[SCM] libav/experimental: RTP multicast begins to work in MPEG1 - simplified stream bandwidth computation (no need to recompute it at each request)
siretart at users.alioth.debian.org
siretart at users.alioth.debian.org
Sun Jun 30 15:34:21 UTC 2013
The following commit has been merged in the experimental branch:
commit 6edd6884b5a8380979649d80ccf3454684497f3b
Author: Fabrice Bellard <fabrice at bellard.org>
Date: Wed Nov 20 18:05:45 2002 +0000
RTP multicast begins to work in MPEG1 - simplified stream bandwidth computation (no need to recompute it at each request)
Originally committed as revision 1260 to svn://svn.ffmpeg.org/ffmpeg/trunk
diff --git a/ffserver.c b/ffserver.c
index 2779e28..a9250ca 100644
--- a/ffserver.c
+++ b/ffserver.c
@@ -120,7 +120,6 @@ typedef struct HTTPContext {
AVFormatContext fmt_ctx; /* instance of FFStream for one user */
int last_packet_sent; /* true if last data packet was sent */
int suppress_log;
- int bandwidth;
DataRateData datarate;
int wmp_client_id;
char protocol[16];
@@ -190,12 +189,15 @@ typedef struct FFStream {
time_t pid_start; /* Of ffmpeg process */
char **child_argv;
struct FFStream *next;
+ int bandwidth; /* bandwidth, in kbits/s */
/* RTSP options */
char *rtsp_option;
/* multicast specific */
int is_multicast;
struct in_addr multicast_ip;
int multicast_port; /* first port used for multicast */
+ int multicast_ttl;
+ int loop; /* if true, send the stream in loops (only meaningful if file) */
/* feed specific */
int feed_opened; /* true if someone is writing to the feed */
@@ -247,7 +249,7 @@ static int prepare_sdp_description(FFStream *stream, UINT8 **pbuffer,
struct in_addr my_ip);
/* RTP handling */
-static HTTPContext *rtp_new_connection(HTTPContext *rtsp_c,
+static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr,
FFStream *stream, const char *session_id);
static int rtp_new_av_stream(HTTPContext *c,
int stream_index, struct sockaddr_in *dest_addr);
@@ -263,8 +265,8 @@ static int need_to_start_children;
int nb_max_connections;
int nb_connections;
-int nb_max_bandwidth;
-int nb_bandwidth;
+int max_bandwidth;
+int current_bandwidth;
static long cur_time; // Making this global saves on passing it around everywhere
@@ -290,25 +292,31 @@ static void http_log(char *fmt, ...)
va_end(ap);
}
-static void log_connection(HTTPContext *c)
+static char *ctime1(char *buf2)
{
- char buf1[32], buf2[32], *p;
time_t ti;
+ char *p;
- if (c->suppress_log)
- return;
-
- /* XXX: reentrant function ? */
- p = inet_ntoa(c->from_addr.sin_addr);
- strcpy(buf1, p);
ti = time(NULL);
p = ctime(&ti);
strcpy(buf2, p);
p = buf2 + strlen(p) - 1;
if (*p == '\n')
*p = '\0';
+ return buf2;
+}
+
+static void log_connection(HTTPContext *c)
+{
+ char buf2[32];
+
+ if (c->suppress_log)
+ return;
+
http_log("%s - - [%s] \"%s %s %s\" %d %lld\n",
- buf1, buf2, c->method, c->url, c->protocol, (c->http_error ? c->http_error : 200), c->data_count);
+ inet_ntoa(c->from_addr.sin_addr),
+ ctime1(buf2), c->method, c->url,
+ c->protocol, (c->http_error ? c->http_error : 200), c->data_count);
}
static void update_datarate(DataRateData *drd, INT64 count)
@@ -331,7 +339,7 @@ static int compute_datarate(DataRateData *drd, INT64 count)
{
if (cur_time == drd->time1)
return 0;
-
+
return ((count - drd->count1) * 1000) / (cur_time - drd->time1);
}
@@ -340,7 +348,6 @@ static int get_longterm_datarate(DataRateData *drd, INT64 count)
/* You get the first 3 seconds flat out */
if (cur_time - drd->time1 < 3000)
return 0;
-
return compute_datarate(drd, count);
}
@@ -431,6 +438,61 @@ static int socket_open_listen(struct sockaddr_in *my_addr)
return server_fd;
}
+/* start all multicast streams */
+static void start_multicast(void)
+{
+ FFStream *stream;
+ char session_id[32];
+ HTTPContext *rtp_c;
+ struct sockaddr_in dest_addr;
+ int default_port, stream_index;
+
+ default_port = 6000;
+ for(stream = first_stream; stream != NULL; stream = stream->next) {
+ if (stream->is_multicast) {
+ /* open the RTP connection */
+ snprintf(session_id, sizeof(session_id),
+ "%08x%08x", (int)random(), (int)random());
+
+ /* choose a port if none given */
+ if (stream->multicast_port == 0) {
+ stream->multicast_port = default_port;
+ default_port += 100;
+ }
+
+ dest_addr.sin_family = AF_INET;
+ 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);
+ if (!rtp_c) {
+ continue;
+ }
+ if (open_input_stream(rtp_c, "") < 0) {
+ fprintf(stderr, "Could not open input stream for stream '%s'\n",
+ stream->filename);
+ 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) {
+ fprintf(stderr, "Could not open input stream %d for stream '%s'\n",
+ stream_index, stream->filename);
+ continue;
+ }
+ }
+
+ /* change state to send data */
+ rtp_c->state = HTTPSTATE_SEND_DATA;
+ }
+ }
+}
/* main loop of the http server */
static int http_server(void)
@@ -454,6 +516,9 @@ static int http_server(void)
first_http_ctx = NULL;
nb_connections = 0;
first_http_ctx = NULL;
+
+ start_multicast();
+
for(;;) {
poll_entry = poll_table;
poll_entry->fd = server_fd;
@@ -669,7 +734,8 @@ static void close_connection(HTTPContext *c)
}
}
- nb_bandwidth -= c->bandwidth;
+ if (c->stream)
+ current_bandwidth -= c->stream->bandwidth;
av_freep(&c->pb_buffer);
av_free(c->buffer);
av_free(c);
@@ -1125,9 +1191,7 @@ static int http_parse_request(HTTPContext *c)
compute_real_filename(filename, sizeof(url) - 1);
} else if (match_ext(filename, "sdp")) {
redir_type = REDIR_SDP;
- printf("before %s\n", filename);
compute_real_filename(filename, sizeof(url) - 1);
- printf("after %s\n", filename);
}
stream = first_stream;
@@ -1174,26 +1238,10 @@ static int http_parse_request(HTTPContext *c)
}
if (post == 0 && stream->stream_type == STREAM_TYPE_LIVE) {
- /* See if we meet the bandwidth requirements */
- for(i=0;i<stream->nb_streams;i++) {
- AVStream *st = stream->streams[i];
- switch(st->codec.codec_type) {
- case CODEC_TYPE_AUDIO:
- c->bandwidth += st->codec.bit_rate;
- break;
- case CODEC_TYPE_VIDEO:
- c->bandwidth += st->codec.bit_rate;
- break;
- default:
- av_abort();
- }
- }
+ current_bandwidth += stream->bandwidth;
}
-
- c->bandwidth /= 1000;
- nb_bandwidth += c->bandwidth;
-
- if (post == 0 && nb_max_bandwidth < nb_bandwidth) {
+
+ if (post == 0 && max_bandwidth < current_bandwidth) {
c->http_error = 200;
q = c->buffer;
q += sprintf(q, "HTTP/1.0 200 Server too busy\r\n");
@@ -1202,7 +1250,7 @@ static int http_parse_request(HTTPContext *c)
q += sprintf(q, "<html><head><title>Too busy</title></head><body>\r\n");
q += sprintf(q, "The server is too busy to serve your request at this time.<p>\r\n");
q += sprintf(q, "The bandwidth being served (including your stream) is %dkbit/sec, and this exceeds the limit of %dkbit/sec\r\n",
- nb_bandwidth, nb_max_bandwidth);
+ current_bandwidth, max_bandwidth);
q += sprintf(q, "</body></html>\r\n");
/* prepare output buffer */
@@ -1580,7 +1628,7 @@ static void compute_stats(HTTPContext *c)
}
url_fprintf(pb, "<TD align=center> %s <TD align=right> %d <TD align=right> %d <TD> %s %s <TD align=right> %d <TD> %s %s",
stream->fmt->name,
- (audio_bit_rate + video_bit_rate) / 1000,
+ stream->bandwidth,
video_bit_rate / 1000, video_codec_name, video_codec_name_extra,
audio_bit_rate / 1000, audio_codec_name, audio_codec_name_extra);
if (stream->feed) {
@@ -1702,7 +1750,7 @@ static void compute_stats(HTTPContext *c)
nb_connections, nb_max_connections);
url_fprintf(pb, "Bandwidth in use: %dk / %dk<BR>\n",
- nb_bandwidth, nb_max_bandwidth);
+ current_bandwidth, max_bandwidth);
url_fprintf(pb, "<TABLE>\n");
url_fprintf(pb, "<TR><th>#<th>File<th>IP<th>Proto<th>State<th>Target bits/sec<th>Actual bits/sec<th>Bytes transferred\n");
@@ -1999,7 +2047,7 @@ static int compute_send_delay(HTTPContext *c)
{
int datarate = 8 * get_longterm_datarate(&c->datarate, c->data_count);
- if (datarate > c->bandwidth * 2000) {
+ if (datarate > c->stream->bandwidth * 2000) {
return 1000;
}
return 0;
@@ -2082,6 +2130,7 @@ static int http_prepare_data(HTTPContext *c)
return 1; /* state changed */
}
}
+ redo:
if (av_read_frame(c->fmt_in, &pkt) < 0) {
if (c->stream->feed && c->stream->feed->feed_opened) {
/* if coming from feed, it means we reached the end of the
@@ -2089,8 +2138,17 @@ static int http_prepare_data(HTTPContext *c)
c->state = HTTPSTATE_WAIT_FEED;
return 1; /* state changed */
} else {
- /* must send trailer now because eof or error */
- c->state = HTTPSTATE_SEND_DATA_TRAILER;
+ if (c->stream->loop) {
+ av_close_input_file(c->fmt_in);
+ c->fmt_in = NULL;
+ if (open_input_stream(c, "") < 0)
+ goto no_loop;
+ goto redo;
+ } else {
+ no_loop:
+ /* must send trailer now because eof or error */
+ c->state = HTTPSTATE_SEND_DATA_TRAILER;
+ }
}
} else {
/* update first pts if needed */
@@ -2143,6 +2201,8 @@ static int http_prepare_data(HTTPContext *c)
c->packet_stream_index = pkt.stream_index;
ctx = c->rtp_ctx[c->packet_stream_index];
codec = &ctx->streams[0]->codec;
+ /* only one stream per RTP connection */
+ pkt.stream_index = 0;
} else {
ctx = &c->fmt_ctx;
/* Fudge here */
@@ -2721,7 +2781,7 @@ static void rtsp_cmd_setup(HTTPContext *c, const char *url,
/* 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, stream, h->session_id);
+ rtp_c = rtp_new_connection(&c->from_addr, stream, h->session_id);
if (!rtp_c) {
rtsp_reply_error(c, RTSP_STATUS_BANDWIDTH);
return;
@@ -2923,7 +2983,7 @@ static void rtsp_cmd_teardown(HTTPContext *c, const char *url, RTSPHeader *h)
/********************************************************************/
/* RTP handling */
-static HTTPContext *rtp_new_connection(HTTPContext *rtsp_c,
+static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr,
FFStream *stream, const char *session_id)
{
HTTPContext *c = NULL;
@@ -2940,7 +3000,7 @@ static HTTPContext *rtp_new_connection(HTTPContext *rtsp_c,
c->fd = -1;
c->poll_entry = NULL;
- c->from_addr = rtsp_c->from_addr;
+ c->from_addr = *from_addr;
c->buffer_size = IOBUFFER_INIT_SIZE;
c->buffer = av_malloc(c->buffer_size);
if (!c->buffer)
@@ -2953,6 +3013,8 @@ static HTTPContext *rtp_new_connection(HTTPContext *rtsp_c,
/* protocol is shown in statistics */
pstrcpy(c->protocol, sizeof(c->protocol), "RTP");
+ current_bandwidth += stream->bandwidth;
+
c->next = first_http_ctx;
first_http_ctx = c;
return c;
@@ -2976,7 +3038,8 @@ static int rtp_new_av_stream(HTTPContext *c,
char *ipaddr;
URLContext *h;
UINT8 *dummy_buf;
-
+ char buf2[32];
+
/* now we can open the relevant output stream */
ctx = av_mallocz(sizeof(AVFormatContext));
if (!ctx)
@@ -3002,10 +3065,19 @@ static int rtp_new_av_stream(HTTPContext *c,
/* build destination RTP address */
ipaddr = inet_ntoa(dest_addr->sin_addr);
- snprintf(ctx->filename, sizeof(ctx->filename),
- "rtp://%s:%d", ipaddr, ntohs(dest_addr->sin_port));
-
- printf("open %s\n", ctx->filename);
+ /* XXX: also pass as parameter to function ? */
+ if (c->stream->is_multicast) {
+ int ttl;
+ ttl = c->stream->multicast_ttl;
+ if (!ttl)
+ ttl = 16;
+ snprintf(ctx->filename, sizeof(ctx->filename),
+ "rtp://%s:%d?multicast=1&ttl=%d",
+ ipaddr, ntohs(dest_addr->sin_port), ttl);
+ } else {
+ snprintf(ctx->filename, sizeof(ctx->filename),
+ "rtp://%s:%d", ipaddr, ntohs(dest_addr->sin_port));
+ }
if (url_open(&h, ctx->filename, URL_WRONLY) < 0)
goto fail;
@@ -3014,6 +3086,11 @@ static int rtp_new_av_stream(HTTPContext *c,
goto fail;
}
+ http_log("%s:%d - - [%s] \"RTPSTART %s/streamid=%d\"\n",
+ ipaddr, ntohs(dest_addr->sin_port),
+ ctime1(buf2),
+ c->stream->filename, stream_index);
+
/* 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) {
@@ -3286,6 +3363,29 @@ void build_feed_streams(void)
}
}
+/* compute the bandwidth used by each stream */
+static void compute_bandwidth(void)
+{
+ int bandwidth, i;
+ FFStream *stream;
+
+ for(stream = first_stream; stream != NULL; stream = stream->next) {
+ bandwidth = 0;
+ for(i=0;i<stream->nb_streams;i++) {
+ AVStream *st = stream->streams[i];
+ switch(st->codec.codec_type) {
+ case CODEC_TYPE_AUDIO:
+ case CODEC_TYPE_VIDEO:
+ bandwidth += st->codec.bit_rate;
+ break;
+ default:
+ break;
+ }
+ }
+ stream->bandwidth = (bandwidth + 999) / 1000;
+ }
+}
+
static void get_arg(char *buf, int buf_size, const char **pp)
{
const char *p;
@@ -3519,7 +3619,7 @@ int parse_ffconfig(const char *filename)
filename, line_num, arg);
errors++;
} else {
- nb_max_bandwidth = val;
+ max_bandwidth = val;
}
} else if (!strcasecmp(cmd, "CustomLog")) {
get_arg(logfilename, sizeof(logfilename), &p);
@@ -3952,12 +4052,22 @@ int parse_ffconfig(const char *filename)
errors++;
}
stream->is_multicast = 1;
+ stream->loop = 1; /* default is looping */
}
} else if (!strcasecmp(cmd, "MulticastPort")) {
get_arg(arg, sizeof(arg), &p);
if (stream) {
stream->multicast_port = atoi(arg);
}
+ } else if (!strcasecmp(cmd, "MulticastTTL")) {
+ get_arg(arg, sizeof(arg), &p);
+ if (stream) {
+ stream->multicast_ttl = atoi(arg);
+ }
+ } else if (!strcasecmp(cmd, "NoLoop")) {
+ if (stream) {
+ stream->loop = 0;
+ }
} else if (!strcasecmp(cmd, "</Stream>")) {
if (!stream) {
fprintf(stderr, "%s:%d: No corresponding <Stream> for </Stream>\n",
@@ -4162,7 +4272,7 @@ int main(int argc, char **argv)
my_rtsp_addr.sin_addr.s_addr = htonl (INADDR_ANY);
nb_max_connections = 5;
- nb_max_bandwidth = 1000;
+ max_bandwidth = 1000;
first_stream = NULL;
logfilename[0] = '\0';
@@ -4180,6 +4290,8 @@ int main(int argc, char **argv)
build_feed_streams();
+ compute_bandwidth();
+
/* put the process in background and detach it from its TTY */
if (ffserver_daemon) {
int pid;
--
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