mirror of https://git.ffmpeg.org/ffmpeg.git
ffserver: unify comment formating & drop unneeded braces
Signed-off-by: Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
This commit is contained in:
Reynaldo H. Verdejo Pinochet
parent
1714fe2990
commit
469c335c55
114
ffserver.c
114
ffserver.c
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@ -31,7 +31,7 @@
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#include <stdlib.h>
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#include <stdio.h>
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#include "libavformat/avformat.h"
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// FIXME those are internal headers, ffserver _really_ shouldn't use them
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/* FIXME: those are internal headers, ffserver _really_ shouldn't use them */
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#include "libavformat/ffm.h"
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#include "libavformat/network.h"
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#include "libavformat/os_support.h"
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@ -251,7 +251,8 @@ static unsigned int nb_connections;
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static uint64_t current_bandwidth;
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static int64_t cur_time; // Making this global saves on passing it around everywhere
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/* Making this global saves on passing it around everywhere */
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static int64_t cur_time;
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static AVLFG random_state;
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@ -630,9 +631,8 @@ static int http_server(void)
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poll_entry++;
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} else {
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/* when ffserver is doing the timing, we work by
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looking at which packet needs to be sent every
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10 ms */
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/* one tick wait XXX: 10 ms assumed */
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* looking at which packet needs to be sent every
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* 10 ms (one tick wait XXX: 10 ms assumed) */
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if (delay > 10)
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delay = 10;
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}
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@ -655,7 +655,7 @@ static int http_server(void)
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}
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/* wait for an event on one connection. We poll at least every
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second to handle timeouts */
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* second to handle timeouts */
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do {
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ret = poll(poll_table, poll_entry - poll_table, delay);
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if (ret < 0 && ff_neterrno() != AVERROR(EAGAIN) &&
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@ -900,11 +900,11 @@ static int handle_connection(HTTPContext *c)
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if ((ptr >= c->buffer + 2 && !memcmp(ptr-2, "\n\n", 2)) ||
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(ptr >= c->buffer + 4 && !memcmp(ptr-4, "\r\n\r\n", 4))) {
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/* request found : parse it and reply */
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if (c->state == HTTPSTATE_WAIT_REQUEST) {
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if (c->state == HTTPSTATE_WAIT_REQUEST)
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ret = http_parse_request(c);
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} else {
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else
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ret = rtsp_parse_request(c);
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}
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if (ret < 0)
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return -1;
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} else if (ptr >= c->buffer_end) {
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@ -949,8 +949,8 @@ static int handle_connection(HTTPContext *c)
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case HTTPSTATE_SEND_DATA_HEADER:
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case HTTPSTATE_SEND_DATA_TRAILER:
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/* for packetized output, we consider we can always write (the
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input streams set the speed). It may be better to verify
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that we do not rely too much on the kernel queues */
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* input streams set the speed). It may be better to verify
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* that we do not rely too much on the kernel queues */
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if (!c->is_packetized) {
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if (c->poll_entry->revents & (POLLERR | POLLHUP))
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return -1;
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@ -1277,8 +1277,10 @@ static int validate_acl(FFServerStream *stream, HTTPContext *c)
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return ret;
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}
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/* compute the real filename of a file by matching it without its
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extensions to all the stream's filenames */
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/**
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* compute the real filename of a file by matching it without its
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* extensions to all the stream's filenames
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*/
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static void compute_real_filename(char *filename, int max_size)
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{
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char file1[1024];
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@ -1396,7 +1398,7 @@ static int http_parse_request(HTTPContext *c)
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compute_real_filename(filename, sizeof(filename) - 1);
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}
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// "redirect" / request to index.html
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/* "redirect" request to index.html */
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if (!strlen(filename))
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av_strlcpy(filename, "index.html", sizeof(filename) - 1);
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@ -1735,8 +1737,9 @@ static int http_parse_request(HTTPContext *c)
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return 0;
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send_status:
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compute_status(c);
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c->http_error = 200; /* horrible : we use this value to avoid
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going to the send data state */
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/* horrible: we use this value to avoid
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* going to the send data state */
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c->http_error = 200;
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c->state = HTTPSTATE_SEND_HEADER;
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return 0;
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}
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@ -1847,8 +1850,8 @@ static void compute_status(HTTPContext *c)
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strcpy(eosf - 3, ".ram");
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else if (stream->fmt && !strcmp(stream->fmt->name, "rtp")) {
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/* generate a sample RTSP director if
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unicast. Generate an SDP redirector if
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multicast */
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* unicast. Generate an SDP redirector if
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* multicast */
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eosf = strrchr(sfilename, '.');
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if (!eosf)
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eosf = sfilename + strlen(sfilename);
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@ -2119,8 +2122,7 @@ static int64_t get_server_clock(HTTPContext *c)
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return (cur_time - c->start_time) * 1000;
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}
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/* return the estimated time at which the current packet must be sent
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(in us) */
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/* return the estimated time (in us) at which the current packet must be sent */
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static int64_t get_packet_send_clock(HTTPContext *c)
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{
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int bytes_left, bytes_sent, frame_bytes;
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@ -2158,7 +2160,8 @@ static int http_prepare_data(HTTPContext *c)
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AVStream *src;
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c->fmt_ctx.streams[i] = av_mallocz(sizeof(AVStream));
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/* if file or feed, then just take streams from FFServerStream struct */
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/* if file or feed, then just take streams from FFServerStream
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* struct */
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if (!c->stream->feed ||
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c->stream->feed == c->stream)
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src = c->stream->streams[i];
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@ -2223,7 +2226,7 @@ static int http_prepare_data(HTTPContext *c)
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if (ret < 0) {
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if (c->stream->feed) {
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/* if coming from feed, it means we reached the end of the
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ffm file, so must wait for more data */
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* ffm file, so must wait for more data */
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c->state = HTTPSTATE_WAIT_FEED;
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return 1; /* state changed */
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}
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@ -2310,9 +2313,9 @@ static int http_prepare_data(HTTPContext *c)
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max_packet_size = c->rtp_handles[c->packet_stream_index]->max_packet_size;
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ret = ffio_open_dyn_packet_buf(&ctx->pb,
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max_packet_size);
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} else {
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} else
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ret = avio_open_dyn_buf(&ctx->pb);
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}
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if (ret < 0) {
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/* XXX: potential leak */
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return -1;
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@ -2375,7 +2378,8 @@ static int http_prepare_data(HTTPContext *c)
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/* should convert the format at the same time */
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/* send data starting at c->buffer_ptr to the output connection
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* (either UDP or TCP) */
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* (either UDP or TCP)
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*/
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static int http_send_data(HTTPContext *c)
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{
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int len, ret;
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@ -2456,8 +2460,8 @@ static int http_send_data(HTTPContext *c)
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rtsp_c->packet_buffer_ptr += len;
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if (rtsp_c->packet_buffer_ptr < rtsp_c->packet_buffer_end) {
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/* if we could not send all the data, we will
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send it later, so a new state is needed to
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"lock" the RTSP TCP connection */
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* send it later, so a new state is needed to
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* "lock" the RTSP TCP connection */
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rtsp_c->state = RTSPSTATE_SEND_PACKET;
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break;
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} else
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@ -2585,12 +2589,11 @@ static int http_receive_data(HTTPContext *c)
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goto fail;
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c->buffer_ptr = c->buffer;
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break;
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} else if (++loop_run > 10) {
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} else if (++loop_run > 10)
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/* no chunk header, abort */
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goto fail;
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} else {
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else
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c->buffer_ptr++;
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}
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}
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if (c->buffer_end > c->buffer_ptr) {
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@ -2623,7 +2626,7 @@ static int http_receive_data(HTTPContext *c)
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if (c->buffer_ptr >= c->buffer_end) {
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FFServerStream *feed = c->stream;
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/* a packet has been received : write it in the store, except
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if header */
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* if header */
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if (c->data_count > FFM_PACKET_SIZE) {
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/* XXX: use llseek or url_seek
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* XXX: Should probably fail? */
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@ -2829,10 +2832,10 @@ static int rtsp_parse_request(HTTPContext *c)
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the_end:
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len = avio_close_dyn_buf(c->pb, &c->pb_buffer);
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c->pb = NULL; /* safety */
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if (len < 0) {
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if (len < 0)
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/* XXX: cannot do more */
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return -1;
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}
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c->buffer_ptr = c->pb_buffer;
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c->buffer_end = c->pb_buffer + len;
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c->state = RTSPSTATE_SEND_REPLY;
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@ -2851,9 +2854,9 @@ static int prepare_sdp_description(FFServerStream *stream, uint8_t **pbuffer,
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*pbuffer = NULL;
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avc = avformat_alloc_context();
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if (!avc || !rtp_format) {
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if (!avc || !rtp_format)
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return -1;
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}
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avc->oformat = rtp_format;
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av_dict_set(&avc->metadata, "title",
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entry ? entry->value : "No Title", 0);
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@ -2862,9 +2865,8 @@ static int prepare_sdp_description(FFServerStream *stream, uint8_t **pbuffer,
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snprintf(avc->filename, 1024, "rtp://%s:%d?multicast=1?ttl=%d",
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inet_ntoa(stream->multicast_ip),
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stream->multicast_port, stream->multicast_ttl);
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} else {
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} else
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snprintf(avc->filename, 1024, "rtp://0.0.0.0");
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}
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avc->streams = av_malloc_array(avc->nb_streams, sizeof(*avc->streams));
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if (!avc->streams)
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@ -2894,7 +2896,7 @@ static int prepare_sdp_description(FFServerStream *stream, uint8_t **pbuffer,
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static void rtsp_cmd_options(HTTPContext *c, const char *url)
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{
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// rtsp_reply_header(c, RTSP_STATUS_OK);
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/* rtsp_reply_header(c, RTSP_STATUS_OK); */
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avio_printf(c->pb, "RTSP/1.0 %d %s\r\n", RTSP_STATUS_OK, "OK");
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avio_printf(c->pb, "CSeq: %d\r\n", c->seq);
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avio_printf(c->pb, "Public: %s\r\n",
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@ -3061,7 +3063,7 @@ static void rtsp_cmd_setup(HTTPContext *c, const char *url,
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}
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/* test if stream is OK (test needed because several SETUP needs
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to be done for a given file) */
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* to be done for a given file) */
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if (rtp_c->stream != stream) {
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rtsp_reply_error(c, RTSP_STATUS_SERVICE);
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return;
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@ -3122,8 +3124,10 @@ static void rtsp_cmd_setup(HTTPContext *c, const char *url,
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}
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/* find an RTP connection by using the session ID. Check consistency
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with filename */
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/**
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* find an RTP connection by using the session ID. Check consistency
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* with filename
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*/
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static HTTPContext *find_rtp_session_with_url(const char *url,
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const char *session_id)
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{
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@ -3146,10 +3150,10 @@ static HTTPContext *find_rtp_session_with_url(const char *url,
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for(s=0; s<rtp_c->stream->nb_streams; ++s) {
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snprintf(buf, sizeof(buf), "%s/streamid=%d",
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rtp_c->stream->filename, s);
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if(!strncmp(path, buf, sizeof(buf))) {
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// XXX: Should we reply with RTSP_STATUS_ONLY_AGGREGATE if nb_streams>1?
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if(!strncmp(path, buf, sizeof(buf)))
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/* XXX: Should we reply with RTSP_STATUS_ONLY_AGGREGATE
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* if nb_streams>1? */
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return rtp_c;
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}
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}
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len = strlen(path);
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if (len > 0 && path[len - 1] == '/' &&
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@ -3227,7 +3231,7 @@ static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr,
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const char *proto_str;
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/* XXX: should output a warning page when coming
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close to the connection limit */
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* close to the connection limit */
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if (nb_connections >= config.nb_max_connections)
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goto fail;
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@ -3282,9 +3286,11 @@ static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr,
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return NULL;
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}
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/* add a new RTP stream in an RTP connection (used in RTSP SETUP
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command). If RTP/TCP protocol is used, TCP connection 'rtsp_c' is
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used. */
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/**
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* add a new RTP stream in an RTP connection (used in RTSP SETUP
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* command). If RTP/TCP protocol is used, TCP connection 'rtsp_c' is
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* used.
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*/
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static int rtp_new_av_stream(HTTPContext *c,
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int stream_index, struct sockaddr_in *dest_addr,
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HTTPContext *rtsp_c)
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@ -3362,10 +3368,10 @@ static int rtp_new_av_stream(HTTPContext *c,
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/* normally, no packets should be output here, but the packet size may
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* be checked */
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if (ffio_open_dyn_packet_buf(&ctx->pb, max_packet_size) < 0) {
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if (ffio_open_dyn_packet_buf(&ctx->pb, max_packet_size) < 0)
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/* XXX: close stream */
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goto fail;
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}
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if (avformat_write_header(ctx, NULL) < 0) {
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fail:
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if (h)
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@ -3402,12 +3408,12 @@ static AVStream *add_av_stream1(FFServerStream *stream,
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return NULL;
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}
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avcodec_copy_context(fst->codec, codec);
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} else {
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} else
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/* live streams must use the actual feed's codec since it may be
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* updated later to carry extradata needed by them.
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*/
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fst->codec = codec;
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}
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fst->priv_data = av_mallocz(sizeof(FeedData));
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fst->index = stream->nb_streams;
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avpriv_set_pts_info(fst, 33, 1, 90000);
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@ -3539,7 +3545,7 @@ static void build_file_streams(void)
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/* open stream */
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if (stream->fmt && !strcmp(stream->fmt->name, "rtp")) {
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/* specific case : if transport stream output to RTP,
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we use a raw transport stream reader */
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* we use a raw transport stream reader */
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av_dict_set(&stream->in_opts, "mpeg2ts_compute_pcr", "1", 0);
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}
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@ -3561,7 +3567,7 @@ static void build_file_streams(void)
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remove_stream(stream);
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} else {
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/* find all the AVStreams inside and reference them in
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'stream' */
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* 'stream' */
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if (avformat_find_stream_info(infile, NULL) < 0) {
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http_log("Could not find codec parameters from '%s'\n",
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stream->feed_filename);
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