} else {
+ int ret;
/* open the tcp connection */
ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
- host, port, NULL);
- if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
- &s->interrupt_callback, NULL, rt->protocols, NULL) < 0) {
- err = AVERROR(EIO);
+ host, port,
+ "?timeout=%d", rt->stimeout);
+ if ((ret = ffurl_open_whitelist(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
+ &s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist, NULL)) < 0) {
+ err = ret;
goto fail;
}
rt->rtsp_hd_out = rt->rtsp_hd;
/* request options supported by the server; this also detects server
* type */
- for (rt->server_type = RTSP_SERVER_RTP;;) {
+ if (rt->server_type != RTSP_SERVER_SATIP)
+ rt->server_type = RTSP_SERVER_RTP;
+ for (;;) {
cmd[0] = 0;
if (rt->server_type == RTSP_SERVER_REAL)
av_strlcat(cmd,
sizeof(cmd));
ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
if (reply->status_code != RTSP_STATUS_OK) {
- err = AVERROR_INVALIDDATA;
+ err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
goto fail;
}
break;
}
- if (CONFIG_RTSP_DEMUXER && s->iformat)
- err = ff_rtsp_setup_input_streams(s, reply);
- else if (CONFIG_RTSP_MUXER)
+#if CONFIG_RTSP_DEMUXER
+ if (s->iformat) {
+ if (rt->server_type == RTSP_SERVER_SATIP)
+ err = init_satip_stream(s);
+ else
+ err = ff_rtsp_setup_input_streams(s, reply);
+ } else
+#endif
+ if (CONFIG_RTSP_MUXER)
err = ff_rtsp_setup_output_streams(s, host);
+ else
+ av_assert0(0);
if (err)
goto fail;
int lower_transport = ff_log2_tab[lower_transport_mask &
~(lower_transport_mask - 1)];
+ if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
+ && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
+ lower_transport = RTSP_LOWER_TRANSPORT_TCP;
+
err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
rt->server_type == RTSP_SERVER_REAL ?
real_challenge : NULL);
ff_rtsp_close_streams(s);
ff_rtsp_close_connections(s);
if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
- av_strlcpy(s->filename, reply->location, sizeof(s->filename));
+ char *new_url = av_strdup(reply->location);
+ if (!new_url) {
+ err = AVERROR(ENOMEM);
+ goto fail2;
+ }
+ ff_format_set_url(s, new_url);
rt->session_id[0] = '\0';
av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
reply->status_code,
- s->filename);
+ s->url);
goto redirect;
}
+ fail2:
ff_network_close();
return err;
}
if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
if (rt->state == RTSP_STATE_STREAMING) {
- if (!ff_rtsp_parse_streaming_commands(s))
- return AVERROR_EOF;
- else
- av_log(s, AV_LOG_WARNING,
- "Unable to answer to TEARDOWN\n");
+ return ff_rtsp_parse_streaming_commands(s);
} else
- return 0;
+ return AVERROR_EOF;
} else {
RTSPMessageHeader reply;
ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
{
RTSPState *rt = s->priv_data;
RTSPStream *rtsp_st;
- int n, i, ret, timeout_cnt = 0;
+ int n, i, ret;
struct pollfd *p = rt->p;
int *fds = NULL, fdsnum, fdsidx;
+ int runs = rt->initial_timeout * 1000LL / POLLING_TIME;
if (!p) {
- p = rt->p = av_malloc_array(2 * (rt->nb_rtsp_streams + 1), sizeof(struct pollfd));
+ p = rt->p = av_malloc_array(2 * rt->nb_rtsp_streams + 1, sizeof(*p));
if (!p)
return AVERROR(ENOMEM);
p[rt->max_p].fd = fds[fdsidx];
p[rt->max_p++].events = POLLIN;
}
- av_free(fds);
+ av_freep(&fds);
}
}
}
return AVERROR_EXIT;
if (wait_end && wait_end - av_gettime_relative() < 0)
return AVERROR(EAGAIN);
- n = poll(p, rt->max_p, POLL_TIMEOUT_MS);
+ n = poll(p, rt->max_p, POLLING_TIME);
if (n > 0) {
int j = rt->rtsp_hd ? 1 : 0;
- timeout_cnt = 0;
for (i = 0; i < rt->nb_rtsp_streams; i++) {
rtsp_st = rt->rtsp_streams[i];
if (rtsp_st->rtp_handle) {
}
}
#endif
- } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
+ } else if (n == 0 && rt->initial_timeout > 0 && --runs <= 0) {
return AVERROR(ETIMEDOUT);
} else if (n < 0 && errno != EINTR)
return AVERROR(errno);
} else if (rt->transport == RTSP_TRANSPORT_RTP) {
ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
} else if (CONFIG_RTPDEC && rt->ts) {
- ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
+ ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
if (ret >= 0) {
rt->recvbuf_pos += ret;
ret = rt->recvbuf_pos < rt->recvbuf_len;
st2->time_base);
}
}
+ // Make real NTP start time available in AVFormatContext
+ if (s->start_time_realtime == AV_NOPTS_VALUE) {
+ s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
+ if (rtpctx->st) {
+ s->start_time_realtime -=
+ av_rescale_q (rtpctx->rtcp_ts_offset, rtpctx->st->time_base, AV_TIME_BASE_Q);
+ }
+ }
}
if (ret == -RTCP_BYE) {
rt->nb_byes++;
}
}
} else if (CONFIG_RTPDEC && rt->ts) {
- ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
+ ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
if (ret >= 0) {
if (ret < len) {
rt->recvbuf_len = len;
#endif /* CONFIG_RTPDEC */
#if CONFIG_SDP_DEMUXER
-static int sdp_probe(AVProbeData *p1)
+static int sdp_probe(const AVProbeData *p1)
{
const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
/* we look for a line beginning "c=IN IP" */
while (p < p_end && *p != '\0') {
- if (p + sizeof("c=IN IP") - 1 < p_end &&
+ if (sizeof("c=IN IP") - 1 < p_end - p &&
av_strstart(p, "c=IN IP", NULL))
return AVPROBE_SCORE_EXTENSION;
RTSPStream *rtsp_st;
int size, i, err;
char *content;
- char url[1024];
+ char url[MAX_URL_SIZE];
if (!ff_network_init())
return AVERROR(EIO);
- if (!rt->protocols) {
- rt->protocols = ffurl_get_protocols(s->protocol_whitelist,
- s->protocol_blacklist);
- if (!rt->protocols)
- return AVERROR(ENOMEM);
- }
-
if (s->max_delay < 0) /* Not set by the caller */
s->max_delay = DEFAULT_REORDERING_DELAY;
if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
/* read the whole sdp file */
/* XXX: better loading */
content = av_malloc(SDP_MAX_SIZE);
- if (!content)
+ if (!content) {
+ ff_network_close();
return AVERROR(ENOMEM);
+ }
size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
if (size <= 0) {
av_free(content);
+ ff_network_close();
return AVERROR_INVALIDDATA;
}
content[size] ='\0';
err = ff_sdp_parse(s, content);
- av_free(content);
+ av_freep(&content);
if (err) goto fail;
/* open each RTP stream */
append_source_addrs(url, sizeof(url), "block",
rtsp_st->nb_exclude_source_addrs,
rtsp_st->exclude_source_addrs);
- err = ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
- &s->interrupt_callback, &opts, rt->protocols, NULL);
+ err = ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ,
+ &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
av_dict_free(&opts);
.version = LIBAVUTIL_VERSION_INT,
};
-AVInputFormat ff_sdp_demuxer = {
+const AVInputFormat ff_sdp_demuxer = {
.name = "sdp",
.long_name = NULL_IF_CONFIG_SMALL("SDP"),
.priv_data_size = sizeof(RTSPState),
#endif /* CONFIG_SDP_DEMUXER */
#if CONFIG_RTP_DEMUXER
-static int rtp_probe(AVProbeData *p)
+static int rtp_probe(const AVProbeData *p)
{
if (av_strstart(p->filename, "rtp:", NULL))
return AVPROBE_SCORE_MAX;
static int rtp_read_header(AVFormatContext *s)
{
uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
- char host[500], sdp[500];
+ char host[500], filters_buf[1000];
int ret, port;
URLContext* in = NULL;
int payload_type;
AVIOContext pb;
socklen_t addrlen = sizeof(addr);
RTSPState *rt = s->priv_data;
+ const char *p;
+ AVBPrint sdp;
+ AVDictionary *opts = NULL;
if (!ff_network_init())
return AVERROR(EIO);
- if (!rt->protocols) {
- rt->protocols = ffurl_get_protocols(s->protocol_whitelist,
- s->protocol_blacklist);
- if (!rt->protocols)
- return AVERROR(ENOMEM);
- }
-
- ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
- &s->interrupt_callback, NULL, rt->protocols, NULL);
+ opts = map_to_opts(rt);
+ ret = ffurl_open_whitelist(&in, s->url, AVIO_FLAG_READ,
+ &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
+ av_dict_free(&opts);
if (ret)
goto fail;
break;
}
getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
- ffurl_close(in);
- in = NULL;
+ ffurl_closep(&in);
par = avcodec_parameters_alloc();
if (!par) {
av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
"without an SDP file describing it\n",
payload_type);
+ ret = AVERROR_INVALIDDATA;
goto fail;
}
if (par->codec_type != AVMEDIA_TYPE_DATA) {
}
av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
- NULL, 0, s->filename);
-
- snprintf(sdp, sizeof(sdp),
- "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
- addr.ss_family == AF_INET ? 4 : 6, host,
- par->codec_type == AVMEDIA_TYPE_DATA ? "application" :
- par->codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
- port, payload_type);
- av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
+ NULL, 0, s->url);
+
+ av_bprint_init(&sdp, 0, AV_BPRINT_SIZE_UNLIMITED);
+ av_bprintf(&sdp, "v=0\r\nc=IN IP%d %s\r\n",
+ addr.ss_family == AF_INET ? 4 : 6, host);
+
+ p = strchr(s->url, '?');
+ if (p) {
+ static const char filters[][2][8] = { { "sources", "incl" },
+ { "block", "excl" } };
+ int i;
+ char *q;
+ for (i = 0; i < FF_ARRAY_ELEMS(filters); i++) {
+ if (av_find_info_tag(filters_buf, sizeof(filters_buf), filters[i][0], p)) {
+ q = filters_buf;
+ while ((q = strchr(q, ',')) != NULL)
+ *q = ' ';
+ av_bprintf(&sdp, "a=source-filter:%s IN IP%d %s %s\r\n",
+ filters[i][1],
+ addr.ss_family == AF_INET ? 4 : 6, host,
+ filters_buf);
+ }
+ }
+ }
+
+ av_bprintf(&sdp, "m=%s %d RTP/AVP %d\r\n",
+ par->codec_type == AVMEDIA_TYPE_DATA ? "application" :
+ par->codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
+ port, payload_type);
+ av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp.str);
+ if (!av_bprint_is_complete(&sdp))
+ goto fail_nobuf;
avcodec_parameters_free(&par);
- ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
+ ffio_init_context(&pb, sdp.str, sdp.len, 0, NULL, NULL, NULL, NULL);
s->pb = &pb;
- /* sdp_read_header initializes this again */
+ /* if sdp_read_header() fails then following ff_network_close() cancels out */
+ /* ff_network_init() at the start of this function. Otherwise it cancels out */
+ /* ff_network_init() inside sdp_read_header() */
ff_network_close();
- rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
+ rt->media_type_mask = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1;
ret = sdp_read_header(s);
s->pb = NULL;
+ av_bprint_finalize(&sdp, NULL);
return ret;
+fail_nobuf:
+ ret = AVERROR(ENOMEM);
+ av_log(s, AV_LOG_ERROR, "rtp_read_header(): not enough buffer space for sdp-headers\n");
+ av_bprint_finalize(&sdp, NULL);
fail:
avcodec_parameters_free(&par);
- if (in)
- ffurl_close(in);
+ ffurl_closep(&in);
ff_network_close();
return ret;
}
.version = LIBAVUTIL_VERSION_INT,
};
-AVInputFormat ff_rtp_demuxer = {
+const AVInputFormat ff_rtp_demuxer = {
.name = "rtp",
.long_name = NULL_IF_CONFIG_SMALL("RTP input"),
.priv_data_size = sizeof(RTSPState),
projects / ffmpeg / blobdiff