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ffmpeg.wasm-core
Commit 469c335c authored by Reynaldo H. Verdejo Pinochet's avatar Reynaldo H. Verdejo Pinochet
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ffserver: unify comment formating & drop unneeded braces 

Signed-off-by: 's avatarReynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
parent 1714fe29
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Showing with 60 additions and 54 deletions
ffserver.c
View file @ 469c335c
......@@ -31,7 +31,7 @@
#include <stdlib.h>
#include <stdio.h>
#include "libavformat/avformat.h"
// FIXME those are internal headers, ffserver _really_ shouldn't use them
/* FIXME: those are internal headers, ffserver _really_ shouldn't use them */
#include "libavformat/ffm.h"
#include "libavformat/network.h"
#include "libavformat/os_support.h"
......@@ -251,7 +251,8 @@ static unsigned int nb_connections;
static uint64_t current_bandwidth;
static int64_t cur_time; // Making this global saves on passing it around everywhere
/* Making this global saves on passing it around everywhere */
static int64_t cur_time;
static AVLFG random_state;
......@@ -630,9 +631,8 @@ static int http_server(void)
poll_entry++;
} else {
/* when ffserver is doing the timing, we work by
looking at which packet needs to be sent every
10 ms */
/* one tick wait XXX: 10 ms assumed */
* looking at which packet needs to be sent every
* 10 ms (one tick wait XXX: 10 ms assumed) */
if (delay > 10)
delay = 10;
}
......@@ -655,7 +655,7 @@ static int http_server(void)
}
/* wait for an event on one connection. We poll at least every
second to handle timeouts */
* second to handle timeouts */
do {
ret = poll(poll_table, poll_entry - poll_table, delay);
if (ret < 0 && ff_neterrno() != AVERROR(EAGAIN) &&
......@@ -900,11 +900,11 @@ static int handle_connection(HTTPContext *c)
if ((ptr >= c->buffer + 2 && !memcmp(ptr-2, "\n\n", 2)) ||
(ptr >= c->buffer + 4 && !memcmp(ptr-4, "\r\n\r\n", 4))) {
/* request found : parse it and reply */
if (c->state == HTTPSTATE_WAIT_REQUEST) {
if (c->state == HTTPSTATE_WAIT_REQUEST)
ret = http_parse_request(c);
} else {
else
ret = rtsp_parse_request(c);
}
if (ret < 0)
return -1;
} else if (ptr >= c->buffer_end) {
......@@ -949,8 +949,8 @@ static int handle_connection(HTTPContext *c)
case HTTPSTATE_SEND_DATA_HEADER:
case HTTPSTATE_SEND_DATA_TRAILER:
/* for packetized output, we consider we can always write (the
input streams set the speed). It may be better to verify
that we do not rely too much on the kernel queues */
* input streams set the speed). It may be better to verify
* that we do not rely too much on the kernel queues */
if (!c->is_packetized) {
if (c->poll_entry->revents & (POLLERR | POLLHUP))
return -1;
......@@ -1277,8 +1277,10 @@ static int validate_acl(FFServerStream *stream, HTTPContext *c)
return ret;
}
/* compute the real filename of a file by matching it without its
extensions to all the stream's filenames */
/**
* compute the real filename of a file by matching it without its
* extensions to all the stream's filenames
*/
static void compute_real_filename(char *filename, int max_size)
{
char file1[1024];
......@@ -1396,7 +1398,7 @@ static int http_parse_request(HTTPContext *c)
compute_real_filename(filename, sizeof(filename) - 1);
}
// "redirect" / request to index.html
/* "redirect" request to index.html */
if (!strlen(filename))
av_strlcpy(filename, "index.html", sizeof(filename) - 1);
......@@ -1735,8 +1737,9 @@ static int http_parse_request(HTTPContext *c)
return 0;
send_status:
compute_status(c);
c->http_error = 200; /* horrible : we use this value to avoid
going to the send data state */
/* horrible: we use this value to avoid
* going to the send data state */
c->http_error = 200;
c->state = HTTPSTATE_SEND_HEADER;
return 0;
}
......@@ -1847,8 +1850,8 @@ static void compute_status(HTTPContext *c)
strcpy(eosf - 3, ".ram");
else if (stream->fmt && !strcmp(stream->fmt->name, "rtp")) {
/* generate a sample RTSP director if
unicast. Generate an SDP redirector if
multicast */
* unicast. Generate an SDP redirector if
* multicast */
eosf = strrchr(sfilename, '.');
if (!eosf)
eosf = sfilename + strlen(sfilename);
......@@ -2119,8 +2122,7 @@ static int64_t get_server_clock(HTTPContext *c)
return (cur_time - c->start_time) * 1000;
}
/* return the estimated time at which the current packet must be sent
(in us) */
/* return the estimated time (in us) at which the current packet must be sent */
static int64_t get_packet_send_clock(HTTPContext *c)
{
int bytes_left, bytes_sent, frame_bytes;
......@@ -2158,7 +2160,8 @@ static int http_prepare_data(HTTPContext *c)
AVStream *src;
c->fmt_ctx.streams[i] = av_mallocz(sizeof(AVStream));
/* if file or feed, then just take streams from FFServerStream struct */
/* if file or feed, then just take streams from FFServerStream
* struct */
if (!c->stream->feed ||
c->stream->feed == c->stream)
src = c->stream->streams[i];
......@@ -2223,7 +2226,7 @@ static int http_prepare_data(HTTPContext *c)
if (ret < 0) {
if (c->stream->feed) {
/* if coming from feed, it means we reached the end of the
ffm file, so must wait for more data */
* ffm file, so must wait for more data */
c->state = HTTPSTATE_WAIT_FEED;
return 1; /* state changed */
}
......@@ -2310,9 +2313,9 @@ static int http_prepare_data(HTTPContext *c)
max_packet_size = c->rtp_handles[c->packet_stream_index]->max_packet_size;
ret = ffio_open_dyn_packet_buf(&ctx->pb,
max_packet_size);
} else {
} else
ret = avio_open_dyn_buf(&ctx->pb);
}
if (ret < 0) {
/* XXX: potential leak */
return -1;
......@@ -2375,7 +2378,8 @@ static int http_prepare_data(HTTPContext *c)
/* should convert the format at the same time */
/* send data starting at c->buffer_ptr to the output connection
* (either UDP or TCP) */
* (either UDP or TCP)
*/
static int http_send_data(HTTPContext *c)
{
int len, ret;
......@@ -2456,8 +2460,8 @@ static int http_send_data(HTTPContext *c)
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 */
* send it later, so a new state is needed to
* "lock" the RTSP TCP connection */
rtsp_c->state = RTSPSTATE_SEND_PACKET;
break;
} else
......@@ -2585,13 +2589,12 @@ static int http_receive_data(HTTPContext *c)
goto fail;
c->buffer_ptr = c->buffer;
break;
} else if (++loop_run > 10) {
} else if (++loop_run > 10)
/* no chunk header, abort */
goto fail;
} else {
else
c->buffer_ptr++;
}
}
if (c->buffer_end > c->buffer_ptr) {
len = recv(c->fd, c->buffer_ptr,
......@@ -2623,7 +2626,7 @@ static int http_receive_data(HTTPContext *c)
if (c->buffer_ptr >= c->buffer_end) {
FFServerStream *feed = c->stream;
/* a packet has been received : write it in the store, except
if header */
* if header */
if (c->data_count > FFM_PACKET_SIZE) {
/* XXX: use llseek or url_seek
* XXX: Should probably fail? */
......@@ -2829,10 +2832,10 @@ static int rtsp_parse_request(HTTPContext *c)
the_end:
len = avio_close_dyn_buf(c->pb, &c->pb_buffer);
c->pb = NULL; /* safety */
if (len < 0) {
if (len < 0)
/* XXX: cannot do more */
return -1;
}
c->buffer_ptr = c->pb_buffer;
c->buffer_end = c->pb_buffer + len;
c->state = RTSPSTATE_SEND_REPLY;
......@@ -2851,9 +2854,9 @@ static int prepare_sdp_description(FFServerStream *stream, uint8_t **pbuffer,
*pbuffer = NULL;
avc = avformat_alloc_context();
if (!avc || !rtp_format) {
if (!avc || !rtp_format)
return -1;
}
avc->oformat = rtp_format;
av_dict_set(&avc->metadata, "title",
entry ? entry->value : "No Title", 0);
......@@ -2862,9 +2865,8 @@ static int prepare_sdp_description(FFServerStream *stream, uint8_t **pbuffer,
snprintf(avc->filename, 1024, "rtp://%s:%d?multicast=1?ttl=%d",
inet_ntoa(stream->multicast_ip),
stream->multicast_port, stream->multicast_ttl);
} else {
} else
snprintf(avc->filename, 1024, "rtp://0.0.0.0");
}
avc->streams = av_malloc_array(avc->nb_streams, sizeof(*avc->streams));
if (!avc->streams)
......@@ -2894,7 +2896,7 @@ static int prepare_sdp_description(FFServerStream *stream, uint8_t **pbuffer,
static void rtsp_cmd_options(HTTPContext *c, const char *url)
{
// rtsp_reply_header(c, RTSP_STATUS_OK);
/* rtsp_reply_header(c, RTSP_STATUS_OK); */
avio_printf(c->pb, "RTSP/1.0 %d %s\r\n", RTSP_STATUS_OK, "OK");
avio_printf(c->pb, "CSeq: %d\r\n", c->seq);
avio_printf(c->pb, "Public: %s\r\n",
......@@ -3061,7 +3063,7 @@ static void rtsp_cmd_setup(HTTPContext *c, const char *url,
}
/* test if stream is OK (test needed because several SETUP needs
to be done for a given file) */
* to be done for a given file) */
if (rtp_c->stream != stream) {
rtsp_reply_error(c, RTSP_STATUS_SERVICE);
return;
......@@ -3122,8 +3124,10 @@ static void rtsp_cmd_setup(HTTPContext *c, const char *url,
}
/* find an RTP connection by using the session ID. Check consistency
with filename */
/**
* find an RTP connection by using the session ID. Check consistency
* with filename
*/
static HTTPContext *find_rtp_session_with_url(const char *url,
const char *session_id)
{
......@@ -3146,11 +3150,11 @@ static HTTPContext *find_rtp_session_with_url(const char *url,
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?
if(!strncmp(path, buf, sizeof(buf)))
/* XXX: Should we reply with RTSP_STATUS_ONLY_AGGREGATE
* if nb_streams>1? */
return rtp_c;
}
}
len = strlen(path);
if (len > 0 && path[len - 1] == '/' &&
!strncmp(path, rtp_c->stream->filename, len - 1))
......@@ -3227,7 +3231,7 @@ static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr,
const char *proto_str;
/* XXX: should output a warning page when coming
close to the connection limit */
* close to the connection limit */
if (nb_connections >= config.nb_max_connections)
goto fail;
......@@ -3282,9 +3286,11 @@ static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr,
return NULL;
}
/* add a new RTP stream in an RTP connection (used in RTSP SETUP
command). If RTP/TCP protocol is used, TCP connection 'rtsp_c' is
used. */
/**
* add a new RTP stream in an RTP connection (used in RTSP SETUP
* 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,
HTTPContext *rtsp_c)
......@@ -3362,10 +3368,10 @@ static int rtp_new_av_stream(HTTPContext *c,
/* normally, no packets should be output here, but the packet size may
* be checked */
if (ffio_open_dyn_packet_buf(&ctx->pb, max_packet_size) < 0) {
if (ffio_open_dyn_packet_buf(&ctx->pb, max_packet_size) < 0)
/* XXX: close stream */
goto fail;
}
if (avformat_write_header(ctx, NULL) < 0) {
fail:
if (h)
......@@ -3402,12 +3408,12 @@ static AVStream *add_av_stream1(FFServerStream *stream,
return NULL;
}
avcodec_copy_context(fst->codec, codec);
} else {
} else
/* live streams must use the actual feed's codec since it may be
* updated later to carry extradata needed by them.
*/
fst->codec = codec;
}
fst->priv_data = av_mallocz(sizeof(FeedData));
fst->index = stream->nb_streams;
avpriv_set_pts_info(fst, 33, 1, 90000);
......@@ -3539,7 +3545,7 @@ static void build_file_streams(void)
/* open stream */
if (stream->fmt && !strcmp(stream->fmt->name, "rtp")) {
/* specific case : if transport stream output to RTP,
we use a raw transport stream reader */
* we use a raw transport stream reader */
av_dict_set(&stream->in_opts, "mpeg2ts_compute_pcr", "1", 0);
}
......@@ -3561,7 +3567,7 @@ static void build_file_streams(void)
remove_stream(stream);
} else {
/* find all the AVStreams inside and reference them in
'stream' */
* 'stream' */
if (avformat_find_stream_info(infile, NULL) < 0) {
http_log("Could not find codec parameters from '%s'\n",
stream->feed_filename);
......
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