* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#define HAVE_AV_CONFIG_H
-#include "common.h"
#include "avformat.h"
#include <stdarg.h>
#include <netinet/in.h>
#include <arpa/inet.h>
#include <netdb.h>
-#include <ctype.h>
#include <signal.h>
#ifdef CONFIG_HAVE_DLFCN
#include <dlfcn.h>
HTTPSTATE_SEND_DATA_TRAILER,
HTTPSTATE_RECEIVE_DATA,
HTTPSTATE_WAIT_FEED, /* wait for data from the feed */
- HTTPSTATE_WAIT, /* wait before sending next packets */
- HTTPSTATE_WAIT_SHORT, /* short wait for short term
- bandwidth limitation */
HTTPSTATE_READY,
RTSPSTATE_WAIT_REQUEST,
RTSPSTATE_SEND_REPLY,
+ RTSPSTATE_SEND_PACKET,
};
const char *http_state[] = {
"SEND_DATA_TRAILER",
"RECEIVE_DATA",
"WAIT_FEED",
- "WAIT",
- "WAIT_SHORT",
"READY",
"RTSP_WAIT_REQUEST",
"RTSP_SEND_REPLY",
+ "RTSP_SEND_PACKET",
};
#define IOBUFFER_INIT_SIZE 8192
AVFormatContext *fmt_in;
long start_time; /* In milliseconds - this wraps fairly often */
int64_t first_pts; /* initial pts value */
- int pts_stream_index; /* stream we choose as clock reference */
+ int64_t cur_pts; /* current pts value from the stream in us */
+ int64_t cur_frame_duration; /* duration of the current frame in us */
+ int cur_frame_bytes; /* output frame size, needed to compute
+ the time at which we send each
+ packet */
+ int pts_stream_index; /* stream we choose as clock reference */
+ int64_t cur_clock; /* current clock reference value in us */
/* output format handling */
struct FFStream *stream;
/* -1 is invalid stream */
uint8_t *pb_buffer; /* XXX: use that in all the code */
ByteIOContext *pb;
int seq; /* RTSP sequence number */
-
+
/* RTP state specific */
enum RTSPProtocol rtp_protocol;
char session_id[32]; /* session id */
AVFormatContext *rtp_ctx[MAX_STREAMS];
+
+ /* RTP/UDP specific */
URLContext *rtp_handles[MAX_STREAMS];
- /* RTP short term bandwidth limitation */
- int packet_byte_count;
- int packet_start_time_us; /* used for short durations (a few
- seconds max) */
+
+ /* RTP/TCP specific */
+ struct HTTPContext *rtsp_c;
+ uint8_t *packet_buffer, *packet_buffer_ptr, *packet_buffer_end;
} HTTPContext;
static AVFrame dummy_frame;
char filename[1024]; /* stream filename */
struct FFStream *feed; /* feed we are using (can be null if
coming from file) */
+ AVFormatParameters *ap_in; /* input parameters */
+ AVInputFormat *ifmt; /* if non NULL, force input format */
AVOutputFormat *fmt;
IPAddressACL *acl;
int nb_streams;
static int open_input_stream(HTTPContext *c, const char *info);
static int http_start_receive_data(HTTPContext *c);
static int http_receive_data(HTTPContext *c);
-static int compute_send_delay(HTTPContext *c);
/* RTSP handling */
static int rtsp_parse_request(HTTPContext *c);
/* RTP handling */
static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr,
- FFStream *stream, const char *session_id);
+ FFStream *stream, const char *session_id,
+ enum RTSPProtocol rtp_protocol);
static int rtp_new_av_stream(HTTPContext *c,
- int stream_index, struct sockaddr_in *dest_addr);
+ int stream_index, struct sockaddr_in *dest_addr,
+ HTTPContext *rtsp_c);
static const char *my_program_name;
static const char *my_program_dir;
static FILE *logfile = NULL;
-static void http_log(const char *fmt, ...)
+static void __attribute__ ((format (printf, 1, 2))) http_log(const char *fmt, ...)
{
va_list ap;
va_start(ap, fmt);
dest_addr.sin_addr = stream->multicast_ip;
dest_addr.sin_port = htons(stream->multicast_port);
- rtp_c = rtp_new_connection(&dest_addr, stream, session_id);
+ rtp_c = rtp_new_connection(&dest_addr, stream, session_id,
+ RTSP_PROTOCOL_RTP_UDP_MULTICAST);
if (!rtp_c) {
continue;
}
continue;
}
- rtp_c->rtp_protocol = RTSP_PROTOCOL_RTP_UDP_MULTICAST;
-
/* open each RTP stream */
for(stream_index = 0; stream_index < stream->nb_streams;
stream_index++) {
dest_addr.sin_port = htons(stream->multicast_port +
2 * stream_index);
- if (rtp_new_av_stream(rtp_c, stream_index, &dest_addr) < 0) {
+ if (rtp_new_av_stream(rtp_c, stream_index, &dest_addr, NULL) < 0) {
fprintf(stderr, "Could not open output stream '%s/streamid=%d'\n",
stream->filename, stream_index);
exit(1);
first_http_ctx = NULL;
nb_connections = 0;
- first_http_ctx = NULL;
start_multicast();
switch(c->state) {
case HTTPSTATE_SEND_HEADER:
case RTSPSTATE_SEND_REPLY:
+ case RTSPSTATE_SEND_PACKET:
c->poll_entry = poll_entry;
poll_entry->fd = fd;
poll_entry->events = POLLOUT;
poll_entry->events = POLLOUT;
poll_entry++;
} else {
- /* not strictly correct, but currently cannot add
- more than one fd in poll entry */
- delay = 0;
+ /* when ffserver is doing the timing, we work by
+ looking at which packet need to be sent every
+ 10 ms */
+ delay1 = 10; /* one tick wait XXX: 10 ms assumed */
+ if (delay1 < delay)
+ delay = delay1;
}
break;
case HTTPSTATE_WAIT_REQUEST:
poll_entry->events = POLLIN;/* Maybe this will work */
poll_entry++;
break;
- case HTTPSTATE_WAIT:
- c->poll_entry = NULL;
- delay1 = compute_send_delay(c);
- if (delay1 < delay)
- delay = delay1;
- break;
- case HTTPSTATE_WAIT_SHORT:
- c->poll_entry = NULL;
- delay1 = 10; /* one tick wait XXX: 10 ms assumed */
- if (delay1 < delay)
- delay = delay1;
- break;
default:
c->poll_entry = NULL;
break;
if (!c)
goto fail;
- c->next = first_http_ctx;
- first_http_ctx = c;
c->fd = fd;
c->poll_entry = NULL;
c->from_addr = from_addr;
c->buffer = av_malloc(c->buffer_size);
if (!c->buffer)
goto fail;
+
+ c->next = first_http_ctx;
+ first_http_ctx = c;
nb_connections++;
start_wait_request(c, is_rtsp);
}
}
+ /* remove references, if any (XXX: do it faster) */
+ for(c1 = first_http_ctx; c1 != NULL; c1 = c1->next) {
+ if (c1->rtsp_c == c)
+ c1->rtsp_c = NULL;
+ }
+
/* remove connection associated resources */
if (c->fd >= 0)
close(c->fd);
url_close(h);
}
}
-
+
ctx = &c->fmt_ctx;
- for(i=0; i<ctx->nb_streams; i++)
- av_free(ctx->streams[i]) ;
-
if (!c->last_packet_sent) {
if (ctx->oformat) {
/* prepare header */
if (url_open_dyn_buf(&ctx->pb) >= 0) {
av_write_trailer(ctx);
- (void) url_close_dyn_buf(&ctx->pb, &c->pb_buffer);
+ url_close_dyn_buf(&ctx->pb, &c->pb_buffer);
}
}
}
+ for(i=0; i<ctx->nb_streams; i++)
+ av_free(ctx->streams[i]) ;
+
if (c->stream)
current_bandwidth -= c->stream->bandwidth;
av_freep(&c->pb_buffer);
+ av_freep(&c->packet_buffer);
av_free(c->buffer);
av_free(c);
nb_connections--;
if (!(c->poll_entry->revents & POLLIN))
return 0;
/* read the data */
- len = read(c->fd, c->buffer_ptr, c->buffer_end - c->buffer_ptr);
+ read_loop:
+ len = read(c->fd, c->buffer_ptr, 1);
if (len < 0) {
if (errno != EAGAIN && errno != EINTR)
return -1;
} else if (len == 0) {
return -1;
} else {
- /* search for end of request. XXX: not fully correct since garbage could come after the end */
+ /* search for end of request. */
uint8_t *ptr;
c->buffer_ptr += len;
ptr = c->buffer_ptr;
} else if (ptr >= c->buffer_end) {
/* request too long: cannot do anything */
return -1;
- }
+ } else goto read_loop;
}
break;
/* nothing to do, we'll be waken up by incoming feed packets */
break;
- case HTTPSTATE_WAIT:
- /* if the delay expired, we can send new packets */
- if (compute_send_delay(c) <= 0)
- c->state = HTTPSTATE_SEND_DATA;
- break;
- case HTTPSTATE_WAIT_SHORT:
- /* just return back to send data */
- c->state = HTTPSTATE_SEND_DATA;
- break;
-
case RTSPSTATE_SEND_REPLY:
if (c->poll_entry->revents & (POLLERR | POLLHUP)) {
av_freep(&c->pb_buffer);
}
}
break;
+ case RTSPSTATE_SEND_PACKET:
+ if (c->poll_entry->revents & (POLLERR | POLLHUP)) {
+ av_freep(&c->packet_buffer);
+ return -1;
+ }
+ /* no need to write if no events */
+ if (!(c->poll_entry->revents & POLLOUT))
+ return 0;
+ len = write(c->fd, c->packet_buffer_ptr,
+ c->packet_buffer_end - c->packet_buffer_ptr);
+ if (len < 0) {
+ if (errno != EAGAIN && errno != EINTR) {
+ /* error : close connection */
+ av_freep(&c->packet_buffer);
+ return -1;
+ }
+ } else {
+ c->packet_buffer_ptr += len;
+ if (c->packet_buffer_ptr >= c->packet_buffer_end) {
+ /* all the buffer was sent : wait for a new request */
+ av_freep(&c->packet_buffer);
+ c->state = RTSPSTATE_WAIT_REQUEST;
+ }
+ }
+ break;
case HTTPSTATE_READY:
/* nothing to do */
break;
video_codec_name = codec->name;
}
break;
+ case CODEC_TYPE_DATA:
+ video_bit_rate += st->codec.bit_rate;
+ break;
default:
av_abort();
}
#endif
/* open stream */
- if (av_open_input_file(&s, input_filename, NULL, buf_size, NULL) < 0) {
+ if (av_open_input_file(&s, input_filename, c->stream->ifmt,
+ buf_size, c->stream->ap_in) < 0) {
http_log("%s not found", input_filename);
return -1;
}
}
}
+#if 0
if (c->fmt_in->iformat->read_seek) {
c->fmt_in->iformat->read_seek(c->fmt_in, stream_pos);
}
+#endif
/* set the start time (needed for maxtime and RTP packet timing) */
c->start_time = cur_time;
c->first_pts = AV_NOPTS_VALUE;
return 0;
}
-/* currently desactivated because the new PTS handling is not
- satisfactory yet */
-//#define AV_READ_FRAME
-#ifdef AV_READ_FRAME
-
-/* XXX: generalize that in ffmpeg for picture/audio/data. Currently
- the return packet MUST NOT be freed */
-int av_read_frame(AVFormatContext *s, AVPacket *pkt)
+/* return the server clock (in us) */
+static int64_t get_server_clock(HTTPContext *c)
{
- AVStream *st;
- int len, ret, old_nb_streams, i;
-
- /* see if remaining frames must be parsed */
- for(;;) {
- if (s->cur_len > 0) {
- st = s->streams[s->cur_pkt.stream_index];
- len = avcodec_parse_frame(&st->codec, &pkt->data, &pkt->size,
- s->cur_ptr, s->cur_len);
- if (len < 0) {
- /* error: get next packet */
- s->cur_len = 0;
- } else {
- s->cur_ptr += len;
- s->cur_len -= len;
- if (pkt->size) {
- /* init pts counter if not done */
- if (st->pts.den == 0) {
- switch(st->codec.codec_type) {
- case CODEC_TYPE_AUDIO:
- st->pts_incr = (int64_t)s->pts_den;
- av_frac_init(&st->pts, st->pts.val, 0,
- (int64_t)s->pts_num * st->codec.sample_rate);
- break;
- case CODEC_TYPE_VIDEO:
- st->pts_incr = (int64_t)s->pts_den * st->codec.frame_rate_base;
- av_frac_init(&st->pts, st->pts.val, 0,
- (int64_t)s->pts_num * st->codec.frame_rate);
- break;
- default:
- av_abort();
- }
- }
-
- /* a frame was read: return it */
- pkt->pts = st->pts.val;
-#if 0
- printf("add pts=%Lx num=%Lx den=%Lx incr=%Lx\n",
- st->pts.val, st->pts.num, st->pts.den, st->pts_incr);
-#endif
- switch(st->codec.codec_type) {
- case CODEC_TYPE_AUDIO:
- av_frac_add(&st->pts, st->pts_incr * st->codec.frame_size);
- break;
- case CODEC_TYPE_VIDEO:
- av_frac_add(&st->pts, st->pts_incr);
- break;
- default:
- av_abort();
- }
- pkt->stream_index = s->cur_pkt.stream_index;
- /* we use the codec indication because it is
- more accurate than the demux flags */
- pkt->flags = 0;
- if (st->codec.coded_frame->key_frame)
- pkt->flags |= PKT_FLAG_KEY;
- return 0;
- }
- }
- } else {
- /* free previous packet */
- av_free_packet(&s->cur_pkt);
-
- old_nb_streams = s->nb_streams;
- ret = av_read_packet(s, &s->cur_pkt);
- if (ret)
- return ret;
- /* open parsers for each new streams */
- for(i = old_nb_streams; i < s->nb_streams; i++)
- open_parser(s, i);
- st = s->streams[s->cur_pkt.stream_index];
-
- /* update current pts (XXX: dts handling) from packet, or
- use current pts if none given */
- if (s->cur_pkt.pts != AV_NOPTS_VALUE) {
- av_frac_set(&st->pts, s->cur_pkt.pts);
- } else {
- s->cur_pkt.pts = st->pts.val;
- }
- if (!st->codec.codec) {
- /* no codec opened: just return the raw packet */
- *pkt = s->cur_pkt;
-
- /* no codec opened: just update the pts by considering we
- have one frame and free the packet */
- if (st->pts.den == 0) {
- switch(st->codec.codec_type) {
- case CODEC_TYPE_AUDIO:
- st->pts_incr = (int64_t)s->pts_den * st->codec.frame_size;
- av_frac_init(&st->pts, st->pts.val, 0,
- (int64_t)s->pts_num * st->codec.sample_rate);
- break;
- case CODEC_TYPE_VIDEO:
- st->pts_incr = (int64_t)s->pts_den * st->codec.frame_rate_base;
- av_frac_init(&st->pts, st->pts.val, 0,
- (int64_t)s->pts_num * st->codec.frame_rate);
- break;
- default:
- av_abort();
- }
- }
- av_frac_add(&st->pts, st->pts_incr);
- return 0;
- } else {
- s->cur_ptr = s->cur_pkt.data;
- s->cur_len = s->cur_pkt.size;
- }
- }
- }
+ /* compute current pts value from system time */
+ return (int64_t)(cur_time - c->start_time) * 1000LL;
}
-static int compute_send_delay(HTTPContext *c)
+/* return the estimated time at which the current packet must be sent
+ (in us) */
+static int64_t get_packet_send_clock(HTTPContext *c)
{
- int64_t cur_pts, delta_pts, next_pts;
- int delay1;
+ int bytes_left, bytes_sent, frame_bytes;
- /* compute current pts value from system time */
- cur_pts = ((int64_t)(cur_time - c->start_time) * c->fmt_in->pts_den) /
- (c->fmt_in->pts_num * 1000LL);
- /* compute the delta from the stream we choose as
- main clock (we do that to avoid using explicit
- buffers to do exact packet reordering for each
- stream */
- /* XXX: really need to fix the number of streams */
- if (c->pts_stream_index >= c->fmt_in->nb_streams)
- next_pts = cur_pts;
- else
- next_pts = c->fmt_in->streams[c->pts_stream_index]->pts.val;
- delta_pts = next_pts - cur_pts;
- if (delta_pts <= 0) {
- delay1 = 0;
+ frame_bytes = c->cur_frame_bytes;
+ if (frame_bytes <= 0) {
+ return c->cur_pts;
} else {
- delay1 = (delta_pts * 1000 * c->fmt_in->pts_num) / c->fmt_in->pts_den;
+ bytes_left = c->buffer_end - c->buffer_ptr;
+ bytes_sent = frame_bytes - bytes_left;
+ return c->cur_pts + (c->cur_frame_duration * bytes_sent) / frame_bytes;
}
- return delay1;
}
-#else
-
-/* just fall backs */
-static int av_read_frame(AVFormatContext *s, AVPacket *pkt)
-{
- return av_read_packet(s, pkt);
-}
-
-static int compute_send_delay(HTTPContext *c)
-{
- int datarate = 8 * get_longterm_datarate(&c->datarate, c->data_count);
- if (datarate > c->stream->bandwidth * 2000) {
- return 1000;
- }
- return 0;
-}
-#endif
-
static int http_prepare_data(HTTPContext *c)
{
int i, len, ret;
AVFormatContext *ctx;
+ av_freep(&c->pb_buffer);
switch(c->state) {
case HTTPSTATE_SEND_DATA_HEADER:
memset(&c->fmt_ctx, 0, sizeof(c->fmt_ctx));
/* We have timed out */
c->state = HTTPSTATE_SEND_DATA_TRAILER;
} else {
- if (1 || c->is_packetized) {
- if (compute_send_delay(c) > 0) {
- c->state = HTTPSTATE_WAIT;
- return 1; /* state changed */
- }
- }
redo:
if (av_read_frame(c->fmt_in, &pkt) < 0) {
if (c->stream->feed && c->stream->feed->feed_opened) {
}
} else {
/* update first pts if needed */
- if (c->first_pts == AV_NOPTS_VALUE)
- c->first_pts = pkt.pts;
-
+ if (c->first_pts == AV_NOPTS_VALUE) {
+ c->first_pts = pkt.dts;
+ c->start_time = cur_time;
+ }
/* send it to the appropriate stream */
if (c->stream->feed) {
/* if coming from a feed, select the right stream */
output stream (one for each RTP
connection). XXX: need more abstract handling */
if (c->is_packetized) {
+ AVStream *st;
+ /* compute send time and duration */
+ st = c->fmt_in->streams[pkt.stream_index];
+ c->cur_pts = pkt.dts;
+ if (st->start_time != AV_NOPTS_VALUE)
+ c->cur_pts -= st->start_time;
+ c->cur_frame_duration = pkt.duration;
+#if 0
+ printf("index=%d pts=%0.3f duration=%0.6f\n",
+ pkt.stream_index,
+ (double)c->cur_pts /
+ AV_TIME_BASE,
+ (double)c->cur_frame_duration /
+ AV_TIME_BASE);
+#endif
+ /* find RTP context */
c->packet_stream_index = pkt.stream_index;
ctx = c->rtp_ctx[c->packet_stream_index];
+ if(!ctx) {
+ av_free_packet(&pkt);
+ break;
+ }
codec = &ctx->streams[0]->codec;
/* only one stream per RTP connection */
pkt.stream_index = 0;
}
codec->coded_frame->key_frame = ((pkt.flags & PKT_FLAG_KEY) != 0);
-
-#ifdef PJSG
- if (codec->codec_type == CODEC_TYPE_AUDIO) {
- codec->frame_size = (codec->sample_rate * pkt.duration + 500000) / 1000000;
- /* printf("Calculated size %d, from sr %d, duration %d\n", codec->frame_size, codec->sample_rate, pkt.duration); */
- }
-#endif
-
if (c->is_packetized) {
- ret = url_open_dyn_packet_buf(&ctx->pb,
- url_get_max_packet_size(c->rtp_handles[c->packet_stream_index]));
- c->packet_byte_count = 0;
- c->packet_start_time_us = av_gettime();
+ int max_packet_size;
+ if (c->rtp_protocol == RTSP_PROTOCOL_RTP_TCP)
+ max_packet_size = RTSP_TCP_MAX_PACKET_SIZE;
+ else
+ max_packet_size = url_get_max_packet_size(c->rtp_handles[c->packet_stream_index]);
+ ret = url_open_dyn_packet_buf(&ctx->pb, max_packet_size);
} else {
ret = url_open_dyn_buf(&ctx->pb);
}
}
len = url_close_dyn_buf(&ctx->pb, &c->pb_buffer);
+ c->cur_frame_bytes = len;
c->buffer_ptr = c->pb_buffer;
c->buffer_end = c->pb_buffer + len;
codec->frame_number++;
+ if (len == 0)
+ goto redo;
}
-#ifndef AV_READ_FRAME
av_free_packet(&pkt);
-#endif
}
}
}
#define SHORT_TERM_BANDWIDTH 8000000
/* should convert the format at the same time */
+/* send data starting at c->buffer_ptr to the output connection
+ (either UDP or TCP connection) */
static int http_send_data(HTTPContext *c)
{
- int len, ret, dt;
-
- while (c->buffer_ptr >= c->buffer_end) {
- av_freep(&c->pb_buffer);
- ret = http_prepare_data(c);
- if (ret < 0)
- return -1;
- else if (ret == 0) {
- continue;
- } else {
- /* state change requested */
- return 0;
- }
- }
+ int len, ret;
- if (c->buffer_ptr < c->buffer_end) {
- if (c->is_packetized) {
- /* RTP/UDP data output */
- len = c->buffer_end - c->buffer_ptr;
- if (len < 4) {
- /* fail safe - should never happen */
- fail1:
- c->buffer_ptr = c->buffer_end;
- return 0;
- }
- len = (c->buffer_ptr[0] << 24) |
- (c->buffer_ptr[1] << 16) |
- (c->buffer_ptr[2] << 8) |
- (c->buffer_ptr[3]);
- if (len > (c->buffer_end - c->buffer_ptr))
- goto fail1;
-
- /* short term bandwidth limitation */
- dt = av_gettime() - c->packet_start_time_us;
- if (dt < 1)
- dt = 1;
-
- if ((c->packet_byte_count + len) * (int64_t)1000000 >=
- (SHORT_TERM_BANDWIDTH / 8) * (int64_t)dt) {
- /* bandwidth overflow : wait at most one tick and retry */
- c->state = HTTPSTATE_WAIT_SHORT;
- return 0;
+ for(;;) {
+ if (c->buffer_ptr >= c->buffer_end) {
+ ret = http_prepare_data(c);
+ if (ret < 0)
+ return -1;
+ else if (ret != 0) {
+ /* state change requested */
+ break;
}
-
- c->buffer_ptr += 4;
- url_write(c->rtp_handles[c->packet_stream_index],
- c->buffer_ptr, len);
- c->buffer_ptr += len;
- c->packet_byte_count += len;
} else {
- /* TCP data output */
- len = write(c->fd, c->buffer_ptr, c->buffer_end - c->buffer_ptr);
- if (len < 0) {
- if (errno != EAGAIN && errno != EINTR) {
- /* error : close connection */
- return -1;
- } else {
+ if (c->is_packetized) {
+ /* RTP data output */
+ len = c->buffer_end - c->buffer_ptr;
+ if (len < 4) {
+ /* fail safe - should never happen */
+ fail1:
+ c->buffer_ptr = c->buffer_end;
return 0;
}
+ len = (c->buffer_ptr[0] << 24) |
+ (c->buffer_ptr[1] << 16) |
+ (c->buffer_ptr[2] << 8) |
+ (c->buffer_ptr[3]);
+ if (len > (c->buffer_end - c->buffer_ptr))
+ goto fail1;
+ if ((get_packet_send_clock(c) - get_server_clock(c)) > 0) {
+ /* nothing to send yet: we can wait */
+ return 0;
+ }
+
+ c->data_count += len;
+ update_datarate(&c->datarate, c->data_count);
+ if (c->stream)
+ c->stream->bytes_served += len;
+
+ if (c->rtp_protocol == RTSP_PROTOCOL_RTP_TCP) {
+ /* RTP packets are sent inside the RTSP TCP connection */
+ ByteIOContext pb1, *pb = &pb1;
+ int interleaved_index, size;
+ uint8_t header[4];
+ HTTPContext *rtsp_c;
+
+ rtsp_c = c->rtsp_c;
+ /* if no RTSP connection left, error */
+ if (!rtsp_c)
+ return -1;
+ /* if already sending something, then wait. */
+ if (rtsp_c->state != RTSPSTATE_WAIT_REQUEST) {
+ break;
+ }
+ if (url_open_dyn_buf(pb) < 0)
+ goto fail1;
+ interleaved_index = c->packet_stream_index * 2;
+ /* RTCP packets are sent at odd indexes */
+ if (c->buffer_ptr[1] == 200)
+ interleaved_index++;
+ /* write RTSP TCP header */
+ header[0] = '$';
+ header[1] = interleaved_index;
+ header[2] = len >> 8;
+ header[3] = len;
+ put_buffer(pb, header, 4);
+ /* write RTP packet data */
+ c->buffer_ptr += 4;
+ put_buffer(pb, c->buffer_ptr, len);
+ size = url_close_dyn_buf(pb, &c->packet_buffer);
+ /* prepare asynchronous TCP sending */
+ rtsp_c->packet_buffer_ptr = c->packet_buffer;
+ rtsp_c->packet_buffer_end = c->packet_buffer + size;
+ c->buffer_ptr += len;
+
+ /* send everything we can NOW */
+ len = write(rtsp_c->fd, rtsp_c->packet_buffer_ptr,
+ rtsp_c->packet_buffer_end - rtsp_c->packet_buffer_ptr);
+ if (len > 0) {
+ rtsp_c->packet_buffer_ptr += len;
+ }
+ if (rtsp_c->packet_buffer_ptr < rtsp_c->packet_buffer_end) {
+ /* if we could not send all the data, we will
+ send it later, so a new state is needed to
+ "lock" the RTSP TCP connection */
+ rtsp_c->state = RTSPSTATE_SEND_PACKET;
+ break;
+ } else {
+ /* all data has been sent */
+ av_freep(&c->packet_buffer);
+ }
+ } else {
+ /* send RTP packet directly in UDP */
+ c->buffer_ptr += 4;
+ url_write(c->rtp_handles[c->packet_stream_index],
+ c->buffer_ptr, len);
+ c->buffer_ptr += len;
+ /* here we continue as we can send several packets per 10 ms slot */
+ }
} else {
- c->buffer_ptr += len;
+ /* TCP data output */
+ len = write(c->fd, c->buffer_ptr, c->buffer_end - c->buffer_ptr);
+ if (len < 0) {
+ if (errno != EAGAIN && errno != EINTR) {
+ /* error : close connection */
+ return -1;
+ } else {
+ return 0;
+ }
+ } else {
+ c->buffer_ptr += len;
+ }
+ c->data_count += len;
+ update_datarate(&c->datarate, c->data_count);
+ if (c->stream)
+ c->stream->bytes_served += len;
+ break;
}
}
- c->data_count += len;
- update_datarate(&c->datarate, c->data_count);
- if (c->stream)
- c->stream->bytes_served += len;
- }
+ } /* for(;;) */
return 0;
}
url_fprintf(pb, "c=IN IP4 %s\n", inet_ntoa(stream->multicast_ip));
}
/* for each stream, we output the necessary info */
- private_payload_type = 96;
+ private_payload_type = RTP_PT_PRIVATE;
for(i = 0; i < stream->nb_streams; i++) {
st = stream->streams[i];
- switch(st->codec.codec_type) {
- case CODEC_TYPE_AUDIO:
- mediatype = "audio";
- break;
- case CODEC_TYPE_VIDEO:
+ if (st->codec.codec_id == CODEC_ID_MPEG2TS) {
mediatype = "video";
- break;
- default:
- mediatype = "application";
- break;
+ } else {
+ switch(st->codec.codec_type) {
+ case CODEC_TYPE_AUDIO:
+ mediatype = "audio";
+ break;
+ case CODEC_TYPE_VIDEO:
+ mediatype = "video";
+ break;
+ default:
+ mediatype = "application";
+ break;
+ }
}
/* NOTE: the port indication is not correct in case of
unicast. It is not an issue because RTSP gives it */
}
url_fprintf(pb, "m=%s %d RTP/AVP %d\n",
mediatype, port, payload_type);
- if (payload_type >= 96) {
+ if (payload_type >= RTP_PT_PRIVATE) {
/* for private payload type, we need to give more info */
switch(st->codec.codec_id) {
case CODEC_ID_MPEG4:
/* we must also add the mpeg4 header */
data = st->codec.extradata;
if (data) {
- url_fprintf(pb, "a=fmtp:%d config=");
+ url_fprintf(pb, "a=fmtp:%d config=", payload_type);
for(j=0;j<st->codec.extradata_size;j++) {
url_fprintf(pb, "%02x", data[j]);
}
/* get the host IP */
len = sizeof(my_addr);
getsockname(c->fd, (struct sockaddr *)&my_addr, &len);
-
content_length = prepare_sdp_description(stream, &content, my_addr.sin_addr);
if (content_length < 0) {
rtsp_reply_error(c, RTSP_STATUS_INTERNAL);
/* find rtp session, and create it if none found */
rtp_c = find_rtp_session(h->session_id);
if (!rtp_c) {
- rtp_c = rtp_new_connection(&c->from_addr, stream, h->session_id);
+ /* always prefer UDP */
+ th = find_transport(h, RTSP_PROTOCOL_RTP_UDP);
+ if (!th) {
+ th = find_transport(h, RTSP_PROTOCOL_RTP_TCP);
+ if (!th) {
+ rtsp_reply_error(c, RTSP_STATUS_TRANSPORT);
+ return;
+ }
+ }
+
+ rtp_c = rtp_new_connection(&c->from_addr, stream, h->session_id,
+ th->protocol);
if (!rtp_c) {
rtsp_reply_error(c, RTSP_STATUS_BANDWIDTH);
return;
rtsp_reply_error(c, RTSP_STATUS_INTERNAL);
return;
}
-
- /* always prefer UDP */
- th = find_transport(h, RTSP_PROTOCOL_RTP_UDP);
- if (!th) {
- th = find_transport(h, RTSP_PROTOCOL_RTP_TCP);
- if (!th) {
- rtsp_reply_error(c, RTSP_STATUS_TRANSPORT);
- return;
- }
- }
- rtp_c->rtp_protocol = th->protocol;
}
/* test if stream is OK (test needed because several SETUP needs
}
/* setup stream */
- if (rtp_new_av_stream(rtp_c, stream_index, &dest_addr) < 0) {
+ if (rtp_new_av_stream(rtp_c, stream_index, &dest_addr, c) < 0) {
rtsp_reply_error(c, RTSP_STATUS_TRANSPORT);
return;
}
HTTPContext *rtp_c;
char path1[1024];
const char *path;
+ char buf[1024];
+ int s;
rtp_c = find_rtp_session(session_id);
if (!rtp_c)
path = path1;
if (*path == '/')
path++;
- if (strcmp(path, rtp_c->stream->filename) != 0)
- return NULL;
- return rtp_c;
+ if(!strcmp(path, rtp_c->stream->filename)) return rtp_c;
+ for(s=0; s<rtp_c->stream->nb_streams; ++s) {
+ snprintf(buf, sizeof(buf), "%s/streamid=%d",
+ rtp_c->stream->filename, s);
+ if(!strncmp(path, buf, sizeof(buf))) {
+ // XXX: Should we reply with RTSP_STATUS_ONLY_AGGREGATE if nb_streams>1?
+ return rtp_c;
+ }
+ }
+ return NULL;
}
static void rtsp_cmd_play(HTTPContext *c, const char *url, RTSPHeader *h)
return;
}
+#if 0
+ /* XXX: seek in stream */
+ if (h->range_start != AV_NOPTS_VALUE) {
+ printf("range_start=%0.3f\n", (double)h->range_start / AV_TIME_BASE);
+ av_seek_frame(rtp_c->fmt_in, -1, h->range_start);
+ }
+#endif
+
rtp_c->state = HTTPSTATE_SEND_DATA;
/* now everything is OK, so we can send the connection parameters */
}
rtp_c->state = HTTPSTATE_READY;
-
+ rtp_c->first_pts = AV_NOPTS_VALUE;
/* now everything is OK, so we can send the connection parameters */
rtsp_reply_header(c, RTSP_STATUS_OK);
/* session ID */
/* RTP handling */
static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr,
- FFStream *stream, const char *session_id)
+ FFStream *stream, const char *session_id,
+ enum RTSPProtocol rtp_protocol)
{
HTTPContext *c = NULL;
-
+ const char *proto_str;
+
/* XXX: should output a warning page when coming
close to the connection limit */
if (nb_connections >= nb_max_connections)
pstrcpy(c->session_id, sizeof(c->session_id), session_id);
c->state = HTTPSTATE_READY;
c->is_packetized = 1;
+ c->rtp_protocol = rtp_protocol;
+
/* protocol is shown in statistics */
- pstrcpy(c->protocol, sizeof(c->protocol), "RTP");
+ switch(c->rtp_protocol) {
+ case RTSP_PROTOCOL_RTP_UDP_MULTICAST:
+ proto_str = "MCAST";
+ break;
+ case RTSP_PROTOCOL_RTP_UDP:
+ proto_str = "UDP";
+ break;
+ case RTSP_PROTOCOL_RTP_TCP:
+ proto_str = "TCP";
+ break;
+ default:
+ proto_str = "???";
+ break;
+ }
+ pstrcpy(c->protocol, sizeof(c->protocol), "RTP/");
+ pstrcat(c->protocol, sizeof(c->protocol), proto_str);
current_bandwidth += stream->bandwidth;
}
/* add a new RTP stream in an RTP connection (used in RTSP SETUP
- command). if dest_addr is NULL, then TCP tunneling in RTSP is
+ command). If RTP/TCP protocol is used, TCP connection 'rtsp_c' is
used. */
static int rtp_new_av_stream(HTTPContext *c,
- int stream_index, struct sockaddr_in *dest_addr)
+ int stream_index, struct sockaddr_in *dest_addr,
+ HTTPContext *rtsp_c)
{
AVFormatContext *ctx;
AVStream *st;
URLContext *h;
uint8_t *dummy_buf;
char buf2[32];
+ int max_packet_size;
/* now we can open the relevant output stream */
ctx = av_mallocz(sizeof(AVFormatContext));
sizeof(AVStream));
}
- if (dest_addr) {
- /* build destination RTP address */
- ipaddr = inet_ntoa(dest_addr->sin_addr);
+ /* build destination RTP address */
+ ipaddr = inet_ntoa(dest_addr->sin_addr);
+
+ switch(c->rtp_protocol) {
+ case RTSP_PROTOCOL_RTP_UDP:
+ case RTSP_PROTOCOL_RTP_UDP_MULTICAST:
+ /* RTP/UDP case */
/* XXX: also pass as parameter to function ? */
if (c->stream->is_multicast) {
if (url_open(&h, ctx->filename, URL_WRONLY) < 0)
goto fail;
c->rtp_handles[stream_index] = h;
- } else {
+ max_packet_size = url_get_max_packet_size(h);
+ break;
+ case RTSP_PROTOCOL_RTP_TCP:
+ /* RTP/TCP case */
+ c->rtsp_c = rtsp_c;
+ max_packet_size = RTSP_TCP_MAX_PACKET_SIZE;
+ break;
+ default:
goto fail;
}
- http_log("%s:%d - - [%s] \"RTPSTART %s/streamid=%d\"\n",
+ http_log("%s:%d - - [%s] \"PLAY %s/streamid=%d %s\"\n",
ipaddr, ntohs(dest_addr->sin_port),
ctime1(buf2),
- c->stream->filename, stream_index);
+ c->stream->filename, stream_index, c->protocol);
/* normally, no packets should be output here, but the packet size may be checked */
- if (url_open_dyn_packet_buf(&ctx->pb,
- url_get_max_packet_size(h)) < 0) {
+ if (url_open_dyn_packet_buf(&ctx->pb, max_packet_size) < 0) {
/* XXX: close stream */
goto fail;
}
for(i=0;i<infile->nb_streams;i++) {
st = infile->streams[i];
if (st->codec.codec_id == CODEC_ID_MPEG4 &&
- st->codec.extradata == NULL) {
+ st->codec.extradata_size == 0) {
mpeg4_count++;
}
}
break;
st = infile->streams[pkt.stream_index];
if (st->codec.codec_id == CODEC_ID_MPEG4 &&
- st->codec.extradata == NULL) {
+ st->codec.extradata_size == 0) {
+ av_freep(&st->codec.extradata);
/* fill extradata with the header */
/* XXX: we make hard suppositions here ! */
p = pkt.data;
/* the stream comes from a file */
/* try to open the file */
/* open stream */
+ stream->ap_in = av_mallocz(sizeof(AVFormatParameters));
+ if (stream->fmt == &rtp_mux) {
+ /* specific case : if transport stream output to RTP,
+ we use a raw transport stream reader */
+ stream->ap_in->mpeg2ts_raw = 1;
+ stream->ap_in->mpeg2ts_compute_pcr = 1;
+ }
+
if (av_open_input_file(&infile, stream->feed_filename,
- NULL, 0, NULL) < 0) {
+ stream->ifmt, 0, stream->ap_in) < 0) {
http_log("%s not found", stream->feed_filename);
/* remove stream (no need to spend more time on it) */
fail:
if (sf->index != ss->index ||
sf->id != ss->id) {
- printf("Index & Id do not match for stream %d\n", i);
+ printf("Index & Id do not match for stream %d (%s)\n",
+ i, feed->feed_filename);
matches = 0;
} else {
AVCodecContext *ccf, *ccs;
if (!argbuf[0])
break;
- feed->child_argv[i] = av_malloc(strlen(argbuf + 1));
+ feed->child_argv[i] = av_malloc(strlen(argbuf) + 1);
strcpy(feed->child_argv[i], argbuf);
}
audio_id = stream->fmt->audio_codec;
video_id = stream->fmt->video_codec;
}
+ } else if (!strcasecmp(cmd, "InputFormat")) {
+ stream->ifmt = av_find_input_format(arg);
+ if (!stream->ifmt) {
+ fprintf(stderr, "%s:%d: Unknown input format: %s\n",
+ filename, line_num, arg);
+ }
} else if (!strcasecmp(cmd, "FaviconURL")) {
if (stream && stream->stream_type == STREAM_TYPE_STATUS) {
get_arg(stream->feed_filename, sizeof(stream->feed_filename), &p);
}
} else if (!strcasecmp(cmd, "VideoHighQuality")) {
if (stream) {
- video_enc.flags |= CODEC_FLAG_HQ;
+ video_enc.mb_decision = FF_MB_DECISION_BITS;
}
} else if (!strcasecmp(cmd, "Video4MotionVector")) {
if (stream) {
- video_enc.flags |= CODEC_FLAG_HQ;
+ video_enc.mb_decision = FF_MB_DECISION_BITS; //FIXME remove
video_enc.flags |= CODEC_FLAG_4MV;
}
} else if (!strcasecmp(cmd, "VideoQDiff")) {
}
#endif
-static void help(void)
+static void show_banner(void)
+{
+ printf("ffserver version " FFMPEG_VERSION ", Copyright (c) 2000-2003 Fabrice Bellard\n");
+}
+
+static void show_help(void)
{
- printf("ffserver version " FFMPEG_VERSION ", Copyright (c) 2000, 2001, 2002 Fabrice Bellard\n"
- "usage: ffserver [-L] [-h] [-f configfile]\n"
+ show_banner();
+ printf("usage: ffserver [-L] [-h] [-f configfile]\n"
"Hyper fast multi format Audio/Video streaming server\n"
"\n"
- "-L : print the LICENCE\n"
+ "-L : print the LICENSE\n"
"-h : this help\n"
"-f configfile : use configfile instead of /etc/ffserver.conf\n"
);
}
-static void licence(void)
+static void show_license(void)
{
+ show_banner();
printf(
- "ffserver version " FFMPEG_VERSION "\n"
- "Copyright (c) 2000, 2001, 2002 Fabrice Bellard\n"
"This library is free software; you can redistribute it and/or\n"
"modify it under the terms of the GNU Lesser General Public\n"
"License as published by the Free Software Foundation; either\n"
break;
switch(c) {
case 'L':
- licence();
+ show_license();
exit(1);
case '?':
case 'h':
- help();
+ show_help();
exit(1);
case 'n':
no_launch = 1;
projects / ffmpeg / blobdiff