* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-/* needed by inet_aton() */
-#define _SVID_SOURCE
-
#include "libavutil/base64.h"
#include "libavutil/avstring.h"
#include "libavutil/intreadwrite.h"
+#include "libavutil/random_seed.h"
#include "avformat.h"
#include <sys/time.h>
#include <sys/select.h>
#endif
#include <strings.h>
+#include "internal.h"
#include "network.h"
+#include "os_support.h"
+#include "http.h"
#include "rtsp.h"
#include "rtpdec.h"
#include "rdt.h"
-#include "rtp_asf.h"
-#include "rtp_vorbis.h"
+#include "rtpdec_asf.h"
//#define DEBUG
//#define DEBUG_RTP_TCP
-#if CONFIG_RTSP_DEMUXER
-static int tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
- uint8_t *buf, int buf_size);
-static int rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
- unsigned char **content_ptr,
- int return_on_interleaved_data);
-#endif
-
#if LIBAVFORMAT_VERSION_INT < (53 << 16)
int rtsp_default_protocols = (1 << RTSP_LOWER_TRANSPORT_UDP);
#endif
+/* Timeout values for socket select, in ms,
+ * and read_packet(), in seconds */
+#define SELECT_TIMEOUT_MS 100
+#define READ_PACKET_TIMEOUT_S 10
+#define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / SELECT_TIMEOUT_MS
+
#define SPACE_CHARS " \t\r\n"
/* we use memchr() instead of strchr() here because strchr() will return
* the terminating '\0' of SPACE_CHARS instead of NULL if c is '\0'. */
get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
}
-/* parse the rtpmap description: <codec_name>/<clock_rate>[/<other
- params>] */
-static int sdp_parse_rtpmap(AVCodecContext *codec, RTSPStream *rtsp_st, int payload_type, const char *p)
+/* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
+static int sdp_parse_rtpmap(AVFormatContext *s,
+ AVCodecContext *codec, RTSPStream *rtsp_st,
+ int payload_type, const char *p)
{
char buf[256];
int i;
const char *c_name;
/* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
- see if we can handle this kind of payload */
- get_word_sep(buf, sizeof(buf), "/", &p);
+ * see if we can handle this kind of payload.
+ * The space should normally not be there but some Real streams or
+ * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
+ * have a trailing space. */
+ get_word_sep(buf, sizeof(buf), "/ ", &p);
if (payload_type >= RTP_PT_PRIVATE) {
- RTPDynamicProtocolHandler *handler= RTPFirstDynamicPayloadHandler;
- while(handler) {
- if (!strcasecmp(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();
- }
+ RTPDynamicProtocolHandler *handler;
+ for (handler = RTPFirstDynamicPayloadHandler;
+ handler; handler = handler->next) {
+ if (!strcasecmp(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 {
- /* We are in a standard case ( from http://www.iana.org/assignments/rtp-parameters) */
+ /* We are in a standard case
+ * (from http://www.iana.org/assignments/rtp-parameters). */
/* search into AVRtpPayloadTypes[] */
codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
}
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;
+ c_name = "(null)";
+
+ get_word_sep(buf, sizeof(buf), "/", &p);
+ i = atoi(buf);
+ switch (codec->codec_type) {
+ case AVMEDIA_TYPE_AUDIO:
+ av_log(s, 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. Patch on its way. (rdm)
}
- return 0;
+ av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
+ codec->sample_rate);
+ av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
+ codec->channels);
+ break;
+ case AVMEDIA_TYPE_VIDEO:
+ av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
+ break;
+ default:
+ break;
}
-
- return -1;
+ return 0;
}
-/* return the length and optionnaly the data */
+/* return the length and optionally the data */
static int hex_to_data(uint8_t *data, const char *p)
{
int c, len, v;
len = 0;
v = 1;
- for(;;) {
+ for (;;) {
skip_spaces(&p);
if (*p == '\0')
break;
- c = toupper((unsigned char)*p++);
+ c = toupper((unsigned char) *p++);
if (c >= '0' && c <= '9')
c = c - '0';
else if (c >= 'A' && c <= 'F')
char *attr, char *value)
{
switch (codec->codec_id) {
- case CODEC_ID_MPEG4:
- case CODEC_ID_AAC:
- if (!strcmp(attr, "config")) {
- /* decode the hexa encoded parameter */
- int len = hex_to_data(NULL, value);
- if (codec->extradata)
- av_free(codec->extradata);
- codec->extradata = av_mallocz(len + FF_INPUT_BUFFER_PADDING_SIZE);
- if (!codec->extradata)
- return;
- codec->extradata_size = len;
- hex_to_data(codec->extradata, value);
- }
- break;
- case CODEC_ID_VORBIS:
- ff_vorbis_parse_fmtp_config(codec, ctx, attr, value);
- break;
- default:
- break;
+ case CODEC_ID_MPEG4:
+ case CODEC_ID_AAC:
+ if (!strcmp(attr, "config")) {
+ /* decode the hexa encoded parameter */
+ int len = hex_to_data(NULL, value);
+ if (codec->extradata)
+ av_free(codec->extradata);
+ codec->extradata = av_mallocz(len + FF_INPUT_BUFFER_PADDING_SIZE);
+ if (!codec->extradata)
+ return;
+ codec->extradata_size = len;
+ hex_to_data(codec->extradata, value);
+ }
+ break;
+ default:
+ break;
}
return;
}
typedef struct {
const char *str;
- uint16_t type;
- uint32_t offset;
+ uint16_t type;
+ uint32_t offset;
} AttrNameMap;
-/* All known fmtp parmeters and the corresping RTPAttrTypeEnum */
+/* All known fmtp parameters and the corresponding RTPAttrTypeEnum */
#define ATTR_NAME_TYPE_INT 0
#define ATTR_NAME_TYPE_STR 1
static const AttrNameMap attr_names[]=
{
- {"SizeLength", ATTR_NAME_TYPE_INT, offsetof(RTPPayloadData, sizelength)},
- {"IndexLength", ATTR_NAME_TYPE_INT, offsetof(RTPPayloadData, indexlength)},
- {"IndexDeltaLength", ATTR_NAME_TYPE_INT, offsetof(RTPPayloadData, indexdeltalength)},
- {"profile-level-id", ATTR_NAME_TYPE_INT, offsetof(RTPPayloadData, profile_level_id)},
- {"StreamType", ATTR_NAME_TYPE_INT, offsetof(RTPPayloadData, streamtype)},
- {"mode", ATTR_NAME_TYPE_STR, offsetof(RTPPayloadData, mode)},
- {NULL, -1, -1},
+ { "SizeLength", ATTR_NAME_TYPE_INT,
+ offsetof(RTPPayloadData, sizelength) },
+ { "IndexLength", ATTR_NAME_TYPE_INT,
+ offsetof(RTPPayloadData, indexlength) },
+ { "IndexDeltaLength", ATTR_NAME_TYPE_INT,
+ offsetof(RTPPayloadData, indexdeltalength) },
+ { "profile-level-id", ATTR_NAME_TYPE_INT,
+ offsetof(RTPPayloadData, profile_level_id) },
+ { "StreamType", ATTR_NAME_TYPE_INT,
+ offsetof(RTPPayloadData, streamtype) },
+ { "mode", ATTR_NAME_TYPE_STR,
+ offsetof(RTPPayloadData, 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)
+/* parse the attribute line from the fmtp a line of an sdp response. This
+ * is broken out as a function because it is used in rtp_h264.c, which is
+ * forthcoming. */
+int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
+ char *value, int value_size)
{
skip_spaces(p);
- if(**p) {
+ if (**p) {
get_word_sep(attr, attr_size, "=", p);
if (**p == '=')
(*p)++;
static void sdp_parse_fmtp(AVStream *st, const char *p)
{
char attr[256];
- /* Vorbis setup headers can be up to 12KB and are sent base64
- * encoded, giving a 12KB * (4/3) = 16KB FMTP line. */
- char value[16384];
+ char value[4096];
int i;
-
RTSPStream *rtsp_st = st->priv_data;
AVCodecContext *codec = st->codec;
RTPPayloadData *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 */
+ while (ff_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, rtsp_st->dynamic_protocol_context,
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);
+ 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);
}
}
}
typedef struct SDPParseState {
/* SDP only */
struct in_addr default_ip;
- int default_ttl;
- int skip_media; ///< set if an unknown m= line occurs
+ int default_ttl;
+ int skip_media; ///< set if an unknown m= line occurs
} SDPParseState;
static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
RTSPState *rt = s->priv_data;
char buf1[64], st_type[64];
const char *p;
- enum CodecType codec_type;
+ enum AVMediaType codec_type;
int payload_type, i;
AVStream *st;
RTSPStream *rtsp_st;
p = buf;
if (s1->skip_media && letter != 'm')
return;
- switch(letter) {
+ switch (letter) {
case 'c':
get_word(buf1, sizeof(buf1), &p);
if (strcmp(buf1, "IN") != 0)
if (strcmp(buf1, "IP4") != 0)
return;
get_word_sep(buf1, sizeof(buf1), "/", &p);
- if (inet_aton(buf1, &sdp_ip) == 0)
+ if (ff_inet_aton(buf1, &sdp_ip) == 0)
return;
ttl = 16;
if (*p == '/') {
}
break;
case 's':
- av_metadata_set(&s->metadata, "title", p);
+ av_metadata_set2(&s->metadata, "title", p, 0);
break;
case 'i':
if (s->nb_streams == 0) {
- av_metadata_set(&s->metadata, "comment", p);
+ av_metadata_set2(&s->metadata, "comment", p, 0);
break;
}
break;
s1->skip_media = 0;
get_word(st_type, sizeof(st_type), &p);
if (!strcmp(st_type, "audio")) {
- codec_type = CODEC_TYPE_AUDIO;
+ codec_type = AVMEDIA_TYPE_AUDIO;
} else if (!strcmp(st_type, "video")) {
- codec_type = CODEC_TYPE_VIDEO;
+ codec_type = AVMEDIA_TYPE_VIDEO;
} else if (!strcmp(st_type, "application")) {
- codec_type = CODEC_TYPE_DATA;
+ codec_type = AVMEDIA_TYPE_DATA;
} else {
s1->skip_media = 1;
return;
}
}
/* put a default control url */
- av_strlcpy(rtsp_st->control_url, s->filename, sizeof(rtsp_st->control_url));
+ av_strlcpy(rtsp_st->control_url, rt->control_uri,
+ sizeof(rtsp_st->control_url));
break;
case 'a':
- if (av_strstart(p, "control:", &p) && s->nb_streams > 0) {
+ if (av_strstart(p, "control:", &p)) {
+ if (s->nb_streams == 0) {
+ if (!strncmp(p, "rtsp://", 7))
+ av_strlcpy(rt->control_uri, p,
+ sizeof(rt->control_uri));
+ } else {
char proto[32];
/* 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, 0, NULL, NULL, 0, p);
+ ff_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
+ NULL, NULL, 0, p);
if (proto[0] == '\0') {
/* relative control URL */
- av_strlcat(rtsp_st->control_url, "/", sizeof(rtsp_st->control_url));
- av_strlcat(rtsp_st->control_url, p, sizeof(rtsp_st->control_url));
- } else {
- av_strlcpy(rtsp_st->control_url, p, sizeof(rtsp_st->control_url));
+ if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
+ av_strlcat(rtsp_st->control_url, "/",
+ sizeof(rtsp_st->control_url));
+ av_strlcat(rtsp_st->control_url, p,
+ sizeof(rtsp_st->control_url));
+ } else
+ av_strlcpy(rtsp_st->control_url, p,
+ sizeof(rtsp_st->control_url));
}
} else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
/* NOTE: rtpmap is only supported AFTER the 'm=' tag */
payload_type = atoi(buf1);
st = s->streams[s->nb_streams - 1];
rtsp_st = st->priv_data;
- sdp_parse_rtpmap(st->codec, rtsp_st, payload_type, p);
+ sdp_parse_rtpmap(s, st->codec, rtsp_st, payload_type, p);
} else if (av_strstart(p, "fmtp:", &p)) {
/* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
get_word(buf1, sizeof(buf1), &p);
payload_type = atoi(buf1);
- for(i = 0; i < s->nb_streams;i++) {
- st = s->streams[i];
+ 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) {
- if(!rtsp_st->dynamic_handler->parse_sdp_a_line(s, i, rtsp_st->dynamic_protocol_context, buf)) {
- sdp_parse_fmtp(st, p);
- }
- } else {
+ if (!(rtsp_st->dynamic_handler &&
+ rtsp_st->dynamic_handler->parse_sdp_a_line &&
+ rtsp_st->dynamic_handler->parse_sdp_a_line(s,
+ i, rtsp_st->dynamic_protocol_context, buf)))
sdp_parse_fmtp(st, p);
- }
}
}
- } else if(av_strstart(p, "framesize:", &p)) {
+ } else if (av_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];
+ 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(s, i, rtsp_st->dynamic_protocol_context, buf);
- }
- }
+ if (rtsp_st->sdp_payload_type == payload_type &&
+ rtsp_st->dynamic_handler &&
+ rtsp_st->dynamic_handler->parse_sdp_a_line)
+ rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
+ rtsp_st->dynamic_protocol_context, buf);
}
- } else if(av_strstart(p, "range:", &p)) {
+ } else if (av_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)
+ s->start_time = start;
+ /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
+ s->duration = (end == AV_NOPTS_VALUE) ?
+ AV_NOPTS_VALUE : end - start;
} else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
if (atoi(p) == 1)
rt->transport = RTSP_TRANSPORT_RDT;
rtsp_st = s->streams[s->nb_streams - 1]->priv_data;
if (rtsp_st->dynamic_handler &&
rtsp_st->dynamic_handler->parse_sdp_a_line)
- rtsp_st->dynamic_handler->parse_sdp_a_line(s, s->nb_streams - 1,
+ rtsp_st->dynamic_handler->parse_sdp_a_line(s,
+ s->nb_streams - 1,
rtsp_st->dynamic_protocol_context, buf);
}
}
* "rulebooks" describing their properties. Therefore, the SDP line
* buffer is large.
*
- * The Vorbis FMTP line can be up to 16KB - see sdp_parse_fmtp. */
+ * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
+ * in rtpdec_xiph.c. */
char buf[16384], *q;
SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
memset(s1, 0, sizeof(SDPParseState));
p = content;
- for(;;) {
+ for (;;) {
skip_spaces(&p);
letter = *p;
if (letter == '\0')
return 0;
}
-static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
- uint8_t *buf, int buf_size)
-{
- RTSPState *rt = s->priv_data;
- RTSPStream *rtsp_st;
- fd_set rfds;
- int fd, fd_max, n, i, ret, tcp_fd;
- struct timeval tv;
-
- for(;;) {
- if (url_interrupt_cb())
- return AVERROR(EINTR);
- FD_ZERO(&rfds);
- if (rt->rtsp_hd) {
- tcp_fd = fd_max = url_get_file_handle(rt->rtsp_hd);
- FD_SET(tcp_fd, &rfds);
- } else {
- fd_max = 0;
- tcp_fd = -1;
- }
- for(i = 0; i < rt->nb_rtsp_streams; i++) {
- rtsp_st = rt->rtsp_streams[i];
- if (rtsp_st->rtp_handle) {
- /* currently, we cannot probe RTCP handle because of
- * blocking restrictions */
- fd = url_get_file_handle(rtsp_st->rtp_handle);
- if (fd > fd_max)
- fd_max = fd;
- FD_SET(fd, &rfds);
- }
- }
- tv.tv_sec = 0;
- tv.tv_usec = 100 * 1000;
- n = select(fd_max + 1, &rfds, NULL, NULL, &tv);
- if (n > 0) {
- for(i = 0; i < rt->nb_rtsp_streams; i++) {
- rtsp_st = rt->rtsp_streams[i];
- if (rtsp_st->rtp_handle) {
- fd = url_get_file_handle(rtsp_st->rtp_handle);
- if (FD_ISSET(fd, &rfds)) {
- ret = url_read(rtsp_st->rtp_handle, buf, buf_size);
- if (ret > 0) {
- *prtsp_st = rtsp_st;
- return ret;
- }
- }
- }
- }
-#if CONFIG_RTSP_DEMUXER
- if (FD_ISSET(tcp_fd, &rfds)) {
- RTSPMessageHeader reply;
-
- rtsp_read_reply(s, &reply, NULL, 0);
- /* XXX: parse message */
- if (rt->state != RTSP_STATE_PLAYING)
- return 0;
- }
-#endif
- }
- }
-}
-
-static int rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
-{
- RTSPState *rt = s->priv_data;
- int ret, len;
- uint8_t buf[10 * RTP_MAX_PACKET_LENGTH];
- RTSPStream *rtsp_st;
-
- /* get next frames from the same RTP packet */
- if (rt->cur_transport_priv) {
- if (rt->transport == RTSP_TRANSPORT_RDT)
- ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
- else
- ret = rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
- if (ret == 0) {
- rt->cur_transport_priv = NULL;
- return 0;
- } else if (ret == 1) {
- return 0;
- } else {
- rt->cur_transport_priv = NULL;
- }
- }
-
- /* read next RTP packet */
- redo:
- switch(rt->lower_transport) {
- default:
-#if CONFIG_RTSP_DEMUXER
- case RTSP_LOWER_TRANSPORT_TCP:
- len = tcp_read_packet(s, &rtsp_st, buf, sizeof(buf));
- break;
-#endif
- case RTSP_LOWER_TRANSPORT_UDP:
- case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
- len = udp_read_packet(s, &rtsp_st, buf, sizeof(buf));
- if (len >=0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
- rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
- break;
- }
- if (len < 0)
- return len;
- if (len == 0)
- return AVERROR_EOF;
- if (rt->transport == RTSP_TRANSPORT_RDT)
- ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, buf, len);
- else
- ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, buf, len);
- if (ret < 0)
- goto redo;
- if (ret == 1) {
- /* more packets may follow, so we save the RTP context */
- rt->cur_transport_priv = rtsp_st->transport_priv;
- }
-
- return ret;
-}
-
/* close and free RTSP streams */
-static void rtsp_close_streams(RTSPState *rt)
+void ff_rtsp_close_streams(AVFormatContext *s)
{
+ RTSPState *rt = s->priv_data;
int i;
RTSPStream *rtsp_st;
- for(i=0;i<rt->nb_rtsp_streams;i++) {
+ for (i = 0; i < rt->nb_rtsp_streams; i++) {
rtsp_st = rt->rtsp_streams[i];
if (rtsp_st) {
if (rtsp_st->transport_priv) {
- if (rt->transport == RTSP_TRANSPORT_RDT)
+ if (s->oformat) {
+ AVFormatContext *rtpctx = rtsp_st->transport_priv;
+ av_write_trailer(rtpctx);
+ if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
+ uint8_t *ptr;
+ url_close_dyn_buf(rtpctx->pb, &ptr);
+ av_free(ptr);
+ } else {
+ url_fclose(rtpctx->pb);
+ }
+ av_metadata_free(&rtpctx->streams[0]->metadata);
+ av_metadata_free(&rtpctx->metadata);
+ av_free(rtpctx->streams[0]);
+ av_free(rtpctx);
+ } else if (rt->transport == RTSP_TRANSPORT_RDT)
ff_rdt_parse_close(rtsp_st->transport_priv);
else
rtp_parse_close(rtsp_st->transport_priv);
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);
+ rtsp_st->dynamic_handler->close(
+ rtsp_st->dynamic_protocol_context);
}
}
av_free(rt->rtsp_streams);
av_close_input_stream (rt->asf_ctx);
rt->asf_ctx = NULL;
}
- av_freep(&rt->auth_b64);
}
-static int
-rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
+static void *rtsp_rtp_mux_open(AVFormatContext *s, AVStream *st,
+ URLContext *handle)
+{
+ RTSPState *rt = s->priv_data;
+ AVFormatContext *rtpctx;
+ int ret;
+ AVOutputFormat *rtp_format = av_guess_format("rtp", NULL, NULL);
+
+ if (!rtp_format)
+ return NULL;
+
+ /* Allocate an AVFormatContext for each output stream */
+ rtpctx = avformat_alloc_context();
+ if (!rtpctx)
+ return NULL;
+
+ rtpctx->oformat = rtp_format;
+ if (!av_new_stream(rtpctx, 0)) {
+ av_free(rtpctx);
+ return NULL;
+ }
+ /* Copy the max delay setting; the rtp muxer reads this. */
+ rtpctx->max_delay = s->max_delay;
+ /* Copy other stream parameters. */
+ rtpctx->streams[0]->sample_aspect_ratio = st->sample_aspect_ratio;
+
+ /* Set the synchronized start time. */
+ rtpctx->start_time_realtime = rt->start_time;
+
+ /* Remove the local codec, link to the original codec
+ * context instead, to give the rtp muxer access to
+ * codec parameters. */
+ av_free(rtpctx->streams[0]->codec);
+ rtpctx->streams[0]->codec = st->codec;
+
+ if (handle) {
+ url_fdopen(&rtpctx->pb, handle);
+ } else
+ url_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
+ ret = av_write_header(rtpctx);
+
+ if (ret) {
+ if (handle) {
+ url_fclose(rtpctx->pb);
+ } else {
+ uint8_t *ptr;
+ url_close_dyn_buf(rtpctx->pb, &ptr);
+ av_free(ptr);
+ }
+ av_free(rtpctx->streams[0]);
+ av_free(rtpctx);
+ return NULL;
+ }
+
+ /* Copy the RTP AVStream timebase back to the original AVStream */
+ st->time_base = rtpctx->streams[0]->time_base;
+ return rtpctx;
+}
+
+static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
{
RTSPState *rt = s->priv_data;
AVStream *st = NULL;
if (!st)
s->ctx_flags |= AVFMTCTX_NOHEADER;
- if (rt->transport == RTSP_TRANSPORT_RDT)
+ if (s->oformat) {
+ rtsp_st->transport_priv = rtsp_rtp_mux_open(s, st, rtsp_st->rtp_handle);
+ /* Ownership of rtp_handle is passed to the rtp mux context */
+ rtsp_st->rtp_handle = NULL;
+ } else if (rt->transport == RTSP_TRANSPORT_RDT)
rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
rtsp_st->dynamic_protocol_context,
rtsp_st->dynamic_handler);
if (!rtsp_st->transport_priv) {
return AVERROR(ENOMEM);
} else if (rt->transport != RTSP_TRANSPORT_RDT) {
- if(rtsp_st->dynamic_handler) {
+ if (rtsp_st->dynamic_handler) {
rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
rtsp_st->dynamic_protocol_context,
rtsp_st->dynamic_handler);
return 0;
}
-#if CONFIG_RTSP_DEMUXER
+#if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
static int rtsp_probe(AVProbeData *p)
{
if (av_strstart(p->filename, "rtsp:", NULL))
reply->nb_transports = 0;
- for(;;) {
+ for (;;) {
skip_spaces(&p);
if (*p == '\0')
break;
if (*p == '=') {
p++;
get_word_sep(buf, sizeof(buf), ";,", &p);
- if (inet_aton(buf, &ipaddr))
+ if (ff_inet_aton(buf, &ipaddr))
th->destination = ntohl(ipaddr.s_addr);
}
}
}
}
-void rtsp_parse_line(RTSPMessageHeader *reply, const char *buf)
+void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
+ HTTPAuthState *auth_state)
{
const char *p;
} else if (av_stristart(p, "Location:", &p)) {
skip_spaces(&p);
av_strlcpy(reply->location, p , sizeof(reply->location));
+ } else if (av_stristart(p, "WWW-Authenticate:", &p) && auth_state) {
+ skip_spaces(&p);
+ ff_http_auth_handle_header(auth_state, "WWW-Authenticate", p);
+ } else if (av_stristart(p, "Authentication-Info:", &p) && auth_state) {
+ skip_spaces(&p);
+ ff_http_auth_handle_header(auth_state, "Authentication-Info", p);
}
}
/* skip a RTP/TCP interleaved packet */
-static void rtsp_skip_packet(AVFormatContext *s)
+void ff_rtsp_skip_packet(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
int ret, len, len1;
}
}
-/**
- * Read a RTSP message from the server, or prepare to read data
- * packets if we're reading data interleaved over the TCP/RTSP
- * connection as well.
- *
- * @param s RTSP demuxer context
- * @param reply pointer where the RTSP message header will be stored
- * @param content_ptr pointer where the RTSP message body, if any, will
- * be stored (length is in reply)
- * @param return_on_interleaved_data whether the function may return if we
- * encounter a data marker ('$'), which precedes data
- * packets over interleaved TCP/RTSP connections. If this
- * is set, this function will return 1 after encountering
- * a '$'. If it is not set, the function will skip any
- * data packets (if they are encountered), until a reply
- * has been fully parsed. If no more data is available
- * without parsing a reply, it will return an error.
- *
- * @returns 1 if a data packets is ready to be received, -1 on error,
- * and 0 on success.
- */
-static int
-rtsp_read_reply (AVFormatContext *s, RTSPMessageHeader *reply,
- unsigned char **content_ptr, int return_on_interleaved_data)
+int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
+ unsigned char **content_ptr,
+ int return_on_interleaved_data)
{
RTSPState *rt = s->priv_data;
char buf[4096], buf1[1024], *q;
/* parse reply (XXX: use buffers) */
rt->last_reply[0] = '\0';
- for(;;) {
+ for (;;) {
q = buf;
- for(;;) {
+ for (;;) {
ret = url_read_complete(rt->rtsp_hd, &ch, 1);
#ifdef DEBUG_RTP_TCP
dprintf(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
if (return_on_interleaved_data) {
return 1;
} else
- rtsp_skip_packet(s);
+ ff_rtsp_skip_packet(s);
} else if (ch != '\r') {
if ((q - buf) < sizeof(buf) - 1)
*q++ = ch;
get_word(buf1, sizeof(buf1), &p);
reply->status_code = atoi(buf1);
} else {
- rtsp_parse_line(reply, p);
+ ff_rtsp_parse_line(reply, p, &rt->auth_state);
av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
}
else
av_free(content);
+ if (rt->seq != reply->seq) {
+ av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
+ rt->seq, reply->seq);
+ }
+
/* EOS */
if (reply->notice == 2101 /* End-of-Stream Reached */ ||
reply->notice == 2104 /* Start-of-Stream Reached */ ||
- reply->notice == 2306 /* Continuous Feed Terminated */)
+ reply->notice == 2306 /* Continuous Feed Terminated */) {
rt->state = RTSP_STATE_IDLE;
- else if (reply->notice >= 4400 && reply->notice < 5500)
+ } else if (reply->notice >= 4400 && reply->notice < 5500) {
return AVERROR(EIO); /* data or server error */
- else if (reply->notice == 2401 /* Ticket Expired */ ||
+ } else if (reply->notice == 2401 /* Ticket Expired */ ||
(reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
return AVERROR(EPERM);
return 0;
}
-static void rtsp_send_cmd_async (AVFormatContext *s,
- const char *cmd, RTSPMessageHeader *reply,
- unsigned char **content_ptr)
+int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
+ const char *method, const char *url,
+ const char *headers,
+ const unsigned char *send_content,
+ int send_content_length)
{
RTSPState *rt = s->priv_data;
- char buf[4096], buf1[1024];
+ char buf[4096], *out_buf;
+ char base64buf[AV_BASE64_SIZE(sizeof(buf))];
+ /* Add in RTSP headers */
+ out_buf = buf;
rt->seq++;
- av_strlcpy(buf, cmd, sizeof(buf));
- snprintf(buf1, sizeof(buf1), "CSeq: %d\r\n", rt->seq);
- av_strlcat(buf, buf1, sizeof(buf));
- if (rt->session_id[0] != '\0' && !strstr(cmd, "\nIf-Match:")) {
- snprintf(buf1, sizeof(buf1), "Session: %s\r\n", rt->session_id);
- av_strlcat(buf, buf1, sizeof(buf));
+ snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
+ if (headers)
+ av_strlcat(buf, headers, sizeof(buf));
+ av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
+ if (rt->session_id[0] != '\0' && (!headers ||
+ !strstr(headers, "\nIf-Match:"))) {
+ av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
+ }
+ if (rt->auth[0]) {
+ char *str = ff_http_auth_create_response(&rt->auth_state,
+ rt->auth, url, method);
+ if (str)
+ av_strlcat(buf, str, sizeof(buf));
+ av_free(str);
}
- if (rt->auth_b64)
- av_strlcatf(buf, sizeof(buf),
- "Authorization: Basic %s\r\n",
- rt->auth_b64);
+ if (send_content_length > 0 && send_content)
+ av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
av_strlcat(buf, "\r\n", sizeof(buf));
+ /* base64 encode rtsp if tunneling */
+ if (rt->control_transport == RTSP_MODE_TUNNEL) {
+ av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
+ out_buf = base64buf;
+ }
+
dprintf(s, "Sending:\n%s--\n", buf);
- url_write(rt->rtsp_hd, buf, strlen(buf));
+ url_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
+ if (send_content_length > 0 && send_content) {
+ if (rt->control_transport == RTSP_MODE_TUNNEL) {
+ av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
+ "with content data not supported\n");
+ return AVERROR_PATCHWELCOME;
+ }
+ url_write(rt->rtsp_hd_out, send_content, send_content_length);
+ }
rt->last_cmd_time = av_gettime();
+
+ return 0;
+}
+
+int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
+ const char *url, const char *headers)
+{
+ return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
}
-static void rtsp_send_cmd (AVFormatContext *s,
- const char *cmd, RTSPMessageHeader *reply,
- unsigned char **content_ptr)
+int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
+ const char *headers, RTSPMessageHeader *reply,
+ unsigned char **content_ptr)
{
- rtsp_send_cmd_async(s, cmd, reply, content_ptr);
+ return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
+ content_ptr, NULL, 0);
+}
+
+int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
+ const char *method, const char *url,
+ const char *header,
+ RTSPMessageHeader *reply,
+ unsigned char **content_ptr,
+ const unsigned char *send_content,
+ int send_content_length)
+{
+ RTSPState *rt = s->priv_data;
+ HTTPAuthType cur_auth_type;
+ int ret;
+
+retry:
+ cur_auth_type = rt->auth_state.auth_type;
+ if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
+ send_content,
+ send_content_length)))
+ return ret;
+
+ if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0) ) < 0)
+ return ret;
- rtsp_read_reply(s, reply, content_ptr, 0);
+ if (reply->status_code == 401 && cur_auth_type == HTTP_AUTH_NONE &&
+ rt->auth_state.auth_type != HTTP_AUTH_NONE)
+ goto retry;
+
+ return 0;
}
/**
- * @returns 0 on success, <0 on error, 1 if protocol is unavailable.
+ * @return 0 on success, <0 on error, 1 if protocol is unavailable.
*/
-static int
-make_setup_request (AVFormatContext *s, const char *host, int port,
- int lower_transport, const char *real_challenge)
+static int make_setup_request(AVFormatContext *s, const char *host, int port,
+ int lower_transport, const char *real_challenge)
{
RTSPState *rt = s->priv_data;
int rtx, j, i, err, interleave = 0;
/* for each stream, make the setup request */
/* XXX: we assume the same server is used for the control of each
- RTSP stream */
+ * RTSP stream */
- for(j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
+ for (j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
char transport[2048];
/**
for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
int len = strlen(rt->rtsp_streams[rtx]->control_url);
if (len >= 4 &&
- !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4, "/rtx"))
+ !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
+ "/rtx"))
break;
}
if (rtx == rt->nb_rtsp_streams)
/* first try in specified port range */
if (RTSP_RTP_PORT_MIN != 0) {
- while(j <= RTSP_RTP_PORT_MAX) {
- snprintf(buf, sizeof(buf), "rtp://%s?localport=%d", host, j);
- j += 2; /* we will use two port by rtp stream (rtp and rtcp) */
- if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0) {
+ while (j <= RTSP_RTP_PORT_MAX) {
+ ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
+ "?localport=%d", j);
+ /* we will use two ports per rtp stream (rtp and rtcp) */
+ j += 2;
+ if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0)
goto rtp_opened;
- }
}
}
-/* then try on any port
-** if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
-** err = AVERROR_INVALIDDATA;
-** goto fail;
-** }
-*/
+#if 0
+ /* then try on any port */
+ if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
+ err = AVERROR_INVALIDDATA;
+ goto fail;
+ }
+#endif
rtp_opened:
port = rtp_get_local_port(rtsp_st->rtp_handle);
* UDP. When trying to set it up for TCP streams, the server
* will return an error. Therefore, we skip those streams. */
if (rt->server_type == RTSP_SERVER_WMS &&
- s->streams[rtsp_st->stream_index]->codec->codec_type == CODEC_TYPE_DATA)
+ s->streams[rtsp_st->stream_index]->codec->codec_type ==
+ AVMEDIA_TYPE_DATA)
continue;
snprintf(transport, sizeof(transport) - 1,
"%s/TCP;", trans_pref);
snprintf(transport, sizeof(transport) - 1,
"%s/UDP;multicast", trans_pref);
}
- if (rt->server_type == RTSP_SERVER_REAL ||
- rt->server_type == RTSP_SERVER_WMS)
+ if (s->oformat) {
+ av_strlcat(transport, ";mode=receive", sizeof(transport));
+ } else if (rt->server_type == RTSP_SERVER_REAL ||
+ rt->server_type == RTSP_SERVER_WMS)
av_strlcat(transport, ";mode=play", sizeof(transport));
snprintf(cmd, sizeof(cmd),
- "SETUP %s RTSP/1.0\r\n"
"Transport: %s\r\n",
- rtsp_st->control_url, transport);
+ transport);
if (i == 0 && rt->server_type == RTSP_SERVER_REAL) {
char real_res[41], real_csum[9];
ff_rdt_calc_response_and_checksum(real_res, real_csum,
"RealChallenge2: %s, sd=%s\r\n",
rt->session_id, real_res, real_csum);
}
- rtsp_send_cmd(s, cmd, reply, NULL);
+ ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
err = 1;
goto fail;
rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
break;
- case RTSP_LOWER_TRANSPORT_UDP:
- {
- char url[1024];
-
- /* XXX: also use address if specified */
- snprintf(url, sizeof(url), "rtp://%s:%d",
- host, reply->transports[0].server_port_min);
- if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
- rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
- err = AVERROR_INVALIDDATA;
- goto fail;
- }
+ case RTSP_LOWER_TRANSPORT_UDP: {
+ char url[1024];
+
+ /* XXX: also use address if specified */
+ ff_url_join(url, sizeof(url), "rtp", NULL, host,
+ reply->transports[0].server_port_min, NULL);
+ if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
+ rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
+ err = AVERROR_INVALIDDATA;
+ goto fail;
}
+ /* Try to initialize the connection state in a
+ * potential NAT router by sending dummy packets.
+ * RTP/RTCP dummy packets are used for RDT, too.
+ */
+ if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat)
+ rtp_send_punch_packets(rtsp_st->rtp_handle);
break;
- case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
- {
- char url[1024];
- struct in_addr in;
- int port, ttl;
-
- if (reply->transports[0].destination) {
- in.s_addr = htonl(reply->transports[0].destination);
- port = reply->transports[0].port_min;
- ttl = reply->transports[0].ttl;
- } else {
- in = rtsp_st->sdp_ip;
- port = rtsp_st->sdp_port;
- ttl = rtsp_st->sdp_ttl;
- }
- snprintf(url, sizeof(url), "rtp://%s:%d?ttl=%d",
- inet_ntoa(in), port, ttl);
- if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
- err = AVERROR_INVALIDDATA;
- goto fail;
- }
+ }
+ case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
+ char url[1024];
+ struct in_addr in;
+ int port, ttl;
+
+ if (reply->transports[0].destination) {
+ in.s_addr = htonl(reply->transports[0].destination);
+ port = reply->transports[0].port_min;
+ ttl = reply->transports[0].ttl;
+ } else {
+ in = rtsp_st->sdp_ip;
+ port = rtsp_st->sdp_port;
+ ttl = rtsp_st->sdp_ttl;
+ }
+ ff_url_join(url, sizeof(url), "rtp", NULL, inet_ntoa(in),
+ port, "?ttl=%d", ttl);
+ if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
+ err = AVERROR_INVALIDDATA;
+ goto fail;
}
break;
}
+ }
if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
goto fail;
return 0;
fail:
- for (i=0; i<rt->nb_rtsp_streams; i++) {
+ for (i = 0; i < rt->nb_rtsp_streams; i++) {
if (rt->rtsp_streams[i]->rtp_handle) {
url_close(rt->rtsp_streams[i]->rtp_handle);
rt->rtsp_streams[i]->rtp_handle = NULL;
{
RTSPState *rt = s->priv_data;
RTSPMessageHeader reply1, *reply = &reply1;
+ int i;
char cmd[1024];
av_log(s, AV_LOG_DEBUG, "hello state=%d\n", rt->state);
if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) {
if (rt->state == RTSP_STATE_PAUSED) {
- snprintf(cmd, sizeof(cmd),
- "PLAY %s RTSP/1.0\r\n",
- s->filename);
+ cmd[0] = 0;
} 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);
+ ff_rtsp_send_cmd(s, "PLAY", rt->control_uri, cmd, reply, NULL);
if (reply->status_code != RTSP_STATUS_OK) {
return -1;
}
+ if (reply->range_start != AV_NOPTS_VALUE &&
+ rt->transport == RTSP_TRANSPORT_RTP) {
+ for (i = 0; i < rt->nb_rtsp_streams; i++) {
+ RTSPStream *rtsp_st = rt->rtsp_streams[i];
+ RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
+ AVStream *st = NULL;
+ if (!rtpctx)
+ continue;
+ if (rtsp_st->stream_index >= 0)
+ st = s->streams[rtsp_st->stream_index];
+ rtpctx->last_rtcp_ntp_time = AV_NOPTS_VALUE;
+ rtpctx->first_rtcp_ntp_time = AV_NOPTS_VALUE;
+ if (st)
+ rtpctx->range_start_offset = av_rescale_q(reply->range_start,
+ AV_TIME_BASE_Q,
+ st->time_base);
+ }
+ }
}
- rt->state = RTSP_STATE_PLAYING;
+ rt->state = RTSP_STATE_STREAMING;
return 0;
}
-static int rtsp_read_header(AVFormatContext *s,
- AVFormatParameters *ap)
+static int rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply)
+{
+ RTSPState *rt = s->priv_data;
+ char cmd[1024];
+ unsigned char *content = NULL;
+ int ret;
+
+ /* describe the stream */
+ snprintf(cmd, sizeof(cmd),
+ "Accept: application/sdp\r\n");
+ if (rt->server_type == RTSP_SERVER_REAL) {
+ /**
+ * The Require: attribute is needed for proper streaming from
+ * Realmedia servers.
+ */
+ av_strlcat(cmd,
+ "Require: com.real.retain-entity-for-setup\r\n",
+ sizeof(cmd));
+ }
+ ff_rtsp_send_cmd(s, "DESCRIBE", rt->control_uri, cmd, reply, &content);
+ if (!content)
+ return AVERROR_INVALIDDATA;
+ if (reply->status_code != RTSP_STATUS_OK) {
+ av_freep(&content);
+ return AVERROR_INVALIDDATA;
+ }
+
+ /* now we got the SDP description, we parse it */
+ ret = sdp_parse(s, (const char *)content);
+ av_freep(&content);
+ if (ret < 0)
+ return AVERROR_INVALIDDATA;
+
+ return 0;
+}
+
+static int rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
+{
+ RTSPState *rt = s->priv_data;
+ RTSPMessageHeader reply1, *reply = &reply1;
+ int i;
+ char *sdp;
+ AVFormatContext sdp_ctx, *ctx_array[1];
+
+ rt->start_time = av_gettime();
+
+ /* Announce the stream */
+ sdp = av_mallocz(8192);
+ if (sdp == NULL)
+ return AVERROR(ENOMEM);
+ /* We create the SDP based on the RTSP AVFormatContext where we
+ * aren't allowed to change the filename field. (We create the SDP
+ * based on the RTSP context since the contexts for the RTP streams
+ * don't exist yet.) In order to specify a custom URL with the actual
+ * peer IP instead of the originally specified hostname, we create
+ * a temporary copy of the AVFormatContext, where the custom URL is set.
+ *
+ * FIXME: Create the SDP without copying the AVFormatContext.
+ * This either requires setting up the RTP stream AVFormatContexts
+ * already here (complicating things immensely) or getting a more
+ * flexible SDP creation interface.
+ */
+ sdp_ctx = *s;
+ ff_url_join(sdp_ctx.filename, sizeof(sdp_ctx.filename),
+ "rtsp", NULL, addr, -1, NULL);
+ ctx_array[0] = &sdp_ctx;
+ if (avf_sdp_create(ctx_array, 1, sdp, 8192)) {
+ av_free(sdp);
+ return AVERROR_INVALIDDATA;
+ }
+ av_log(s, AV_LOG_INFO, "SDP:\n%s\n", sdp);
+ ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri,
+ "Content-Type: application/sdp\r\n",
+ reply, NULL, sdp, strlen(sdp));
+ av_free(sdp);
+ if (reply->status_code != RTSP_STATUS_OK)
+ return AVERROR_INVALIDDATA;
+
+ /* Set up the RTSPStreams for each AVStream */
+ for (i = 0; i < s->nb_streams; i++) {
+ RTSPStream *rtsp_st;
+ AVStream *st = s->streams[i];
+
+ rtsp_st = av_mallocz(sizeof(RTSPStream));
+ if (!rtsp_st)
+ return AVERROR(ENOMEM);
+ dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
+
+ st->priv_data = rtsp_st;
+ rtsp_st->stream_index = i;
+
+ av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url));
+ /* Note, this must match the relative uri set in the sdp content */
+ av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url),
+ "/streamid=%d", i);
+ }
+
+ return 0;
+}
+
+void ff_rtsp_close_connections(AVFormatContext *s)
+{
+ RTSPState *rt = s->priv_data;
+ if (rt->rtsp_hd_out != rt->rtsp_hd) url_close(rt->rtsp_hd_out);
+ url_close(rt->rtsp_hd);
+}
+
+int ff_rtsp_connect(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
char *option_list, *option, *filename;
- URLContext *rtsp_hd;
- int port, ret, err;
- RTSPMessageHeader reply1, *reply = &reply1;
- unsigned char *content = NULL;
+ URLContext *rtsp_hd, *rtsp_hd_out;
+ int port, err, tcp_fd;
+ RTSPMessageHeader reply1 = {}, *reply = &reply1;
int lower_transport_mask = 0;
char real_challenge[64];
- redirect:
+ struct sockaddr_storage peer;
+ socklen_t peer_len = sizeof(peer);
+
+ if (!ff_network_init())
+ return AVERROR(EIO);
+redirect:
+ rt->control_transport = RTSP_MODE_PLAIN;
/* extract hostname and port */
- url_split(NULL, 0, auth, sizeof(auth),
- host, sizeof(host), &port, path, sizeof(path), s->filename);
+ ff_url_split(NULL, 0, auth, sizeof(auth),
+ host, sizeof(host), &port, path, sizeof(path), s->filename);
if (*auth) {
- int auth_len = strlen(auth), b64_len = ((auth_len + 2) / 3) * 4 + 1;
-
- if (!(rt->auth_b64 = av_malloc(b64_len)))
- return AVERROR(ENOMEM);
- if (!av_base64_encode(rt->auth_b64, b64_len, auth, auth_len)) {
- err = AVERROR(EINVAL);
- goto fail;
- }
+ av_strlcpy(rt->auth, auth, sizeof(rt->auth));
}
if (port < 0)
port = RTSP_DEFAULT_PORT;
/* search for options */
- option_list = strchr(path, '?');
+ option_list = strrchr(path, '?');
if (option_list) {
- filename = strchr(s->filename, '?');
- while(option_list) {
+ /* Strip out the RTSP specific options, write out the rest of
+ * the options back into the same string. */
+ filename = option_list;
+ while (option_list) {
/* move the option pointer */
option = ++option_list;
option_list = strchr(option_list, '&');
*option_list = 0;
/* handle the options */
- if (strcmp(option, "udp") == 0)
- lower_transport_mask = (1<< RTSP_LOWER_TRANSPORT_UDP);
- else if (strcmp(option, "multicast") == 0)
- lower_transport_mask = (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST);
- else if (strcmp(option, "tcp") == 0)
- lower_transport_mask = (1<< RTSP_LOWER_TRANSPORT_TCP);
- else {
- strcpy(++filename, option);
- filename += strlen(option);
+ if (!strcmp(option, "udp")) {
+ lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP);
+ } else if (!strcmp(option, "multicast")) {
+ lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST);
+ } else if (!strcmp(option, "tcp")) {
+ lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
+ } else if(!strcmp(option, "http")) {
+ lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
+ rt->control_transport = RTSP_MODE_TUNNEL;
+ } else {
+ /* Write options back into the buffer, using memmove instead
+ * of strcpy since the strings may overlap. */
+ int len = strlen(option);
+ memmove(++filename, option, len);
+ filename += len;
if (option_list) *filename = '&';
}
}
if (!lower_transport_mask)
lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
- /* open the tcp connexion */
- snprintf(tcpname, sizeof(tcpname), "tcp://%s:%d", host, port);
- if (url_open(&rtsp_hd, tcpname, URL_RDWR) < 0) {
- err = AVERROR(EIO);
- goto fail;
+ if (s->oformat) {
+ /* Only UDP or TCP - UDP multicast isn't supported. */
+ lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
+ (1 << RTSP_LOWER_TRANSPORT_TCP);
+ if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
+ av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
+ "only UDP and TCP are supported for output.\n");
+ err = AVERROR(EINVAL);
+ goto fail;
+ }
+ }
+
+ /* Construct the URI used in request; this is similar to s->filename,
+ * but with authentication credentials removed and RTSP specific options
+ * stripped out. */
+ ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
+ host, port, "%s", path);
+
+ if (rt->control_transport == RTSP_MODE_TUNNEL) {
+ /* set up initial handshake for tunneling */
+ char httpname[1024];
+ char sessioncookie[17];
+ char headers[1024];
+
+ ff_url_join(httpname, sizeof(httpname), "http", NULL, host, port, "%s", path);
+ snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
+ av_get_random_seed(), av_get_random_seed());
+
+ /* GET requests */
+ if (url_open(&rtsp_hd, httpname, URL_RDONLY) < 0) {
+ err = AVERROR(EIO);
+ goto fail;
+ }
+
+ /* generate GET headers */
+ snprintf(headers, sizeof(headers),
+ "x-sessioncookie: %s\r\n"
+ "Accept: application/x-rtsp-tunnelled\r\n"
+ "Pragma: no-cache\r\n"
+ "Cache-Control: no-cache\r\n",
+ sessioncookie);
+ ff_http_set_headers(rtsp_hd, headers);
+
+ /* complete the connection */
+ if (url_read(rtsp_hd, NULL, 0)) {
+ err = AVERROR(EIO);
+ goto fail;
+ }
+
+ /* POST requests */
+ if (url_open(&rtsp_hd_out, httpname, URL_WRONLY) < 0 ) {
+ err = AVERROR(EIO);
+ goto fail;
+ }
+
+ /* generate POST headers */
+ snprintf(headers, sizeof(headers),
+ "x-sessioncookie: %s\r\n"
+ "Content-Type: application/x-rtsp-tunnelled\r\n"
+ "Pragma: no-cache\r\n"
+ "Cache-Control: no-cache\r\n"
+ "Content-Length: 32767\r\n"
+ "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
+ sessioncookie);
+ ff_http_set_headers(rtsp_hd_out, headers);
+ ff_http_set_chunked_transfer_encoding(rtsp_hd_out, 0);
+
+ } else {
+ /* open the tcp connection */
+ ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
+ if (url_open(&rtsp_hd, tcpname, URL_RDWR) < 0) {
+ err = AVERROR(EIO);
+ goto fail;
+ }
+ rtsp_hd_out = rtsp_hd;
}
rt->rtsp_hd = rtsp_hd;
+ rt->rtsp_hd_out = rtsp_hd_out;
rt->seq = 0;
- /* request options supported by the server; this also detects server type */
+ tcp_fd = url_get_file_handle(rtsp_hd);
+ if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
+ getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
+ NULL, 0, NI_NUMERICHOST);
+ }
+
+ /* request options supported by the server; this also detects server
+ * type */
for (rt->server_type = RTSP_SERVER_RTP;;) {
- snprintf(cmd, sizeof(cmd),
- "OPTIONS %s RTSP/1.0\r\n", s->filename);
+ cmd[0] = 0;
if (rt->server_type == RTSP_SERVER_REAL)
av_strlcat(cmd,
/**
"CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
"GUID: 00000000-0000-0000-0000-000000000000\r\n",
sizeof(cmd));
- rtsp_send_cmd(s, cmd, reply, NULL);
+ ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
if (reply->status_code != RTSP_STATUS_OK) {
err = AVERROR_INVALIDDATA;
goto fail;
continue;
} else if (!strncasecmp(reply->server, "WMServer/", 9)) {
rt->server_type = RTSP_SERVER_WMS;
- } else if (rt->server_type == RTSP_SERVER_REAL) {
+ } else if (rt->server_type == RTSP_SERVER_REAL)
strcpy(real_challenge, reply->real_challenge);
- }
break;
}
- /* describe the stream */
- snprintf(cmd, sizeof(cmd),
- "DESCRIBE %s RTSP/1.0\r\n"
- "Accept: application/sdp\r\n",
- s->filename);
- if (rt->server_type == RTSP_SERVER_REAL) {
- /**
- * The Require: attribute is needed for proper streaming from
- * Realmedia servers.
- */
- av_strlcat(cmd,
- "Require: com.real.retain-entity-for-setup\r\n",
- sizeof(cmd));
- }
- rtsp_send_cmd(s, cmd, reply, &content);
- if (!content) {
- err = AVERROR_INVALIDDATA;
- goto fail;
- }
- if (reply->status_code != RTSP_STATUS_OK) {
- err = AVERROR_INVALIDDATA;
- goto fail;
- }
-
- /* now we got the SDP description, we parse it */
- ret = sdp_parse(s, (const char *)content);
- av_freep(&content);
- if (ret < 0) {
- err = AVERROR_INVALIDDATA;
+ if (s->iformat)
+ err = rtsp_setup_input_streams(s, reply);
+ else
+ err = rtsp_setup_output_streams(s, host);
+ if (err)
goto fail;
- }
do {
- int lower_transport = ff_log2_tab[lower_transport_mask & ~(lower_transport_mask - 1)];
+ int lower_transport = ff_log2_tab[lower_transport_mask &
+ ~(lower_transport_mask - 1)];
err = make_setup_request(s, host, port, lower_transport,
rt->server_type == RTSP_SERVER_REAL ?
goto fail;
lower_transport_mask &= ~(1 << lower_transport);
if (lower_transport_mask == 0 && err == 1) {
- err = AVERROR(FF_NETERROR(EPROTONOSUPPORT));
+ err = FF_NETERROR(EPROTONOSUPPORT);
goto fail;
}
} while (err);
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;
- }
- }
+ rt->seek_timestamp = 0; /* default is to start stream at position zero */
return 0;
fail:
- rtsp_close_streams(rt);
- av_freep(&content);
- url_close(rt->rtsp_hd);
- if (reply->status_code >=300 && reply->status_code < 400) {
+ 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));
av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
reply->status_code,
s->filename);
goto redirect;
}
+ ff_network_close();
return err;
}
+#endif
+
+#if CONFIG_RTSP_DEMUXER
+static int rtsp_read_header(AVFormatContext *s,
+ AVFormatParameters *ap)
+{
+ int ret;
+
+ ret = ff_rtsp_connect(s);
+ if (ret)
+ return ret;
+
+ if (ap->initial_pause) {
+ /* do not start immediately */
+ } else {
+ if (rtsp_read_play(s) < 0) {
+ ff_rtsp_close_streams(s);
+ ff_rtsp_close_connections(s);
+ return AVERROR_INVALIDDATA;
+ }
+ }
+
+ return 0;
+}
+
+static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
+ uint8_t *buf, int buf_size)
+{
+ RTSPState *rt = s->priv_data;
+ RTSPStream *rtsp_st;
+ fd_set rfds;
+ int fd, fd_max, n, i, ret, tcp_fd, timeout_cnt = 0;
+ struct timeval tv;
+
+ for (;;) {
+ if (url_interrupt_cb())
+ return AVERROR(EINTR);
+ FD_ZERO(&rfds);
+ if (rt->rtsp_hd) {
+ tcp_fd = fd_max = url_get_file_handle(rt->rtsp_hd);
+ FD_SET(tcp_fd, &rfds);
+ } else {
+ fd_max = 0;
+ tcp_fd = -1;
+ }
+ for (i = 0; i < rt->nb_rtsp_streams; i++) {
+ rtsp_st = rt->rtsp_streams[i];
+ if (rtsp_st->rtp_handle) {
+ /* currently, we cannot probe RTCP handle because of
+ * blocking restrictions */
+ fd = url_get_file_handle(rtsp_st->rtp_handle);
+ if (fd > fd_max)
+ fd_max = fd;
+ FD_SET(fd, &rfds);
+ }
+ }
+ tv.tv_sec = 0;
+ tv.tv_usec = SELECT_TIMEOUT_MS * 1000;
+ n = select(fd_max + 1, &rfds, NULL, NULL, &tv);
+ if (n > 0) {
+ timeout_cnt = 0;
+ for (i = 0; i < rt->nb_rtsp_streams; i++) {
+ rtsp_st = rt->rtsp_streams[i];
+ if (rtsp_st->rtp_handle) {
+ fd = url_get_file_handle(rtsp_st->rtp_handle);
+ if (FD_ISSET(fd, &rfds)) {
+ ret = url_read(rtsp_st->rtp_handle, buf, buf_size);
+ if (ret > 0) {
+ *prtsp_st = rtsp_st;
+ return ret;
+ }
+ }
+ }
+ }
+#if CONFIG_RTSP_DEMUXER
+ if (tcp_fd != -1 && FD_ISSET(tcp_fd, &rfds)) {
+ RTSPMessageHeader reply;
+
+ ret = ff_rtsp_read_reply(s, &reply, NULL, 0);
+ if (ret < 0)
+ return ret;
+ /* XXX: parse message */
+ if (rt->state != RTSP_STATE_STREAMING)
+ return 0;
+ }
+#endif
+ } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
+ return FF_NETERROR(ETIMEDOUT);
+ } else if (n < 0 && errno != EINTR)
+ return AVERROR(errno);
+ }
+}
static int tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
uint8_t *buf, int buf_size)
#ifdef DEBUG_RTP_TCP
dprintf(s, "tcp_read_packet:\n");
#endif
- redo:
- for(;;) {
+redo:
+ for (;;) {
RTSPMessageHeader reply;
- ret = rtsp_read_reply(s, &reply, NULL, 1);
+ ret = ff_rtsp_read_reply(s, &reply, NULL, 1);
if (ret == -1)
return -1;
if (ret == 1) /* received '$' */
break;
/* XXX: parse message */
- if (rt->state != RTSP_STATE_PLAYING)
+ if (rt->state != RTSP_STATE_STREAMING)
return 0;
}
ret = url_read_complete(rt->rtsp_hd, buf, 3);
if (ret != 3)
return -1;
- id = buf[0];
+ id = buf[0];
len = AV_RB16(buf + 1);
#ifdef DEBUG_RTP_TCP
dprintf(s, "id=%d len=%d\n", id, len);
return -1;
/* find the matching stream */
- for(i = 0; i < rt->nb_rtsp_streams; i++) {
+ 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:
+found:
*prtsp_st = rtsp_st;
return len;
}
-static int rtsp_read_packet(AVFormatContext *s,
- AVPacket *pkt)
+static int rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
+{
+ RTSPState *rt = s->priv_data;
+ int ret, len;
+ uint8_t buf[10 * RTP_MAX_PACKET_LENGTH];
+ RTSPStream *rtsp_st;
+
+ /* get next frames from the same RTP packet */
+ if (rt->cur_transport_priv) {
+ if (rt->transport == RTSP_TRANSPORT_RDT) {
+ ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
+ } else
+ ret = rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
+ if (ret == 0) {
+ rt->cur_transport_priv = NULL;
+ return 0;
+ } else if (ret == 1) {
+ return 0;
+ } else
+ rt->cur_transport_priv = NULL;
+ }
+
+ /* read next RTP packet */
+ redo:
+ switch(rt->lower_transport) {
+ default:
+#if CONFIG_RTSP_DEMUXER
+ case RTSP_LOWER_TRANSPORT_TCP:
+ len = tcp_read_packet(s, &rtsp_st, buf, sizeof(buf));
+ break;
+#endif
+ case RTSP_LOWER_TRANSPORT_UDP:
+ case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
+ len = udp_read_packet(s, &rtsp_st, buf, sizeof(buf));
+ if (len >=0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
+ rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
+ break;
+ }
+ if (len < 0)
+ return len;
+ if (len == 0)
+ return AVERROR_EOF;
+ if (rt->transport == RTSP_TRANSPORT_RDT) {
+ ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, buf, len);
+ } else {
+ ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, buf, len);
+ if (ret < 0) {
+ /* Either bad packet, or a RTCP packet. Check if the
+ * first_rtcp_ntp_time field was initialized. */
+ RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
+ if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
+ /* first_rtcp_ntp_time has been initialized for this stream,
+ * copy the same value to all other uninitialized streams,
+ * in order to map their timestamp origin to the same ntp time
+ * as this one. */
+ int i;
+ for (i = 0; i < rt->nb_rtsp_streams; i++) {
+ RTPDemuxContext *rtpctx2 = rtsp_st->transport_priv;
+ if (rtpctx2 &&
+ rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE)
+ rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
+ }
+ }
+ }
+ }
+ if (ret < 0)
+ goto redo;
+ if (ret == 1)
+ /* more packets may follow, so we save the RTP context */
+ rt->cur_transport_priv = rtsp_st->transport_priv;
+
+ return ret;
+}
+
+static int rtsp_read_packet(AVFormatContext *s, AVPacket *pkt)
{
RTSPState *rt = s->priv_data;
int ret;
if (!rt->need_subscription) {
if (memcmp (cache, rt->real_setup_cache,
sizeof(enum AVDiscard) * s->nb_streams)) {
- av_strlcatf(cmd, sizeof(cmd),
- "SET_PARAMETER %s RTSP/1.0\r\n"
- "Unsubscribe: %s\r\n",
- s->filename, rt->last_subscription);
- rtsp_send_cmd(s, cmd, reply, NULL);
+ snprintf(cmd, sizeof(cmd),
+ "Unsubscribe: %s\r\n",
+ rt->last_subscription);
+ ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri,
+ cmd, reply, NULL);
if (reply->status_code != RTSP_STATUS_OK)
return AVERROR_INVALIDDATA;
rt->need_subscription = 1;
rt->last_subscription[0] = 0;
snprintf(cmd, sizeof(cmd),
- "SET_PARAMETER %s RTSP/1.0\r\n"
- "Subscribe: ",
- s->filename);
+ "Subscribe: ");
for (i = 0; i < rt->nb_rtsp_streams; i++) {
rule_nr = 0;
for (r = 0; r < s->nb_streams; r++) {
}
}
av_strlcatf(cmd, sizeof(cmd), "%s\r\n", rt->last_subscription);
- rtsp_send_cmd(s, cmd, reply, NULL);
+ ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri,
+ cmd, reply, NULL);
if (reply->status_code != RTSP_STATUS_OK)
return AVERROR_INVALIDDATA;
rt->need_subscription = 0;
- if (rt->state == RTSP_STATE_PLAYING)
+ if (rt->state == RTSP_STATE_STREAMING)
rtsp_read_play (s);
}
}
ret = rtsp_fetch_packet(s, pkt);
- if (ret < 0) {
+ if (ret < 0)
return ret;
- }
/* send dummy request to keep TCP connection alive */
if ((rt->server_type == RTSP_SERVER_WMS ||
rt->server_type == RTSP_SERVER_REAL) &&
(av_gettime() - rt->last_cmd_time) / 1000000 >= rt->timeout / 2) {
if (rt->server_type == RTSP_SERVER_WMS) {
- snprintf(cmd, sizeof(cmd) - 1,
- "GET_PARAMETER %s RTSP/1.0\r\n",
- s->filename);
- rtsp_send_cmd_async(s, cmd, reply, NULL);
+ ff_rtsp_send_cmd_async(s, "GET_PARAMETER", rt->control_uri, NULL);
} else {
- rtsp_send_cmd_async(s, "OPTIONS * RTSP/1.0\r\n",
- reply, NULL);
+ ff_rtsp_send_cmd_async(s, "OPTIONS", "*", NULL);
}
}
{
RTSPState *rt = s->priv_data;
RTSPMessageHeader reply1, *reply = &reply1;
- char cmd[1024];
-
- rt = s->priv_data;
- if (rt->state != RTSP_STATE_PLAYING)
+ if (rt->state != RTSP_STATE_STREAMING)
return 0;
else if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) {
- snprintf(cmd, sizeof(cmd),
- "PAUSE %s RTSP/1.0\r\n",
- s->filename);
- rtsp_send_cmd(s, cmd, reply, NULL);
+ ff_rtsp_send_cmd(s, "PAUSE", rt->control_uri, NULL, reply, NULL);
if (reply->status_code != RTSP_STATUS_OK) {
return -1;
}
{
RTSPState *rt = s->priv_data;
- rt->seek_timestamp = av_rescale_q(timestamp, s->streams[stream_index]->time_base, AV_TIME_BASE_Q);
+ rt->seek_timestamp = av_rescale_q(timestamp,
+ s->streams[stream_index]->time_base,
+ AV_TIME_BASE_Q);
switch(rt->state) {
default:
case RTSP_STATE_IDLE:
break;
- case RTSP_STATE_PLAYING:
+ case RTSP_STATE_STREAMING:
if (rtsp_read_pause(s) != 0)
return -1;
rt->state = RTSP_STATE_SEEKING;
static int rtsp_read_close(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
- RTSPMessageHeader reply1, *reply = &reply1;
- char cmd[1024];
#if 0
/* NOTE: it is valid to flush the buffer here */
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);
+ ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
- rtsp_close_streams(rt);
- url_close(rt->rtsp_hd);
+ ff_rtsp_close_streams(s);
+ ff_rtsp_close_connections(s);
+ ff_network_close();
return 0;
}
/* we look for a line beginning "c=IN IP4" */
while (p < p_end && *p != '\0') {
- if (p + sizeof("c=IN IP4") - 1 < p_end && av_strstart(p, "c=IN IP4", NULL))
+ if (p + sizeof("c=IN IP4") - 1 < p_end &&
+ av_strstart(p, "c=IN IP4", NULL))
return AVPROBE_SCORE_MAX / 2;
- while(p < p_end - 1 && *p != '\n') p++;
+ while (p < p_end - 1 && *p != '\n') p++;
if (++p >= p_end)
break;
if (*p == '\r')
#define SDP_MAX_SIZE 8192
-static int sdp_read_header(AVFormatContext *s,
- AVFormatParameters *ap)
+static int sdp_read_header(AVFormatContext *s, AVFormatParameters *ap)
{
RTSPState *rt = s->priv_data;
RTSPStream *rtsp_st;
char *content;
char url[1024];
+ if (!ff_network_init())
+ return AVERROR(EIO);
+
/* read the whole sdp file */
/* XXX: better loading */
content = av_malloc(SDP_MAX_SIZE);
av_free(content);
/* open each RTP stream */
- for(i=0;i<rt->nb_rtsp_streams;i++) {
+ for (i = 0; i < rt->nb_rtsp_streams; i++) {
rtsp_st = rt->rtsp_streams[i];
- snprintf(url, sizeof(url), "rtp://%s:%d?localport=%d&ttl=%d",
- inet_ntoa(rtsp_st->sdp_ip),
- rtsp_st->sdp_port,
- rtsp_st->sdp_port,
- rtsp_st->sdp_ttl);
+ ff_url_join(url, sizeof(url), "rtp", NULL,
+ inet_ntoa(rtsp_st->sdp_ip), rtsp_st->sdp_port,
+ "?localport=%d&ttl=%d", rtsp_st->sdp_port,
+ rtsp_st->sdp_ttl);
if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
err = AVERROR_INVALIDDATA;
goto fail;
goto fail;
}
return 0;
- fail:
- rtsp_close_streams(rt);
+fail:
+ ff_rtsp_close_streams(s);
+ ff_network_close();
return err;
}
static int sdp_read_close(AVFormatContext *s)
{
- RTSPState *rt = s->priv_data;
- rtsp_close_streams(rt);
+ ff_rtsp_close_streams(s);
+ ff_network_close();
return 0;
}
projects / ffmpeg / blobdiff