#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 "internal.h"
#include "network.h"
#include "os_support.h"
+#include "http.h"
#include "rtsp.h"
#include "rtpdec.h"
#include "rdt.h"
-#include "rtpdec_asf.h"
-#include "rtpdec_vorbis.h"
+#include "rtpdec_formats.h"
+#include "rtpenc_chain.h"
//#define DEBUG
//#define DEBUG_RTP_TCP
-#if LIBAVFORMAT_VERSION_INT < (53 << 16)
-int rtsp_default_protocols = (1 << RTSP_LOWER_TRANSPORT_UDP);
-#endif
-
-#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'. */
-#define redir_isspace(c) memchr(SPACE_CHARS, c, 4)
-static void skip_spaces(const char **pp)
-{
- const char *p;
- p = *pp;
- while (redir_isspace(*p))
- p++;
- *pp = p;
-}
+/* 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 SDP_MAX_SIZE 16384
+#define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
static void get_word_until_chars(char *buf, int buf_size,
const char *sep, const char **pp)
char *q;
p = *pp;
- skip_spaces(&p);
+ p += strspn(p, SPACE_CHARS);
q = buf;
while (!strchr(sep, *p) && *p != '\0') {
if ((q - buf) < buf_size - 1)
get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
}
+/** 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];
+
+ p += strspn(p, SPACE_CHARS);
+ if (!av_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);
+}
+
+static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
+{
+ struct addrinfo hints, *ai = NULL;
+ memset(&hints, 0, sizeof(hints));
+ hints.ai_flags = AI_NUMERICHOST;
+ if (getaddrinfo(buf, NULL, &hints, &ai))
+ return -1;
+ memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
+ freeaddrinfo(ai);
+ return 0;
+}
+
+#if CONFIG_RTPDEC
+static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
+ RTSPStream *rtsp_st, AVCodecContext *codec)
+{
+ if (!handler)
+ return;
+ codec->codec_id = handler->codec_id;
+ rtsp_st->dynamic_handler = handler;
+ if (handler->open)
+ rtsp_st->dynamic_protocol_context = handler->open();
+}
+
/* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
static int sdp_parse_rtpmap(AVFormatContext *s,
- AVCodecContext *codec, RTSPStream *rtsp_st,
+ AVStream *st, RTSPStream *rtsp_st,
int payload_type, const char *p)
{
+ AVCodecContext *codec = st->codec;
char buf[256];
int i;
AVCodec *c;
* have a trailing space. */
get_word_sep(buf, sizeof(buf), "/ ", &p);
if (payload_type >= RTP_PT_PRIVATE) {
- 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;
- }
- }
+ RTPDynamicProtocolHandler *handler =
+ ff_rtp_handler_find_by_name(buf, codec->codec_type);
+ init_rtp_handler(handler, rtsp_st, codec);
+ /* If no dynamic handler was found, check with the list of standard
+ * allocated types, if such a stream for some reason happens to
+ * use a private payload type. This isn't handled in rtpdec.c, since
+ * the format name from the rtpmap line never is passed into rtpdec. */
+ if (!rtsp_st->dynamic_handler)
+ codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
} else {
/* We are in a standard case
* (from http://www.iana.org/assignments/rtp-parameters). */
get_word_sep(buf, sizeof(buf), "/", &p);
i = atoi(buf);
switch (codec->codec_type) {
- case CODEC_TYPE_AUDIO:
+ 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;
+ av_set_pts_info(st, 32, 1, codec->sample_rate);
get_word_sep(buf, sizeof(buf), "/", &p);
i = atoi(buf);
if (i > 0)
av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
codec->channels);
break;
- case CODEC_TYPE_VIDEO:
+ case AVMEDIA_TYPE_VIDEO:
av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
+ if (i > 0)
+ av_set_pts_info(st, 32, 1, i);
break;
default:
break;
return 0;
}
-/* 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 (;;) {
- skip_spaces(&p);
- if (*p == '\0')
- break;
- c = toupper((unsigned char) *p++);
- if (c >= '0' && c <= '9')
- c = c - '0';
- else if (c >= 'A' && c <= 'F')
- c = c - 'A' + 10;
- else
- break;
- v = (v << 4) | c;
- if (v & 0x100) {
- if (data)
- data[len] = v;
- len++;
- v = 1;
- }
- }
- return len;
-}
-
-static void sdp_parse_fmtp_config(AVCodecContext * codec, void *ctx,
- 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;
- }
- return;
-}
-
-typedef struct {
- const char *str;
- uint16_t type;
- uint32_t offset;
-} AttrNameMap;
-
-/* All known fmtp parmeters and the corresping 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 },
-};
-
-/* parse the attribute line from the fmtp a line of an sdp resonse. This
+/* 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);
+ *p += strspn(*p, SPACE_CHARS);
if (**p) {
get_word_sep(attr, attr_size, "=", p);
if (**p == '=')
return 0;
}
-/* parse a SDP line and save stream attributes */
-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];
- 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 (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);
- }
- }
- }
-}
-
-/** 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 (!av_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;
+ struct sockaddr_storage default_ip;
int default_ttl;
int skip_media; ///< set if an unknown m= line occurs
} SDPParseState;
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;
- struct in_addr sdp_ip;
+ struct sockaddr_storage sdp_ip;
int ttl;
dprintf(s, "sdp: %c='%s'\n", letter, buf);
if (strcmp(buf1, "IN") != 0)
return;
get_word(buf1, sizeof(buf1), &p);
- if (strcmp(buf1, "IP4") != 0)
+ if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
return;
get_word_sep(buf1, sizeof(buf1), "/", &p);
- if (ff_inet_aton(buf1, &sdp_ip) == 0)
+ if (get_sockaddr(buf1, &sdp_ip))
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;
rtsp_st->stream_index = st->index;
st->codec->codec_type = codec_type;
if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
+ RTPDynamicProtocolHandler *handler;
/* if standard payload type, we can find the codec right now */
ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
+ if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
+ st->codec->sample_rate > 0)
+ av_set_pts_info(st, 32, 1, st->codec->sample_rate);
+ /* Even static payload types may need a custom depacketizer */
+ handler = ff_rtp_handler_find_by_id(
+ rtsp_st->sdp_payload_type, st->codec->codec_type);
+ init_rtp_handler(handler, rtsp_st, st->codec);
}
}
/* put a default control url */
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;
+ 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 */
- ff_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
- NULL, NULL, 0, p);
- if (proto[0] == '\0') {
- /* relative 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));
+ /* XXX: may need to add full url resolution */
+ av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
+ NULL, NULL, 0, p);
+ if (proto[0] == '\0') {
+ /* relative 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(s, st->codec, rtsp_st, payload_type, p);
- } else if (av_strstart(p, "fmtp:", &p)) {
+ sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
+ } else if (av_strstart(p, "fmtp:", &p) ||
+ av_strstart(p, "framesize:", &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];
- 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)))
- sdp_parse_fmtp(st, 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);
} else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
if (atoi(p) == 1)
rt->transport = RTSP_TRANSPORT_RDT;
+ } else if (av_strstart(p, "SampleRate:integer;", &p) &&
+ s->nb_streams > 0) {
+ st = s->streams[s->nb_streams - 1];
+ st->codec->sample_rate = atoi(p);
} else {
if (rt->server_type == RTSP_SERVER_WMS)
ff_wms_parse_sdp_a_line(s, p);
}
}
-static int sdp_parse(AVFormatContext *s, const char *content)
+int ff_sdp_parse(AVFormatContext *s, const char *content)
{
const char *p;
int letter;
* "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 (;;) {
- skip_spaces(&p);
+ p += strspn(p, SPACE_CHARS);
letter = *p;
if (letter == '\0')
break;
}
return 0;
}
+#endif /* CONFIG_RTPDEC */
/* close and free RTSP streams */
void ff_rtsp_close_streams(AVFormatContext *s)
av_metadata_free(&rtpctx->metadata);
av_free(rtpctx->streams[0]);
av_free(rtpctx);
- } else if (rt->transport == RTSP_TRANSPORT_RDT)
+ } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
ff_rdt_parse_close(rtsp_st->transport_priv);
- else
+ else if (CONFIG_RTPDEC)
rtp_parse_close(rtsp_st->transport_priv);
}
if (rtsp_st->rtp_handle)
av_close_input_stream (rt->asf_ctx);
rt->asf_ctx = NULL;
}
- av_freep(&rt->auth_b64);
-}
-
-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;
+ av_free(rt->recvbuf);
}
static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
if (!st)
s->ctx_flags |= AVFMTCTX_NOHEADER;
- if (s->oformat) {
- rtsp_st->transport_priv = rtsp_rtp_mux_open(s, st, rtsp_st->rtp_handle);
- /* Ownage of rtp_handle is passed to the rtp mux context */
+ if (s->oformat && CONFIG_RTSP_MUXER) {
+ rtsp_st->transport_priv = ff_rtp_chain_mux_open(s, st,
+ rtsp_st->rtp_handle,
+ RTSP_TCP_MAX_PACKET_SIZE);
+ /* Ownership of rtp_handle is passed to the rtp mux context */
rtsp_st->rtp_handle = NULL;
- } else if (rt->transport == RTSP_TRANSPORT_RDT)
+ } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
rtsp_st->dynamic_protocol_context,
rtsp_st->dynamic_handler);
- else
+ else if (CONFIG_RTPDEC)
rtsp_st->transport_priv = rtp_parse_open(s, st, rtsp_st->rtp_handle,
rtsp_st->sdp_payload_type,
- &rtsp_st->rtp_payload_data);
+ (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
+ ? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
if (!rtsp_st->transport_priv) {
return AVERROR(ENOMEM);
- } else if (rt->transport != RTSP_TRANSPORT_RDT) {
+ } else if (rt->transport != RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) {
if (rtsp_st->dynamic_handler) {
rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
rtsp_st->dynamic_protocol_context,
}
#if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
-static int rtsp_probe(AVProbeData *p)
-{
- if (av_strstart(p->filename, "rtsp:", NULL))
- return AVPROBE_SCORE_MAX;
- return 0;
-}
-
static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
{
const char *p;
int v;
p = *pp;
- skip_spaces(&p);
+ p += strspn(p, SPACE_CHARS);
v = strtol(p, (char **)&p, 10);
if (*p == '-') {
p++;
reply->nb_transports = 0;
for (;;) {
- skip_spaces(&p);
+ p += strspn(p, SPACE_CHARS);
if (*p == '\0')
break;
th->ttl = strtol(p, (char **)&p, 10);
}
} else if (!strcmp(parameter, "destination")) {
- struct in_addr ipaddr;
-
if (*p == '=') {
p++;
get_word_sep(buf, sizeof(buf), ";,", &p);
- if (ff_inet_aton(buf, &ipaddr))
- th->destination = ntohl(ipaddr.s_addr);
+ get_sockaddr(buf, &th->destination);
+ }
+ } else if (!strcmp(parameter, "source")) {
+ if (*p == '=') {
+ p++;
+ get_word_sep(buf, sizeof(buf), ";,", &p);
+ av_strlcpy(th->source, buf, sizeof(th->source));
}
}
+
while (*p != ';' && *p != '\0' && *p != ',')
p++;
if (*p == ';')
}
}
-void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf)
+static void handle_rtp_info(RTSPState *rt, const char *url,
+ uint32_t seq, uint32_t rtptime)
+{
+ int i;
+ if (!rtptime || !url[0])
+ return;
+ if (rt->transport != RTSP_TRANSPORT_RTP)
+ return;
+ for (i = 0; i < rt->nb_rtsp_streams; i++) {
+ RTSPStream *rtsp_st = rt->rtsp_streams[i];
+ RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
+ if (!rtpctx)
+ continue;
+ if (!strcmp(rtsp_st->control_url, url)) {
+ rtpctx->base_timestamp = rtptime;
+ break;
+ }
+ }
+}
+
+static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
+{
+ int read = 0;
+ char key[20], value[1024], url[1024] = "";
+ uint32_t seq = 0, rtptime = 0;
+
+ for (;;) {
+ p += strspn(p, SPACE_CHARS);
+ if (!*p)
+ break;
+ get_word_sep(key, sizeof(key), "=", &p);
+ if (*p != '=')
+ break;
+ p++;
+ get_word_sep(value, sizeof(value), ";, ", &p);
+ read++;
+ if (!strcmp(key, "url"))
+ av_strlcpy(url, value, sizeof(url));
+ else if (!strcmp(key, "seq"))
+ seq = strtol(value, NULL, 10);
+ else if (!strcmp(key, "rtptime"))
+ rtptime = strtol(value, NULL, 10);
+ if (*p == ',') {
+ handle_rtp_info(rt, url, seq, rtptime);
+ url[0] = '\0';
+ seq = rtptime = 0;
+ read = 0;
+ }
+ if (*p)
+ p++;
+ }
+ if (read > 0)
+ handle_rtp_info(rt, url, seq, rtptime);
+}
+
+void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
+ RTSPState *rt, const char *method)
{
const char *p;
} else if (av_stristart(p, "Range:", &p)) {
rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
} else if (av_stristart(p, "RealChallenge1:", &p)) {
- skip_spaces(&p);
+ p += strspn(p, SPACE_CHARS);
av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
} else if (av_stristart(p, "Server:", &p)) {
- skip_spaces(&p);
+ p += strspn(p, SPACE_CHARS);
av_strlcpy(reply->server, p, sizeof(reply->server));
} else if (av_stristart(p, "Notice:", &p) ||
av_stristart(p, "X-Notice:", &p)) {
reply->notice = strtol(p, NULL, 10);
} else if (av_stristart(p, "Location:", &p)) {
- skip_spaces(&p);
+ p += strspn(p, SPACE_CHARS);
av_strlcpy(reply->location, p , sizeof(reply->location));
+ } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
+ p += strspn(p, SPACE_CHARS);
+ ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
+ } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
+ p += strspn(p, SPACE_CHARS);
+ ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
+ } else if (av_stristart(p, "Content-Base:", &p) && rt) {
+ p += strspn(p, SPACE_CHARS);
+ if (method && !strcmp(method, "DESCRIBE"))
+ av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
+ } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
+ p += strspn(p, SPACE_CHARS);
+ if (method && !strcmp(method, "PLAY"))
+ rtsp_parse_rtp_info(rt, p);
}
}
int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
unsigned char **content_ptr,
- int return_on_interleaved_data)
+ int return_on_interleaved_data, const char *method)
{
RTSPState *rt = s->priv_data;
char buf[4096], buf1[1024], *q;
dprintf(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
#endif
if (ret != 1)
- return -1;
+ return AVERROR_EOF;
if (ch == '\n')
break;
if (ch == '$') {
get_word(buf1, sizeof(buf1), &p);
get_word(buf1, sizeof(buf1), &p);
reply->status_code = atoi(buf1);
+ av_strlcpy(reply->reason, p, sizeof(reply->reason));
} else {
- ff_rtsp_parse_line(reply, p);
+ ff_rtsp_parse_line(reply, p, rt, method);
av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
}
return 0;
}
-void ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
- const char *cmd,
- const unsigned char *send_content,
- int send_content_length)
+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));
- if (send_content_length > 0 && send_content)
- url_write(rt->rtsp_hd, send_content, send_content_length);
+ 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;
}
-void ff_rtsp_send_cmd_async(AVFormatContext *s, const char *cmd)
+int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
+ const char *url, const char *headers)
{
- ff_rtsp_send_cmd_with_content_async(s, cmd, NULL, 0);
+ return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
}
-void ff_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)
{
- ff_rtsp_send_cmd_with_content(s, cmd, reply, content_ptr, NULL, 0);
+ return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
+ content_ptr, NULL, 0);
}
-void ff_rtsp_send_cmd_with_content(AVFormatContext *s,
- const char *cmd,
- RTSPMessageHeader *reply,
- unsigned char **content_ptr,
- const unsigned char *send_content,
- int send_content_length)
+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)
{
- ff_rtsp_send_cmd_with_content_async(s, cmd, send_content, 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;
- ff_rtsp_read_reply(s, reply, content_ptr, 0);
+ if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
+ return ret;
+
+ if (reply->status_code == 401 && cur_auth_type == HTTP_AUTH_NONE &&
+ rt->auth_state.auth_type != HTTP_AUTH_NONE)
+ goto retry;
+
+ if (reply->status_code > 400){
+ av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
+ method,
+ reply->status_code,
+ reply->reason);
+ av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
+ }
+
+ 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 ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
int lower_transport, const char *real_challenge)
{
RTSPState *rt = s->priv_data;
err = AVERROR_INVALIDDATA;
goto fail;
}
+#else
+ av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
+ err = AVERROR(EIO);
+ goto fail;
#endif
rtp_opened:
- port = rtp_get_local_port(rtsp_st->rtp_handle);
+ port = rtp_get_local_rtp_port(rtsp_st->rtp_handle);
have_port:
snprintf(transport, sizeof(transport) - 1,
"%s/UDP;", trans_pref);
* 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)
+ AVMEDIA_TYPE_DATA)
continue;
snprintf(transport, sizeof(transport) - 1,
"%s/TCP;", trans_pref);
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);
- if (i == 0 && rt->server_type == RTSP_SERVER_REAL) {
+ transport);
+ if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
char real_res[41], real_csum[9];
ff_rdt_calc_response_and_checksum(real_res, real_csum,
real_challenge);
"RealChallenge2: %s, sd=%s\r\n",
rt->session_id, real_res, real_csum);
}
- ff_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;
rt->transport = reply->transports[0].transport;
}
- /* close RTP connection if not choosen */
- if (reply->transports[0].lower_transport != RTSP_LOWER_TRANSPORT_UDP &&
- (lower_transport == RTSP_LOWER_TRANSPORT_UDP)) {
- url_close(rtsp_st->rtp_handle);
- rtsp_st->rtp_handle = NULL;
+ /* Fail if the server responded with another lower transport mode
+ * than what we requested. */
+ if (reply->transports[0].lower_transport != lower_transport) {
+ av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
+ err = AVERROR_INVALIDDATA;
+ goto fail;
}
switch(reply->transports[0].lower_transport) {
break;
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);
+ char url[1024], options[30] = "";
+
+ if (rt->filter_source)
+ av_strlcpy(options, "?connect=1", sizeof(options));
+ /* Use source address if specified */
+ if (reply->transports[0].source[0]) {
+ ff_url_join(url, sizeof(url), "rtp", NULL,
+ reply->transports[0].source,
+ reply->transports[0].server_port_min, options);
+ } else {
+ ff_url_join(url, sizeof(url), "rtp", NULL, host,
+ reply->transports[0].server_port_min, options);
+ }
if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
err = AVERROR_INVALIDDATA;
* 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)
+ if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
+ CONFIG_RTPDEC)
rtp_send_punch_packets(rtsp_st->rtp_handle);
break;
}
case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
- char url[1024];
- struct in_addr in;
+ char url[1024], namebuf[50];
+ struct sockaddr_storage addr;
int port, ttl;
- if (reply->transports[0].destination) {
- in.s_addr = htonl(reply->transports[0].destination);
+ if (reply->transports[0].destination.ss_family) {
+ addr = reply->transports[0].destination;
port = reply->transports[0].port_min;
ttl = reply->transports[0].ttl;
} else {
- in = rtsp_st->sdp_ip;
+ addr = 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),
+ getnameinfo((struct sockaddr*) &addr, sizeof(addr),
+ namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
+ ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
port, "?ttl=%d", ttl);
if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
err = AVERROR_INVALIDDATA;
return err;
}
-static int rtsp_read_play(AVFormatContext *s)
+void ff_rtsp_close_connections(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
- RTSPMessageHeader reply1, *reply = &reply1;
- 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",
- rt->control_uri);
- } else {
- snprintf(cmd, sizeof(cmd),
- "PLAY %s RTSP/1.0\r\n"
- "Range: npt=%0.3f-\r\n",
- rt->control_uri,
- (double)rt->seek_timestamp / AV_TIME_BASE);
- }
- ff_rtsp_send_cmd(s, cmd, reply, NULL);
- if (reply->status_code != RTSP_STATUS_OK) {
- return -1;
- }
- }
- rt->state = RTSP_STATE_STREAMING;
- return 0;
-}
-
-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),
- "DESCRIBE %s RTSP/1.0\r\n"
- "Accept: application/sdp\r\n",
- rt->control_uri);
- 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, 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;
- char cmd[1024];
- int i;
- char *sdp;
- AVFormatContext sdp_ctx, *ctx_array[1];
-
- rt->start_time = av_gettime();
-
- /* Announce the stream */
- snprintf(cmd, sizeof(cmd),
- "ANNOUNCE %s RTSP/1.0\r\n"
- "Content-Type: application/sdp\r\n",
- rt->control_uri);
- 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, cmd, 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;
+ if (rt->rtsp_hd_out != rt->rtsp_hd) url_close(rt->rtsp_hd_out);
+ url_close(rt->rtsp_hd);
+ rt->rtsp_hd = rt->rtsp_hd_out = NULL;
}
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, err, tcp_fd;
- RTSPMessageHeader reply1, *reply = &reply1;
+ RTSPMessageHeader reply1 = {0}, *reply = &reply1;
int lower_transport_mask = 0;
char real_challenge[64];
struct sockaddr_storage peer;
if (!ff_network_init())
return AVERROR(EIO);
redirect:
+ rt->control_transport = RTSP_MODE_PLAIN;
/* extract hostname and port */
- ff_url_split(NULL, 0, auth, sizeof(auth),
+ av_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;
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 if (!strcmp(option, "filter_src")) {
+ rt->filter_source = 1;
} else {
/* Write options back into the buffer, using memmove instead
* of strcpy since the strings may overlap. */
/* 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) {
+ 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);
}
}
- /* open the tcp connexion */
- 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;
+ /* 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", auth, host, port, "%s", path);
+ snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
+ av_get_random_seed(), av_get_random_seed());
+
+ /* GET requests */
+ if (url_alloc(&rt->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(rt->rtsp_hd, headers);
+
+ /* complete the connection */
+ if (url_connect(rt->rtsp_hd)) {
+ err = AVERROR(EIO);
+ goto fail;
+ }
+
+ /* POST requests */
+ if (url_alloc(&rt->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(rt->rtsp_hd_out, headers);
+ ff_http_set_chunked_transfer_encoding(rt->rtsp_hd_out, 0);
+
+ /* Initialize the authentication state for the POST session. The HTTP
+ * protocol implementation doesn't properly handle multi-pass
+ * authentication for POST requests, since it would require one of
+ * the following:
+ * - implementing Expect: 100-continue, which many HTTP servers
+ * don't support anyway, even less the RTSP servers that do HTTP
+ * tunneling
+ * - sending the whole POST data until getting a 401 reply specifying
+ * what authentication method to use, then resending all that data
+ * - waiting for potential 401 replies directly after sending the
+ * POST header (waiting for some unspecified time)
+ * Therefore, we copy the full auth state, which works for both basic
+ * and digest. (For digest, we would have to synchronize the nonce
+ * count variable between the two sessions, if we'd do more requests
+ * with the original session, though.)
+ */
+ ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
+
+ /* complete the connection */
+ if (url_connect(rt->rtsp_hd_out)) {
+ err = AVERROR(EIO);
+ goto fail;
+ }
+ } else {
+ /* open the tcp connection */
+ ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
+ if (url_open(&rt->rtsp_hd, tcpname, URL_RDWR) < 0) {
+ err = AVERROR(EIO);
+ goto fail;
+ }
+ rt->rtsp_hd_out = rt->rtsp_hd;
}
- rt->rtsp_hd = rtsp_hd;
rt->seq = 0;
- tcp_fd = url_get_file_handle(rtsp_hd);
+ tcp_fd = url_get_file_handle(rt->rtsp_hd);
if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
NULL, 0, NI_NUMERICHOST);
}
- /* 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);
/* 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", rt->control_uri);
+ 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));
- ff_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;
break;
}
- if (s->iformat)
- err = rtsp_setup_input_streams(s, reply);
- else
- err = rtsp_setup_output_streams(s, host);
+ if (s->iformat && CONFIG_RTSP_DEMUXER)
+ err = ff_rtsp_setup_input_streams(s, reply);
+ else if (CONFIG_RTSP_MUXER)
+ err = ff_rtsp_setup_output_streams(s, host);
if (err)
goto fail;
int lower_transport = ff_log2_tab[lower_transport_mask &
~(lower_transport_mask - 1)];
- err = make_setup_request(s, host, port, lower_transport,
+ err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
rt->server_type == RTSP_SERVER_REAL ?
real_challenge : NULL);
if (err < 0)
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);
return 0;
fail:
ff_rtsp_close_streams(s);
- url_close(rt->rtsp_hd);
+ 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",
ff_network_close();
return err;
}
-#endif
-
-#if CONFIG_RTSP_DEMUXER
-static int rtsp_read_header(AVFormatContext *s,
- AVFormatParameters *ap)
-{
- RTSPState *rt = s->priv_data;
- 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);
- url_close(rt->rtsp_hd);
- return AVERROR_INVALIDDATA;
- }
- }
-
- return 0;
-}
+#endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
+#if CONFIG_RTPDEC
static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
- uint8_t *buf, int buf_size)
+ uint8_t *buf, int buf_size, int64_t wait_end)
{
RTSPState *rt = s->priv_data;
RTSPStream *rtsp_st;
fd_set rfds;
- int fd, fd_max, n, i, ret, tcp_fd;
+ int fd, fd_rtcp, fd_max, n, i, ret, tcp_fd, timeout_cnt = 0;
struct timeval tv;
for (;;) {
if (url_interrupt_cb())
return AVERROR(EINTR);
+ if (wait_end && wait_end - av_gettime() < 0)
+ return AVERROR(EAGAIN);
FD_ZERO(&rfds);
if (rt->rtsp_hd) {
tcp_fd = fd_max = url_get_file_handle(rt->rtsp_hd);
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_rtcp = rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
+ if (FFMAX(fd, fd_rtcp) > fd_max)
+ fd_max = FFMAX(fd, fd_rtcp);
FD_SET(fd, &rfds);
+ FD_SET(fd_rtcp, &rfds);
}
}
tv.tv_sec = 0;
- tv.tv_usec = 100 * 1000;
+ 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)) {
+ fd_rtcp = rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
+ if (FD_ISSET(fd_rtcp, &rfds) || FD_ISSET(fd, &rfds)) {
ret = url_read(rtsp_st->rtp_handle, buf, buf_size);
if (ret > 0) {
*prtsp_st = rtsp_st;
if (tcp_fd != -1 && FD_ISSET(tcp_fd, &rfds)) {
RTSPMessageHeader reply;
- ret = ff_rtsp_read_reply(s, &reply, NULL, 0);
+ ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
if (ret < 0)
return ret;
/* XXX: parse message */
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)
-{
- RTSPState *rt = s->priv_data;
- int id, len, i, ret;
- RTSPStream *rtsp_st;
-
-#ifdef DEBUG_RTP_TCP
- dprintf(s, "tcp_read_packet:\n");
-#endif
-redo:
- for (;;) {
- RTSPMessageHeader reply;
-
- 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_STREAMING)
- return 0;
- }
- ret = url_read_complete(rt->rtsp_hd, buf, 3);
- if (ret != 3)
- return -1;
- id = buf[0];
- len = AV_RB16(buf + 1);
-#ifdef DEBUG_RTP_TCP
- dprintf(s, "id=%d len=%d\n", id, len);
-#endif
- if (len > buf_size || len < 12)
- goto redo;
- /* get the data */
- ret = url_read_complete(rt->rtsp_hd, buf, len);
- if (ret != len)
- return -1;
- if (rt->transport == RTSP_TRANSPORT_RDT &&
- ff_rdt_parse_header(buf, len, &id, NULL, NULL, NULL, NULL) < 0)
- return -1;
-
- /* find the matching stream */
- 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:
- *prtsp_st = rtsp_st;
- return len;
-}
-
-static int rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
+int ff_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;
+ RTSPStream *rtsp_st, *first_queue_st = NULL;
+ int64_t wait_end = 0;
+
+ if (rt->nb_byes == rt->nb_rtsp_streams)
+ return AVERROR_EOF;
/* get next frames from the same RTP packet */
if (rt->cur_transport_priv) {
rt->cur_transport_priv = NULL;
}
+ if (rt->transport == RTSP_TRANSPORT_RTP) {
+ int i;
+ int64_t first_queue_time = 0;
+ for (i = 0; i < rt->nb_rtsp_streams; i++) {
+ RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
+ int64_t queue_time;
+ if (!rtpctx)
+ continue;
+ queue_time = ff_rtp_queued_packet_time(rtpctx);
+ if (queue_time && (queue_time - first_queue_time < 0 ||
+ !first_queue_time)) {
+ first_queue_time = queue_time;
+ first_queue_st = rt->rtsp_streams[i];
+ }
+ }
+ if (first_queue_time)
+ wait_end = first_queue_time + s->max_delay;
+ }
+
/* read next RTP packet */
redo:
+ if (!rt->recvbuf) {
+ rt->recvbuf = av_malloc(RECVBUF_SIZE);
+ if (!rt->recvbuf)
+ return AVERROR(ENOMEM);
+ }
+
switch(rt->lower_transport) {
default:
#if CONFIG_RTSP_DEMUXER
case RTSP_LOWER_TRANSPORT_TCP:
- len = tcp_read_packet(s, &rtsp_st, buf, sizeof(buf));
+ len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
break;
#endif
case RTSP_LOWER_TRANSPORT_UDP:
case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
- len = udp_read_packet(s, &rtsp_st, buf, sizeof(buf));
+ len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
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 == AVERROR(EAGAIN) && first_queue_st &&
+ rt->transport == RTSP_TRANSPORT_RTP) {
+ rtsp_st = first_queue_st;
+ ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
+ goto end;
+ }
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;
-}
-
-static int rtsp_read_packet(AVFormatContext *s, AVPacket *pkt)
-{
- RTSPState *rt = s->priv_data;
- int ret;
- RTSPMessageHeader reply1, *reply = &reply1;
- char cmd[1024];
-
- if (rt->server_type == RTSP_SERVER_REAL) {
- int i;
- enum AVDiscard cache[MAX_STREAMS];
-
- for (i = 0; i < s->nb_streams; i++)
- cache[i] = s->streams[i]->discard;
-
- if (!rt->need_subscription) {
- if (memcmp (cache, rt->real_setup_cache,
- sizeof(enum AVDiscard) * s->nb_streams)) {
- snprintf(cmd, sizeof(cmd),
- "SET_PARAMETER %s RTSP/1.0\r\n"
- "Unsubscribe: %s\r\n",
- rt->control_uri, rt->last_subscription);
- ff_rtsp_send_cmd(s, cmd, reply, NULL);
- if (reply->status_code != RTSP_STATUS_OK)
- return AVERROR_INVALIDDATA;
- rt->need_subscription = 1;
- }
- }
-
- if (rt->need_subscription) {
- int r, rule_nr, first = 1;
-
- memcpy(rt->real_setup_cache, cache,
- sizeof(enum AVDiscard) * s->nb_streams);
- rt->last_subscription[0] = 0;
-
- snprintf(cmd, sizeof(cmd),
- "SET_PARAMETER %s RTSP/1.0\r\n"
- "Subscribe: ",
- rt->control_uri);
- for (i = 0; i < rt->nb_rtsp_streams; i++) {
- rule_nr = 0;
- for (r = 0; r < s->nb_streams; r++) {
- if (s->streams[r]->priv_data == rt->rtsp_streams[i]) {
- if (s->streams[r]->discard != AVDISCARD_ALL) {
- if (!first)
- av_strlcat(rt->last_subscription, ",",
- sizeof(rt->last_subscription));
- ff_rdt_subscribe_rule(
- rt->last_subscription,
- sizeof(rt->last_subscription), i, rule_nr);
- first = 0;
- }
- rule_nr++;
+ ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
+ } else {
+ ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, 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;
+ AVStream *st = NULL;
+ if (rtsp_st->stream_index >= 0)
+ st = s->streams[rtsp_st->stream_index];
+ for (i = 0; i < rt->nb_rtsp_streams; i++) {
+ RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
+ AVStream *st2 = NULL;
+ if (rt->rtsp_streams[i]->stream_index >= 0)
+ st2 = s->streams[rt->rtsp_streams[i]->stream_index];
+ if (rtpctx2 && st && st2 &&
+ rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
+ rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
+ rtpctx2->rtcp_ts_offset = av_rescale_q(
+ rtpctx->rtcp_ts_offset, st->time_base,
+ st2->time_base);
}
}
}
- av_strlcatf(cmd, sizeof(cmd), "%s\r\n", rt->last_subscription);
- ff_rtsp_send_cmd(s, cmd, reply, NULL);
- if (reply->status_code != RTSP_STATUS_OK)
- return AVERROR_INVALIDDATA;
- rt->need_subscription = 0;
-
- if (rt->state == RTSP_STATE_STREAMING)
- rtsp_read_play (s);
- }
- }
-
- ret = rtsp_fetch_packet(s, pkt);
- 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",
- rt->control_uri);
- ff_rtsp_send_cmd_async(s, cmd);
- } else {
- ff_rtsp_send_cmd_async(s, "OPTIONS * RTSP/1.0\r\n");
- }
- }
-
- return 0;
-}
-
-/* pause the stream */
-static int rtsp_read_pause(AVFormatContext *s)
-{
- RTSPState *rt = s->priv_data;
- RTSPMessageHeader reply1, *reply = &reply1;
- char cmd[1024];
+ if (ret == -RTCP_BYE) {
+ rt->nb_byes++;
- rt = s->priv_data;
+ av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
+ rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
- 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",
- rt->control_uri);
- ff_rtsp_send_cmd(s, cmd, reply, NULL);
- if (reply->status_code != RTSP_STATUS_OK) {
- return -1;
+ if (rt->nb_byes == rt->nb_rtsp_streams)
+ return AVERROR_EOF;
+ }
}
}
- 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;
-
- 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_STREAMING:
- if (rtsp_read_pause(s) != 0)
- return -1;
- rt->state = RTSP_STATE_SEEKING;
- if (rtsp_read_play(s) != 0)
- return -1;
- break;
- case RTSP_STATE_PAUSED:
- rt->state = RTSP_STATE_IDLE;
- break;
- }
- return 0;
-}
-
-static int rtsp_read_close(AVFormatContext *s)
-{
- RTSPState *rt = s->priv_data;
- char cmd[1024];
-
-#if 0
- /* NOTE: it is valid to flush the buffer here */
- if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
- url_fclose(&rt->rtsp_gb);
- }
-#endif
- snprintf(cmd, sizeof(cmd),
- "TEARDOWN %s RTSP/1.0\r\n",
- rt->control_uri);
- ff_rtsp_send_cmd_async(s, cmd);
+end:
+ 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;
- ff_rtsp_close_streams(s);
- url_close(rt->rtsp_hd);
- ff_network_close();
- return 0;
+ return ret;
}
+#endif /* CONFIG_RTPDEC */
-AVInputFormat rtsp_demuxer = {
- "rtsp",
- NULL_IF_CONFIG_SMALL("RTSP input format"),
- sizeof(RTSPState),
- rtsp_probe,
- 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
-
+#if CONFIG_SDP_DEMUXER
static int sdp_probe(AVProbeData *p1)
{
const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
- /* we look for a line beginning "c=IN IP4" */
+ /* we look for a line beginning "c=IN IP" */
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 IP") - 1 < p_end &&
+ av_strstart(p, "c=IN IP", NULL))
return AVPROBE_SCORE_MAX / 2;
while (p < p_end - 1 && *p != '\n') p++;
return 0;
}
-#define SDP_MAX_SIZE 8192
-
static int sdp_read_header(AVFormatContext *s, AVFormatParameters *ap)
{
RTSPState *rt = s->priv_data;
}
content[size] ='\0';
- sdp_parse(s, content);
+ ff_sdp_parse(s, content);
av_free(content);
/* open each RTP stream */
for (i = 0; i < rt->nb_rtsp_streams; i++) {
+ char namebuf[50];
rtsp_st = rt->rtsp_streams[i];
+ getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
+ namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
ff_url_join(url, sizeof(url), "rtp", NULL,
- inet_ntoa(rtsp_st->sdp_ip), rtsp_st->sdp_port,
+ namebuf, 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) {
sizeof(RTSPState),
sdp_probe,
sdp_read_header,
- rtsp_fetch_packet,
+ ff_rtsp_fetch_packet,
sdp_read_close,
};
+#endif /* CONFIG_SDP_DEMUXER */
+
+#if CONFIG_RTP_DEMUXER
+static int rtp_probe(AVProbeData *p)
+{
+ if (av_strstart(p->filename, "rtp:", NULL))
+ return AVPROBE_SCORE_MAX;
+ return 0;
+}
+
+static int rtp_read_header(AVFormatContext *s,
+ AVFormatParameters *ap)
+{
+ uint8_t recvbuf[1500];
+ char host[500], sdp[500];
+ int ret, port;
+ URLContext* in = NULL;
+ int payload_type;
+ AVCodecContext codec;
+ struct sockaddr_storage addr;
+ ByteIOContext pb;
+ socklen_t addrlen = sizeof(addr);
+
+ if (!ff_network_init())
+ return AVERROR(EIO);
+
+ ret = url_open(&in, s->filename, URL_RDONLY);
+ if (ret)
+ goto fail;
+
+ while (1) {
+ ret = url_read(in, recvbuf, sizeof(recvbuf));
+ if (ret == AVERROR(EAGAIN))
+ continue;
+ if (ret < 0)
+ goto fail;
+ if (ret < 12) {
+ av_log(s, AV_LOG_WARNING, "Received too short packet\n");
+ continue;
+ }
+
+ if ((recvbuf[0] & 0xc0) != 0x80) {
+ av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
+ "received\n");
+ continue;
+ }
+
+ payload_type = recvbuf[1] & 0x7f;
+ break;
+ }
+ getsockname(url_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
+ url_close(in);
+ in = NULL;
+
+ memset(&codec, 0, sizeof(codec));
+ if (ff_rtp_get_codec_info(&codec, payload_type)) {
+ av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
+ "without an SDP file describing it\n",
+ payload_type);
+ goto fail;
+ }
+ if (codec.codec_type != AVMEDIA_TYPE_DATA) {
+ av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
+ "properly you need an SDP file "
+ "describing it\n");
+ }
+
+ av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
+ NULL, 0, s->filename);
+
+ snprintf(sdp, sizeof(sdp),
+ "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
+ addr.ss_family == AF_INET ? 4 : 6, host,
+ codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
+ codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
+ port, payload_type);
+ av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
+
+ init_put_byte(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
+ s->pb = &pb;
+
+ /* sdp_read_header initializes this again */
+ ff_network_close();
+
+ ret = sdp_read_header(s, ap);
+ s->pb = NULL;
+ return ret;
+
+fail:
+ if (in)
+ url_close(in);
+ ff_network_close();
+ return ret;
+}
+
+AVInputFormat rtp_demuxer = {
+ "rtp",
+ NULL_IF_CONFIG_SMALL("RTP input format"),
+ sizeof(RTSPState),
+ rtp_probe,
+ rtp_read_header,
+ ff_rtsp_fetch_packet,
+ sdp_read_close,
+ .flags = AVFMT_NOFILE,
+};
+#endif /* CONFIG_RTP_DEMUXER */
+
projects / ffmpeg / blobdiff