* RTSP/SDP client
* Copyright (c) 2002 Fabrice Bellard.
*
- * This library is free software; you can redistribute it and/or
+ * 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 of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
*
- * This library 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 this library; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ * 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"
#include <sys/time.h>
-#include <netinet/in.h>
-#include <sys/socket.h>
-#include <ctype.h>
-#ifndef __BEOS__
-# include <arpa/inet.h>
-#else
-# include "barpainet.h"
-#endif
+#include <unistd.h> /* for select() prototype */
+#include "network.h"
+
+#include "rtp_internal.h"
//#define DEBUG
+//#define DEBUG_RTP_TCP
+
+enum RTSPClientState {
+ RTSP_STATE_IDLE,
+ RTSP_STATE_PLAYING,
+ RTSP_STATE_PAUSED,
+};
typedef struct RTSPState {
URLContext *rtsp_hd; /* RTSP TCP connexion handle */
- ByteIOContext rtsp_gb;
+ int nb_rtsp_streams;
+ struct RTSPStream **rtsp_streams;
+
+ enum RTSPClientState state;
+ int64_t seek_timestamp;
+
+ /* XXX: currently we use unbuffered input */
+ // ByteIOContext rtsp_gb;
int seq; /* RTSP command sequence number */
char session_id[512];
enum RTSPProtocol protocol;
char last_reply[2048]; /* XXX: allocate ? */
+ RTPDemuxContext *cur_rtp;
} RTSPState;
typedef struct RTSPStream {
- AVFormatContext *ic;
+ URLContext *rtp_handle; /* RTP stream handle */
+ RTPDemuxContext *rtp_ctx; /* RTP parse context */
+
+ int stream_index; /* corresponding stream index, if any. -1 if none (MPEG2TS case) */
int interleaved_min, interleaved_max; /* interleave ids, if TCP transport */
char control_url[1024]; /* url for this stream (from SDP) */
struct in_addr sdp_ip; /* IP address (from SDP content - not used in RTSP) */
int sdp_ttl; /* IP TTL (from SDP content - not used in RTSP) */
int sdp_payload_type; /* payload type - only used in SDP */
+ rtp_payload_data_t rtp_payload_data; /* rtp payload parsing infos from SDP */
+
+ RTPDynamicProtocolHandler *dynamic_handler; ///< Only valid if it's a dynamic protocol. (This is the handler structure)
+ void *dynamic_protocol_context; ///< Only valid if it's a dynamic protocol. (This is any private data associated with the dynamic protocol)
} RTSPStream;
-/* suppress this hack */
-int rtsp_abort_req = 0;
+static int rtsp_read_play(AVFormatContext *s);
/* XXX: currently, the only way to change the protocols consists in
changing this variable */
-int rtsp_default_protocols = (1 << RTSP_PROTOCOL_RTP_TCP) | (1 << RTSP_PROTOCOL_RTP_UDP) | (1 << RTSP_PROTOCOL_RTP_UDP_MULTICAST);
-/* if non zero, then set a range for RTP ports */
-int rtsp_rtp_port_min = 0;
-int rtsp_rtp_port_max = 0;
-
-FFRTSPCallback *ff_rtsp_callback = NULL;
+int rtsp_default_protocols = (1 << RTSP_PROTOCOL_RTP_UDP);
static int rtsp_probe(AVProbeData *p)
{
*pp = p;
}
-static void get_word_sep(char *buf, int buf_size, const char *sep,
+static void get_word_sep(char *buf, int buf_size, const char *sep,
const char **pp)
{
const char *p;
char *q;
p = *pp;
+ if (*p == '/')
+ p++;
skip_spaces(&p);
q = buf;
while (!strchr(sep, *p) && *p != '\0') {
/* parse the rtpmap description: <codec_name>/<clock_rate>[/<other
params>] */
-static int sdp_parse_rtpmap(AVCodecContext *codec, const char *p)
+static int sdp_parse_rtpmap(AVCodecContext *codec, RTSPStream *rtsp_st, int payload_type, const char *p)
{
char buf[256];
+ int i;
+ AVCodec *c;
+ const char *c_name;
- /* codec name */
+ /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
+ see if we can handle this kind of payload */
get_word_sep(buf, sizeof(buf), "/", &p);
- if (!strcmp(buf, "MP4V-ES")) {
- codec->codec_id = CODEC_ID_MPEG4;
- return 0;
+ if (payload_type >= RTP_PT_PRIVATE) {
+ RTPDynamicProtocolHandler *handler= RTPFirstDynamicPayloadHandler;
+ while(handler) {
+ if (!strcmp(buf, handler->enc_name) && (codec->codec_type == handler->codec_type)) {
+ codec->codec_id = handler->codec_id;
+ rtsp_st->dynamic_handler= handler;
+ if(handler->open) {
+ rtsp_st->dynamic_protocol_context= handler->open();
+ }
+ break;
+ }
+ handler= handler->next;
+ }
} else {
- return -1;
+ /* We are in a standard case ( from http://www.iana.org/assignments/rtp-parameters) */
+ /* search into AVRtpPayloadTypes[] */
+ for (i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i)
+ if (!strcmp(buf, AVRtpPayloadTypes[i].enc_name) && (codec->codec_type == AVRtpPayloadTypes[i].codec_type)){
+ codec->codec_id = AVRtpPayloadTypes[i].codec_id;
+ break;
+ }
}
+
+ c = avcodec_find_decoder(codec->codec_id);
+ if (c && c->name)
+ c_name = c->name;
+ else
+ c_name = (char *)NULL;
+
+ if (c_name) {
+ get_word_sep(buf, sizeof(buf), "/", &p);
+ i = atoi(buf);
+ switch (codec->codec_type) {
+ case CODEC_TYPE_AUDIO:
+ av_log(codec, AV_LOG_DEBUG, " audio codec set to : %s\n", c_name);
+ codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
+ codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
+ if (i > 0) {
+ codec->sample_rate = i;
+ get_word_sep(buf, sizeof(buf), "/", &p);
+ i = atoi(buf);
+ if (i > 0)
+ codec->channels = i;
+ // TODO: there is a bug here; if it is a mono stream, and less than 22000Hz, faad upconverts to stereo and twice the
+ // frequency. No problem, but the sample rate is being set here by the sdp line. Upcoming patch forthcoming. (rdm)
+ }
+ av_log(codec, AV_LOG_DEBUG, " audio samplerate set to : %i\n", codec->sample_rate);
+ av_log(codec, AV_LOG_DEBUG, " audio channels set to : %i\n", codec->channels);
+ break;
+ case CODEC_TYPE_VIDEO:
+ av_log(codec, AV_LOG_DEBUG, " video codec set to : %s\n", c_name);
+ break;
+ default:
+ break;
+ }
+ return 0;
+ }
+
+ return -1;
}
/* return the length and optionnaly the data */
return len;
}
-static void sdp_parse_fmtp(AVCodecContext *codec, const char *p)
+static void sdp_parse_fmtp_config(AVCodecContext *codec, char *attr, char *value)
{
- char attr[256];
- char value[4096];
- int len;
-
- /* loop on each attribute */
- for(;;) {
- skip_spaces(&p);
- if (*p == '\0')
- break;
- get_word_sep(attr, sizeof(attr), "=", &p);
- if (*p == '=')
- p++;
- get_word_sep(value, sizeof(value), ";", &p);
- if (*p == ';')
- p++;
- /* handle MPEG4 video */
- switch(codec->codec_id) {
+ switch (codec->codec_id) {
case CODEC_ID_MPEG4:
+ case CODEC_ID_AAC:
if (!strcmp(attr, "config")) {
/* decode the hexa encoded parameter */
- len = hex_to_data(NULL, value);
- codec->extradata = av_mallocz(len);
+ int len = hex_to_data(NULL, value);
+ codec->extradata = av_mallocz(len + FF_INPUT_BUFFER_PADDING_SIZE);
if (!codec->extradata)
- goto fail;
+ return;
codec->extradata_size = len;
hex_to_data(codec->extradata, value);
}
break;
default:
- /* ignore data for other codecs */
break;
+ }
+ return;
+}
+
+typedef struct attrname_map
+{
+ const char *str;
+ uint16_t type;
+ uint32_t offset;
+} attrname_map_t;
+
+/* All known fmtp parmeters and the corresping RTPAttrTypeEnum */
+#define ATTR_NAME_TYPE_INT 0
+#define ATTR_NAME_TYPE_STR 1
+static attrname_map_t attr_names[]=
+{
+ {"SizeLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, sizelength)},
+ {"IndexLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, indexlength)},
+ {"IndexDeltaLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, indexdeltalength)},
+ {"profile-level-id", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, profile_level_id)},
+ {"StreamType", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, streamtype)},
+ {"mode", ATTR_NAME_TYPE_STR, offsetof(rtp_payload_data_t, mode)},
+ {NULL, -1, -1},
+};
+
+/** parse the attribute line from the fmtp a line of an sdp resonse. This is broken out as a function
+* because it is used in rtp_h264.c, which is forthcoming.
+*/
+int rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size)
+{
+ skip_spaces(p);
+ if(**p)
+ {
+ get_word_sep(attr, attr_size, "=", p);
+ if (**p == '=')
+ (*p)++;
+ get_word_sep(value, value_size, ";", p);
+ if (**p == ';')
+ (*p)++;
+ return 1;
+ }
+ return 0;
+}
+
+/* parse a SDP line and save stream attributes */
+static void sdp_parse_fmtp(AVStream *st, const char *p)
+{
+ char attr[256];
+ char value[4096];
+ int i;
+
+ RTSPStream *rtsp_st = st->priv_data;
+ AVCodecContext *codec = st->codec;
+ rtp_payload_data_t *rtp_payload_data = &rtsp_st->rtp_payload_data;
+
+ /* loop on each attribute */
+ while(rtsp_next_attr_and_value(&p, attr, sizeof(attr), value, sizeof(value)))
+ {
+ /* grab the codec extra_data from the config parameter of the fmtp line */
+ sdp_parse_fmtp_config(codec, attr, value);
+ /* Looking for a known attribute */
+ for (i = 0; attr_names[i].str; ++i) {
+ if (!strcasecmp(attr, attr_names[i].str)) {
+ if (attr_names[i].type == ATTR_NAME_TYPE_INT)
+ *(int *)((char *)rtp_payload_data + attr_names[i].offset) = atoi(value);
+ else if (attr_names[i].type == ATTR_NAME_TYPE_STR)
+ *(char **)((char *)rtp_payload_data + attr_names[i].offset) = av_strdup(value);
+ }
}
- fail: ;
- // printf("'%s' = '%s'\n", attr, value);
}
}
+/** Parse a string \p in the form of Range:npt=xx-xx, and determine the start
+ * and end time.
+ * Used for seeking in the rtp stream.
+ */
+static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
+{
+ char buf[256];
+
+ skip_spaces(&p);
+ if (!stristart(p, "npt=", &p))
+ return;
+
+ *start = AV_NOPTS_VALUE;
+ *end = AV_NOPTS_VALUE;
+
+ get_word_sep(buf, sizeof(buf), "-", &p);
+ *start = parse_date(buf, 1);
+ if (*p == '-') {
+ p++;
+ get_word_sep(buf, sizeof(buf), "-", &p);
+ *end = parse_date(buf, 1);
+ }
+// av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
+// av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
+}
+
typedef struct SDPParseState {
/* SDP only */
struct in_addr default_ip;
static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
int letter, const char *buf)
{
+ RTSPState *rt = s->priv_data;
char buf1[64], st_type[64];
const char *p;
int codec_type, payload_type, i;
rtsp_st = av_mallocz(sizeof(RTSPStream));
if (!rtsp_st)
return;
- st = av_new_stream(s, s->nb_streams);
- if (!st)
- return;
- st->priv_data = rtsp_st;
+ rtsp_st->stream_index = -1;
+ dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
rtsp_st->sdp_ip = s1->default_ip;
rtsp_st->sdp_ttl = s1->default_ttl;
- st->codec.codec_type = codec_type;
-
get_word(buf1, sizeof(buf1), &p); /* port */
rtsp_st->sdp_port = atoi(buf1);
get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */
-
+
/* XXX: handle list of formats */
get_word(buf1, sizeof(buf1), &p); /* format list */
rtsp_st->sdp_payload_type = atoi(buf1);
- if (rtsp_st->sdp_payload_type < 96) {
- /* if standard payload type, we can find the codec right now */
- rtp_get_codec_info(&st->codec, rtsp_st->sdp_payload_type);
- }
+ if (!strcmp(AVRtpPayloadTypes[rtsp_st->sdp_payload_type].enc_name, "MP2T")) {
+ /* no corresponding stream */
+ } else {
+ st = av_new_stream(s, 0);
+ if (!st)
+ return;
+ st->priv_data = rtsp_st;
+ rtsp_st->stream_index = st->index;
+ st->codec->codec_type = codec_type;
+ if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
+ /* if standard payload type, we can find the codec right now */
+ rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
+ }
+ }
/* put a default control url */
pstrcpy(rtsp_st->control_url, sizeof(rtsp_st->control_url), s->filename);
break;
/* get the control url */
st = s->streams[s->nb_streams - 1];
rtsp_st = st->priv_data;
-
+
/* XXX: may need to add full url resolution */
- url_split(proto, sizeof(proto), NULL, 0, NULL, NULL, 0, p);
+ url_split(proto, sizeof(proto), NULL, 0, NULL, 0, NULL, NULL, 0, p);
if (proto[0] == '\0') {
/* relative control URL */
pstrcat(rtsp_st->control_url, sizeof(rtsp_st->control_url), "/");
}
} else if (strstart(p, "rtpmap:", &p)) {
/* NOTE: rtpmap is only supported AFTER the 'm=' tag */
- get_word(buf1, sizeof(buf1), &p);
+ get_word(buf1, sizeof(buf1), &p);
payload_type = atoi(buf1);
for(i = 0; i < s->nb_streams;i++) {
st = s->streams[i];
rtsp_st = st->priv_data;
if (rtsp_st->sdp_payload_type == payload_type) {
- sdp_parse_rtpmap(&st->codec, p);
+ sdp_parse_rtpmap(st->codec, rtsp_st, payload_type, p);
}
}
} else if (strstart(p, "fmtp:", &p)) {
/* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
- get_word(buf1, sizeof(buf1), &p);
+ get_word(buf1, sizeof(buf1), &p);
payload_type = atoi(buf1);
for(i = 0; i < s->nb_streams;i++) {
st = s->streams[i];
rtsp_st = st->priv_data;
if (rtsp_st->sdp_payload_type == payload_type) {
- sdp_parse_fmtp(&st->codec, p);
+ if(rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->parse_sdp_a_line) {
+ if(!rtsp_st->dynamic_handler->parse_sdp_a_line(st, rtsp_st->dynamic_protocol_context, buf)) {
+ sdp_parse_fmtp(st, p);
+ }
+ } else {
+ sdp_parse_fmtp(st, p);
+ }
}
}
+ } else if(strstart(p, "framesize:", &p)) {
+ // let dynamic protocol handlers have a stab at the line.
+ get_word(buf1, sizeof(buf1), &p);
+ payload_type = atoi(buf1);
+ for(i = 0; i < s->nb_streams;i++) {
+ st = s->streams[i];
+ rtsp_st = st->priv_data;
+ if (rtsp_st->sdp_payload_type == payload_type) {
+ if(rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->parse_sdp_a_line) {
+ rtsp_st->dynamic_handler->parse_sdp_a_line(st, rtsp_st->dynamic_protocol_context, buf);
+ }
+ }
+ }
+ } else if(strstart(p, "range:", &p)) {
+ int64_t start, end;
+
+ // this is so that seeking on a streamed file can work.
+ rtsp_parse_range_npt(p, &start, &end);
+ s->start_time= start;
+ s->duration= (end==AV_NOPTS_VALUE)?AV_NOPTS_VALUE:end-start; // AV_NOPTS_VALUE means live broadcast (and can't seek)
}
break;
}
}
-int sdp_parse(AVFormatContext *s, const char *content)
+static int sdp_parse(AVFormatContext *s, const char *content)
{
const char *p;
int letter;
char buf[1024], *q;
SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
-
+
memset(s1, 0, sizeof(SDPParseState));
p = content;
for(;;) {
p++;
/* get the content */
q = buf;
- while (*p != '\n' && *p != '\0') {
+ while (*p != '\n' && *p != '\r' && *p != '\0') {
if ((q - buf) < sizeof(buf) - 1)
*q++ = *p;
p++;
char parameter[16];
RTSPTransportField *th;
char buf[256];
-
+
reply->nb_transports = 0;
-
+
for(;;) {
skip_spaces(&p);
if (*p == '\0')
th = &reply->transports[reply->nb_transports];
- get_word_sep(transport_protocol, sizeof(transport_protocol),
+ get_word_sep(transport_protocol, sizeof(transport_protocol),
"/", &p);
if (*p == '/')
p++;
get_word_sep(profile, sizeof(profile), "/;,", &p);
lower_transport[0] = '\0';
if (*p == '/') {
- get_word_sep(lower_transport, sizeof(lower_transport),
+ p++;
+ get_word_sep(lower_transport, sizeof(lower_transport),
";,", &p);
}
- if (!strcmp(lower_transport, "TCP"))
+ if (!strcasecmp(lower_transport, "TCP"))
th->protocol = RTSP_PROTOCOL_RTP_TCP;
else
th->protocol = RTSP_PROTOCOL_RTP_UDP;
-
+
if (*p == ';')
p++;
/* get each parameter */
} else if (!strcmp(parameter, "client_port")) {
if (*p == '=') {
p++;
- rtsp_parse_range(&th->client_port_min,
+ rtsp_parse_range(&th->client_port_min,
&th->client_port_max, &p);
}
} else if (!strcmp(parameter, "server_port")) {
if (*p == '=') {
p++;
- rtsp_parse_range(&th->server_port_min,
+ rtsp_parse_range(&th->server_port_min,
&th->server_port_max, &p);
}
} else if (!strcmp(parameter, "interleaved")) {
if (*p == '=') {
p++;
- rtsp_parse_range(&th->interleaved_min,
+ rtsp_parse_range(&th->interleaved_min,
&th->interleaved_max, &p);
}
} else if (!strcmp(parameter, "multicast")) {
if (*p == '=') {
p++;
get_word_sep(buf, sizeof(buf), ";,", &p);
- if (inet_aton(buf, &ipaddr))
+ if (inet_aton(buf, &ipaddr))
th->destination = ntohl(ipaddr.s_addr);
}
}
rtsp_parse_transport(reply, p);
} else if (stristart(p, "CSeq:", &p)) {
reply->seq = strtol(p, NULL, 10);
+ } else if (stristart(p, "Range:", &p)) {
+ rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
+ }
+}
+
+static int url_readbuf(URLContext *h, unsigned char *buf, int size)
+{
+ int ret, len;
+
+ len = 0;
+ while (len < size) {
+ ret = url_read(h, buf+len, size-len);
+ if (ret < 1)
+ return ret;
+ len += ret;
}
+ return len;
}
+/* skip a RTP/TCP interleaved packet */
+static void rtsp_skip_packet(AVFormatContext *s)
+{
+ RTSPState *rt = s->priv_data;
+ int ret, len, len1;
+ uint8_t buf[1024];
+
+ ret = url_readbuf(rt->rtsp_hd, buf, 3);
+ if (ret != 3)
+ return;
+ len = (buf[1] << 8) | buf[2];
+#ifdef DEBUG
+ printf("skipping RTP packet len=%d\n", len);
+#endif
+ /* skip payload */
+ while (len > 0) {
+ len1 = len;
+ if (len1 > sizeof(buf))
+ len1 = sizeof(buf);
+ ret = url_readbuf(rt->rtsp_hd, buf, len1);
+ if (ret != len1)
+ return;
+ len -= len1;
+ }
+}
-static void rtsp_send_cmd(AVFormatContext *s,
- const char *cmd, RTSPHeader *reply,
+static void rtsp_send_cmd(AVFormatContext *s,
+ const char *cmd, RTSPHeader *reply,
unsigned char **content_ptr)
{
RTSPState *rt = s->priv_data;
rt->seq++;
pstrcpy(buf, sizeof(buf), cmd);
- snprintf(buf1, sizeof(buf1), "CSeq: %d\n", rt->seq);
+ snprintf(buf1, sizeof(buf1), "CSeq: %d\r\n", rt->seq);
pstrcat(buf, sizeof(buf), buf1);
if (rt->session_id[0] != '\0' && !strstr(cmd, "\nIf-Match:")) {
- snprintf(buf1, sizeof(buf1), "Session: %s\n", rt->session_id);
+ snprintf(buf1, sizeof(buf1), "Session: %s\r\n", rt->session_id);
pstrcat(buf, sizeof(buf), buf1);
}
- pstrcat(buf, sizeof(buf), "\n");
+ pstrcat(buf, sizeof(buf), "\r\n");
#ifdef DEBUG
printf("Sending:\n%s--\n", buf);
#endif
for(;;) {
q = buf;
for(;;) {
- if (url_read(rt->rtsp_hd, &ch, 1) == 0)
+ if (url_readbuf(rt->rtsp_hd, &ch, 1) != 1)
break;
if (ch == '\n')
break;
- if (ch != '\r') {
+ if (ch == '$') {
+ /* XXX: only parse it if first char on line ? */
+ rtsp_skip_packet(s);
+ } else if (ch != '\r') {
if ((q - buf) < sizeof(buf) - 1)
*q++ = ch;
}
}
line_count++;
}
-
+
if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
pstrcpy(rt->session_id, sizeof(rt->session_id), reply->session_id);
-
+
content_length = reply->content_length;
if (content_length > 0) {
/* leave some room for a trailing '\0' (useful for simple parsing) */
content = av_malloc(content_length + 1);
- url_read(rt->rtsp_hd, content, content_length);
+ (void)url_readbuf(rt->rtsp_hd, content, content_length);
content[content_length] = '\0';
}
if (content_ptr)
*content_ptr = content;
}
-/* useful for modules: set RTSP callback function */
-void rtsp_set_callback(FFRTSPCallback *rtsp_cb)
+/* close and free RTSP streams */
+static void rtsp_close_streams(RTSPState *rt)
{
- ff_rtsp_callback = rtsp_cb;
-}
+ int i;
+ RTSPStream *rtsp_st;
+ for(i=0;i<rt->nb_rtsp_streams;i++) {
+ rtsp_st = rt->rtsp_streams[i];
+ if (rtsp_st) {
+ if (rtsp_st->rtp_ctx)
+ rtp_parse_close(rtsp_st->rtp_ctx);
+ if (rtsp_st->rtp_handle)
+ url_close(rtsp_st->rtp_handle);
+ if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
+ rtsp_st->dynamic_handler->close(rtsp_st->dynamic_protocol_context);
+ }
+ av_free(rtsp_st);
+ }
+ av_free(rt->rtsp_streams);
+}
static int rtsp_read_header(AVFormatContext *s,
AVFormatParameters *ap)
{
RTSPState *rt = s->priv_data;
- char host[1024], path[1024], tcpname[1024], cmd[2048];
+ char host[1024], path[1024], tcpname[1024], cmd[2048], *option_list, *option;
URLContext *rtsp_hd;
- int port, i, ret, err;
+ int port, i, j, ret, err;
RTSPHeader reply1, *reply = &reply1;
unsigned char *content = NULL;
- AVStream *st;
RTSPStream *rtsp_st;
- int protocol_mask;
+ int protocol_mask = 0;
+ AVStream *st;
- rtsp_abort_req = 0;
-
/* extract hostname and port */
- url_split(NULL, 0,
+ url_split(NULL, 0, NULL, 0,
host, sizeof(host), &port, path, sizeof(path), s->filename);
if (port < 0)
port = RTSP_DEFAULT_PORT;
+ /* search for options */
+ option_list = strchr(path, '?');
+ if (option_list) {
+ /* remove the options from the path */
+ *option_list++ = 0;
+ while(option_list) {
+ /* move the option pointer */
+ option = option_list;
+ option_list = strchr(option_list, '&');
+ if (option_list)
+ *(option_list++) = 0;
+ /* handle the options */
+ if (strcmp(option, "udp") == 0)
+ protocol_mask = (1<< RTSP_PROTOCOL_RTP_UDP);
+ else if (strcmp(option, "multicast") == 0)
+ protocol_mask = (1<< RTSP_PROTOCOL_RTP_UDP_MULTICAST);
+ else if (strcmp(option, "tcp") == 0)
+ protocol_mask = (1<< RTSP_PROTOCOL_RTP_TCP);
+ }
+ }
+
+ if (!protocol_mask)
+ protocol_mask = rtsp_default_protocols;
+
/* open the tcp connexion */
snprintf(tcpname, sizeof(tcpname), "tcp://%s:%d", host, port);
if (url_open(&rtsp_hd, tcpname, URL_RDWR) < 0)
return AVERROR_IO;
rt->rtsp_hd = rtsp_hd;
rt->seq = 0;
-
+
/* describe the stream */
- snprintf(cmd, sizeof(cmd),
- "DESCRIBE %s RTSP/1.0\n"
- "Accept: application/sdp\n",
+ snprintf(cmd, sizeof(cmd),
+ "DESCRIBE %s RTSP/1.0\r\n"
+ "Accept: application/sdp\r\n",
s->filename);
rtsp_send_cmd(s, cmd, reply, &content);
if (!content) {
err = AVERROR_INVALIDDATA;
goto fail;
}
-
+
/* now we got the SDP description, we parse it */
ret = sdp_parse(s, (const char *)content);
av_freep(&content);
err = AVERROR_INVALIDDATA;
goto fail;
}
-
- protocol_mask = rtsp_default_protocols;
/* for each stream, make the setup request */
/* XXX: we assume the same server is used for the control of each
RTSP stream */
- for(i=0;i<s->nb_streams;i++) {
+
+ for(j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
char transport[2048];
- AVInputFormat *fmt;
- st = s->streams[i];
- rtsp_st = st->priv_data;
+ rtsp_st = rt->rtsp_streams[i];
/* compute available transports */
transport[0] = '\0';
/* RTP/UDP */
if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP)) {
char buf[256];
- int j;
/* first try in specified port range */
- if (rtsp_rtp_port_min != 0) {
- for(j=rtsp_rtp_port_min;j<=rtsp_rtp_port_max;j++) {
+ if (RTSP_RTP_PORT_MIN != 0) {
+ while(j <= RTSP_RTP_PORT_MAX) {
snprintf(buf, sizeof(buf), "rtp://?localport=%d", j);
- if (!av_open_input_file(&rtsp_st->ic, buf,
- &rtp_demux, 0, NULL))
+ if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0) {
+ j += 2; /* we will use two port by rtp stream (rtp and rtcp) */
goto rtp_opened;
+ }
}
}
- /* then try on any port */
- if (av_open_input_file(&rtsp_st->ic, "rtp://",
- &rtp_demux, 0, NULL) < 0) {
- err = AVERROR_INVALIDDATA;
- goto fail;
- }
+/* then try on any port
+** if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
+** err = AVERROR_INVALIDDATA;
+** goto fail;
+** }
+*/
rtp_opened:
- port = rtp_get_local_port(url_fileno(&rtsp_st->ic->pb));
+ port = rtp_get_local_port(rtsp_st->rtp_handle);
if (transport[0] != '\0')
pstrcat(transport, sizeof(transport), ",");
snprintf(transport + strlen(transport), sizeof(transport) - strlen(transport) - 1,
}
/* RTP/TCP */
- if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_TCP)) {
+ else if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_TCP)) {
if (transport[0] != '\0')
pstrcat(transport, sizeof(transport), ",");
snprintf(transport + strlen(transport), sizeof(transport) - strlen(transport) - 1,
"RTP/AVP/TCP");
}
- if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP_MULTICAST)) {
+ else if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP_MULTICAST)) {
if (transport[0] != '\0')
pstrcat(transport, sizeof(transport), ",");
- snprintf(transport + strlen(transport),
+ snprintf(transport + strlen(transport),
sizeof(transport) - strlen(transport) - 1,
"RTP/AVP/UDP;multicast");
}
-
- snprintf(cmd, sizeof(cmd),
- "SETUP %s RTSP/1.0\n"
- "Transport: %s\n",
+ snprintf(cmd, sizeof(cmd),
+ "SETUP %s RTSP/1.0\r\n"
+ "Transport: %s\r\n",
rtsp_st->control_url, transport);
rtsp_send_cmd(s, cmd, reply, NULL);
if (reply->status_code != RTSP_STATUS_OK ||
/* close RTP connection if not choosen */
if (reply->transports[0].protocol != RTSP_PROTOCOL_RTP_UDP &&
(protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP))) {
- av_close_input_file(rtsp_st->ic);
- rtsp_st->ic = NULL;
+ url_close(rtsp_st->rtp_handle);
+ rtsp_st->rtp_handle = NULL;
}
switch(reply->transports[0].protocol) {
case RTSP_PROTOCOL_RTP_TCP:
- fmt = &rtp_demux;
- if (av_open_input_file(&rtsp_st->ic, "null", fmt, 0, NULL) < 0) {
- err = AVERROR_INVALIDDATA;
- goto fail;
- }
rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
break;
-
+
case RTSP_PROTOCOL_RTP_UDP:
{
char url[1024];
-
+
/* XXX: also use address if specified */
- snprintf(url, sizeof(url), "rtp://%s:%d",
+ snprintf(url, sizeof(url), "rtp://%s:%d",
host, reply->transports[0].server_port_min);
- if (rtp_set_remote_url(url_fileno(&rtsp_st->ic->pb), url) < 0) {
+ if (rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
err = AVERROR_INVALIDDATA;
goto fail;
}
char url[1024];
int ttl;
- fmt = &rtp_demux;
ttl = reply->transports[0].ttl;
if (!ttl)
ttl = 16;
- snprintf(url, sizeof(url), "rtp://%s:%d?multicast=1&ttl=%d",
- host,
+ snprintf(url, sizeof(url), "rtp://%s:%d?multicast=1&ttl=%d",
+ host,
reply->transports[0].server_port_min,
ttl);
- if (av_open_input_file(&rtsp_st->ic, url, fmt, 0, NULL) < 0) {
+ if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
err = AVERROR_INVALIDDATA;
goto fail;
}
}
break;
}
- }
-
- /* use callback if available to extend setup */
- if (ff_rtsp_callback) {
- if (ff_rtsp_callback(RTSP_ACTION_CLIENT_SETUP, rt->session_id,
- NULL, 0, rt->last_reply) < 0) {
- err = AVERROR_INVALIDDATA;
+ /* open the RTP context */
+ st = NULL;
+ if (rtsp_st->stream_index >= 0)
+ st = s->streams[rtsp_st->stream_index];
+ if (!st)
+ s->ctx_flags |= AVFMTCTX_NOHEADER;
+ rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->rtp_handle, rtsp_st->sdp_payload_type, &rtsp_st->rtp_payload_data);
+
+ if (!rtsp_st->rtp_ctx) {
+ err = AVERROR_NOMEM;
goto fail;
+ } else {
+ if(rtsp_st->dynamic_handler) {
+ rtsp_st->rtp_ctx->dynamic_protocol_context= rtsp_st->dynamic_protocol_context;
+ rtsp_st->rtp_ctx->parse_packet= rtsp_st->dynamic_handler->parse_packet;
+ }
}
}
-
- /* start playing */
- snprintf(cmd, sizeof(cmd),
- "PLAY %s RTSP/1.0\n",
- s->filename);
- rtsp_send_cmd(s, cmd, reply, NULL);
- if (reply->status_code != RTSP_STATUS_OK) {
- err = AVERROR_INVALIDDATA;
- goto fail;
- }
- /* open TCP with bufferized input */
- if (rt->protocol == RTSP_PROTOCOL_RTP_TCP) {
- if (url_fdopen(&rt->rtsp_gb, rt->rtsp_hd) < 0) {
- err = AVERROR_NOMEM;
+ rt->state = RTSP_STATE_IDLE;
+ rt->seek_timestamp = 0; /* default is to start stream at position
+ zero */
+ if (ap->initial_pause) {
+ /* do not start immediately */
+ } else {
+ if (rtsp_read_play(s) < 0) {
+ err = AVERROR_INVALIDDATA;
goto fail;
}
}
-
return 0;
fail:
- for(i=0;i<s->nb_streams;i++) {
- st = s->streams[i];
- rtsp_st = st->priv_data;
- if (rtsp_st) {
- if (rtsp_st->ic)
- av_close_input_file(rtsp_st->ic);
- }
- av_free(rtsp_st);
- }
+ rtsp_close_streams(rt);
av_freep(&content);
url_close(rt->rtsp_hd);
return err;
}
-static int tcp_read_packet(AVFormatContext *s,
- AVPacket *pkt)
+static int tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
+ uint8_t *buf, int buf_size)
{
RTSPState *rt = s->priv_data;
- ByteIOContext *rtsp_gb = &rt->rtsp_gb;
- int c, id, len, i, ret;
- AVStream *st;
+ int id, len, i, ret;
RTSPStream *rtsp_st;
- char buf[RTP_MAX_PACKET_LENGTH];
+#ifdef DEBUG_RTP_TCP
+ printf("tcp_read_packet:\n");
+#endif
redo:
for(;;) {
- c = url_fgetc(rtsp_gb);
- if (c == URL_EOF)
- return AVERROR_IO;
- if (c == '$')
+ ret = url_readbuf(rt->rtsp_hd, buf, 1);
+#ifdef DEBUG_RTP_TCP
+ printf("ret=%d c=%02x [%c]\n", ret, buf[0], buf[0]);
+#endif
+ if (ret != 1)
+ return -1;
+ if (buf[0] == '$')
break;
}
- id = get_byte(rtsp_gb);
- len = get_be16(rtsp_gb);
- if (len > RTP_MAX_PACKET_LENGTH || len < 12)
+ ret = url_readbuf(rt->rtsp_hd, buf, 3);
+ if (ret != 3)
+ return -1;
+ id = buf[0];
+ len = (buf[1] << 8) | buf[2];
+#ifdef DEBUG_RTP_TCP
+ printf("id=%d len=%d\n", id, len);
+#endif
+ if (len > buf_size || len < 12)
goto redo;
/* get the data */
- get_buffer(rtsp_gb, buf, len);
-
+ ret = url_readbuf(rt->rtsp_hd, buf, len);
+ if (ret != len)
+ return -1;
+
/* find the matching stream */
- for(i = 0; i < s->nb_streams; i++) {
- st = s->streams[i];
- rtsp_st = st->priv_data;
- if (i >= rtsp_st->interleaved_min &&
- i <= rtsp_st->interleaved_max)
+ for(i = 0; i < rt->nb_rtsp_streams; i++) {
+ rtsp_st = rt->rtsp_streams[i];
+ if (id >= rtsp_st->interleaved_min &&
+ id <= rtsp_st->interleaved_max)
goto found;
}
goto redo;
found:
- ret = rtp_parse_packet(rtsp_st->ic, pkt, buf, len);
- if (ret < 0)
- goto redo;
- pkt->stream_index = i;
- return ret;
+ *prtsp_st = rtsp_st;
+ return len;
}
-/* NOTE: output one packet at a time. May need to add a small fifo */
-static int udp_read_packet(AVFormatContext *s,
- AVPacket *pkt)
+static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
+ uint8_t *buf, int buf_size)
{
- AVFormatContext *ic;
- AVStream *st;
+ RTSPState *rt = s->priv_data;
RTSPStream *rtsp_st;
fd_set rfds;
int fd1, fd2, fd_max, n, i, ret;
- char buf[RTP_MAX_PACKET_LENGTH];
struct timeval tv;
for(;;) {
- if (rtsp_abort_req)
- return -EIO;
+ if (url_interrupt_cb())
+ return -1;
FD_ZERO(&rfds);
fd_max = -1;
- for(i = 0; i < s->nb_streams; i++) {
- st = s->streams[i];
- rtsp_st = st->priv_data;
- ic = rtsp_st->ic;
+ for(i = 0; i < rt->nb_rtsp_streams; i++) {
+ rtsp_st = rt->rtsp_streams[i];
/* currently, we cannot probe RTCP handle because of blocking restrictions */
- rtp_get_file_handles(url_fileno(&ic->pb), &fd1, &fd2);
+ rtp_get_file_handles(rtsp_st->rtp_handle, &fd1, &fd2);
if (fd1 > fd_max)
fd_max = fd1;
FD_SET(fd1, &rfds);
}
- /* XXX: also add proper API to abort */
tv.tv_sec = 0;
- tv.tv_usec = 500000;
+ tv.tv_usec = 100 * 1000;
n = select(fd_max + 1, &rfds, NULL, NULL, &tv);
if (n > 0) {
- for(i = 0; i < s->nb_streams; i++) {
- st = s->streams[i];
- rtsp_st = st->priv_data;
- ic = rtsp_st->ic;
- rtp_get_file_handles(url_fileno(&ic->pb), &fd1, &fd2);
+ for(i = 0; i < rt->nb_rtsp_streams; i++) {
+ rtsp_st = rt->rtsp_streams[i];
+ rtp_get_file_handles(rtsp_st->rtp_handle, &fd1, &fd2);
if (FD_ISSET(fd1, &rfds)) {
- ret = url_read(url_fileno(&ic->pb), buf, sizeof(buf));
- if (ret >= 0 &&
- rtp_parse_packet(ic, pkt, buf, ret) == 0) {
- pkt->stream_index = i;
+ ret = url_read(rtsp_st->rtp_handle, buf, buf_size);
+ if (ret > 0) {
+ *prtsp_st = rtsp_st;
return ret;
}
}
AVPacket *pkt)
{
RTSPState *rt = s->priv_data;
- int ret;
+ RTSPStream *rtsp_st;
+ int ret, len;
+ uint8_t buf[RTP_MAX_PACKET_LENGTH];
+
+ /* get next frames from the same RTP packet */
+ if (rt->cur_rtp) {
+ ret = rtp_parse_packet(rt->cur_rtp, pkt, NULL, 0);
+ if (ret == 0) {
+ rt->cur_rtp = NULL;
+ return 0;
+ } else if (ret == 1) {
+ return 0;
+ } else {
+ rt->cur_rtp = NULL;
+ }
+ }
+ /* read next RTP packet */
+ redo:
switch(rt->protocol) {
default:
case RTSP_PROTOCOL_RTP_TCP:
- ret = tcp_read_packet(s, pkt);
+ len = tcp_read_packet(s, &rtsp_st, buf, sizeof(buf));
break;
case RTSP_PROTOCOL_RTP_UDP:
- ret = udp_read_packet(s, pkt);
+ case RTSP_PROTOCOL_RTP_UDP_MULTICAST:
+ len = udp_read_packet(s, &rtsp_st, buf, sizeof(buf));
+ if (rtsp_st->rtp_ctx)
+ rtp_check_and_send_back_rr(rtsp_st->rtp_ctx, len);
break;
}
- return ret;
+ if (len < 0)
+ return AVERROR_IO;
+ ret = rtp_parse_packet(rtsp_st->rtp_ctx, pkt, buf, len);
+ if (ret < 0)
+ goto redo;
+ if (ret == 1) {
+ /* more packets may follow, so we save the RTP context */
+ rt->cur_rtp = rtsp_st->rtp_ctx;
+ }
+ return 0;
}
-static int rtsp_read_close(AVFormatContext *s)
+static int rtsp_read_play(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
- AVStream *st;
- RTSPStream *rtsp_st;
RTSPHeader reply1, *reply = &reply1;
- int i;
char cmd[1024];
- /* NOTE: it is valid to flush the buffer here */
- if (rt->protocol == RTSP_PROTOCOL_RTP_TCP) {
- url_fclose(&rt->rtsp_gb);
+ av_log(s, AV_LOG_DEBUG, "hello state=%d\n", rt->state);
+
+ if (rt->state == RTSP_STATE_PAUSED) {
+ snprintf(cmd, sizeof(cmd),
+ "PLAY %s RTSP/1.0\r\n",
+ s->filename);
+ } else {
+ snprintf(cmd, sizeof(cmd),
+ "PLAY %s RTSP/1.0\r\n"
+ "Range: npt=%0.3f-\r\n",
+ s->filename,
+ (double)rt->seek_timestamp / AV_TIME_BASE);
}
+ rtsp_send_cmd(s, cmd, reply, NULL);
+ if (reply->status_code != RTSP_STATUS_OK) {
+ return -1;
+ } else {
+ rt->state = RTSP_STATE_PLAYING;
+ return 0;
+ }
+}
+
+/* pause the stream */
+static int rtsp_read_pause(AVFormatContext *s)
+{
+ RTSPState *rt = s->priv_data;
+ RTSPHeader reply1, *reply = &reply1;
+ char cmd[1024];
+
+ rt = s->priv_data;
+
+ if (rt->state != RTSP_STATE_PLAYING)
+ return 0;
- snprintf(cmd, sizeof(cmd),
- "TEARDOWN %s RTSP/1.0\n",
+ snprintf(cmd, sizeof(cmd),
+ "PAUSE %s RTSP/1.0\r\n",
s->filename);
rtsp_send_cmd(s, cmd, reply, NULL);
+ if (reply->status_code != RTSP_STATUS_OK) {
+ return -1;
+ } else {
+ rt->state = RTSP_STATE_PAUSED;
+ return 0;
+ }
+}
+
+static int rtsp_read_seek(AVFormatContext *s, int stream_index,
+ int64_t timestamp, int flags)
+{
+ RTSPState *rt = s->priv_data;
- if (ff_rtsp_callback) {
- ff_rtsp_callback(RTSP_ACTION_CLIENT_TEARDOWN, rt->session_id,
- NULL, 0, NULL);
+ rt->seek_timestamp = timestamp;
+ switch(rt->state) {
+ default:
+ case RTSP_STATE_IDLE:
+ break;
+ case RTSP_STATE_PLAYING:
+ if (rtsp_read_play(s) != 0)
+ return -1;
+ break;
+ case RTSP_STATE_PAUSED:
+ rt->state = RTSP_STATE_IDLE;
+ break;
}
+ return 0;
+}
- for(i=0;i<s->nb_streams;i++) {
- st = s->streams[i];
- rtsp_st = st->priv_data;
- if (rtsp_st) {
- if (rtsp_st->ic)
- av_close_input_file(rtsp_st->ic);
- }
- av_free(rtsp_st);
+static int rtsp_read_close(AVFormatContext *s)
+{
+ RTSPState *rt = s->priv_data;
+ RTSPHeader reply1, *reply = &reply1;
+ char cmd[1024];
+
+#if 0
+ /* NOTE: it is valid to flush the buffer here */
+ if (rt->protocol == RTSP_PROTOCOL_RTP_TCP) {
+ url_fclose(&rt->rtsp_gb);
}
+#endif
+ snprintf(cmd, sizeof(cmd),
+ "TEARDOWN %s RTSP/1.0\r\n",
+ s->filename);
+ rtsp_send_cmd(s, cmd, reply, NULL);
+
+ rtsp_close_streams(rt);
url_close(rt->rtsp_hd);
return 0;
}
-static AVInputFormat rtsp_demux = {
+#ifdef CONFIG_RTSP_DEMUXER
+AVInputFormat rtsp_demuxer = {
"rtsp",
"RTSP input format",
sizeof(RTSPState),
rtsp_read_header,
rtsp_read_packet,
rtsp_read_close,
+ rtsp_read_seek,
.flags = AVFMT_NOFILE,
+ .read_play = rtsp_read_play,
+ .read_pause = rtsp_read_pause,
};
+#endif
static int sdp_probe(AVProbeData *p1)
{
- const char *p;
+ const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
/* we look for a line beginning "c=IN IP4" */
- p = p1->buf;
- while (*p != '\0') {
- if (strstart(p, "c=IN IP4", NULL))
+ while (p < p_end && *p != '\0') {
+ if (p + sizeof("c=IN IP4") - 1 < p_end && strstart(p, "c=IN IP4", NULL))
return AVPROBE_SCORE_MAX / 2;
- p = strchr(p, '\n');
- if (!p)
+
+ while(p < p_end - 1 && *p != '\n') p++;
+ if (++p >= p_end)
break;
- p++;
if (*p == '\r')
p++;
}
static int sdp_read_header(AVFormatContext *s,
AVFormatParameters *ap)
{
- AVStream *st;
+ RTSPState *rt = s->priv_data;
RTSPStream *rtsp_st;
int size, i, err;
char *content;
char url[1024];
+ AVStream *st;
/* read the whole sdp file */
/* XXX: better loading */
av_free(content);
/* open each RTP stream */
- for(i=0;i<s->nb_streams;i++) {
- st = s->streams[i];
- rtsp_st = st->priv_data;
-
- snprintf(url, sizeof(url), "rtp://%s:%d?multicast=1&ttl=%d",
- inet_ntoa(rtsp_st->sdp_ip),
+ for(i=0;i<rt->nb_rtsp_streams;i++) {
+ rtsp_st = rt->rtsp_streams[i];
+
+ snprintf(url, sizeof(url), "rtp://%s:%d?multicast=1&ttl=%d",
+ inet_ntoa(rtsp_st->sdp_ip),
rtsp_st->sdp_port,
rtsp_st->sdp_ttl);
- if (av_open_input_file(&rtsp_st->ic, url, &rtp_demux, 0, NULL) < 0) {
+ if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
err = AVERROR_INVALIDDATA;
goto fail;
}
+ /* open the RTP context */
+ st = NULL;
+ if (rtsp_st->stream_index >= 0)
+ st = s->streams[rtsp_st->stream_index];
+ if (!st)
+ s->ctx_flags |= AVFMTCTX_NOHEADER;
+ rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->rtp_handle, rtsp_st->sdp_payload_type, &rtsp_st->rtp_payload_data);
+ if (!rtsp_st->rtp_ctx) {
+ err = AVERROR_NOMEM;
+ goto fail;
+ } else {
+ if(rtsp_st->dynamic_handler) {
+ rtsp_st->rtp_ctx->dynamic_protocol_context= rtsp_st->dynamic_protocol_context;
+ rtsp_st->rtp_ctx->parse_packet= rtsp_st->dynamic_handler->parse_packet;
+ }
+ }
}
return 0;
fail:
- for(i=0;i<s->nb_streams;i++) {
- st = s->streams[i];
- rtsp_st = st->priv_data;
- if (rtsp_st) {
- if (rtsp_st->ic)
- av_close_input_file(rtsp_st->ic);
- }
- av_free(rtsp_st);
- }
+ rtsp_close_streams(rt);
return err;
}
static int sdp_read_packet(AVFormatContext *s,
AVPacket *pkt)
{
- return udp_read_packet(s, pkt);
+ return rtsp_read_packet(s, pkt);
}
static int sdp_read_close(AVFormatContext *s)
{
- AVStream *st;
- RTSPStream *rtsp_st;
- int i;
-
- for(i=0;i<s->nb_streams;i++) {
- st = s->streams[i];
- rtsp_st = st->priv_data;
- if (rtsp_st) {
- if (rtsp_st->ic)
- av_close_input_file(rtsp_st->ic);
- }
- av_free(rtsp_st);
- }
+ RTSPState *rt = s->priv_data;
+ rtsp_close_streams(rt);
return 0;
}
-
-static AVInputFormat sdp_demux = {
+#ifdef CONFIG_SDP_DEMUXER
+AVInputFormat sdp_demuxer = {
"sdp",
"SDP",
sizeof(RTSPState),
sdp_read_packet,
sdp_read_close,
};
+#endif
-
+#ifdef CONFIG_REDIR_DEMUXER
/* dummy redirector format (used directly in av_open_input_file now) */
static int redir_probe(AVProbeData *pd)
{
return 0;
}
-AVInputFormat redir_demux = {
+AVInputFormat redir_demuxer = {
"redir",
"Redirector format",
0,
NULL,
NULL,
};
-
-int rtsp_init(void)
-{
- av_register_input_format(&rtsp_demux);
- av_register_input_format(&redir_demux);
- av_register_input_format(&sdp_demux);
- return 0;
-}
+#endif
projects / ffmpeg / blobdiff