ffserver: unify comment formating & drop unneeded braces

Signed-off-by: Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

1 files changed, 60 insertions, 54 deletions

diff --git a/ffserver.c b/ffserver.c
index 6b537b3304..1be545f909 100644
--- a/ffserver.c
+++ b/ffserver.c
@@ -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
@@ -2585,12 +2589,11 @@ 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) {
@@ -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)
 {
@@ -3146,10 +3150,10 @@ 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] == '/' &&
@@ -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);