[FFmpeg-cvslog] r13541 - trunk/ffserver.c
bcoudurier
subversion
Fri May 30 01:11:26 CEST 2008
Author: bcoudurier
Date: Fri May 30 01:11:25 2008
New Revision: 13541
Log:
cosmetics, reindent
Modified:
trunk/ffserver.c
Modified: trunk/ffserver.c
==============================================================================
--- trunk/ffserver.c (original)
+++ trunk/ffserver.c Fri May 30 01:11:25 2008
@@ -2043,152 +2043,152 @@ static int http_prepare_data(HTTPContext
break;
case HTTPSTATE_SEND_DATA:
/* find a new packet */
- /* read a packet from the input stream */
- if (c->stream->feed)
- ffm_set_write_index(c->fmt_in,
- c->stream->feed->feed_write_index,
- c->stream->feed->feed_size);
+ /* read a packet from the input stream */
+ if (c->stream->feed)
+ ffm_set_write_index(c->fmt_in,
+ c->stream->feed->feed_write_index,
+ c->stream->feed->feed_size);
- if (c->stream->max_time &&
- c->stream->max_time + c->start_time - cur_time < 0)
- /* We have timed out */
- c->state = HTTPSTATE_SEND_DATA_TRAILER;
- else {
- AVPacket pkt;
- 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
- ffm file, so must wait for more data */
- c->state = HTTPSTATE_WAIT_FEED;
- return 1; /* state changed */
- } else {
- 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;
- }
- }
+ if (c->stream->max_time &&
+ c->stream->max_time + c->start_time - cur_time < 0)
+ /* We have timed out */
+ c->state = HTTPSTATE_SEND_DATA_TRAILER;
+ else {
+ AVPacket pkt;
+ 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
+ ffm file, so must wait for more data */
+ c->state = HTTPSTATE_WAIT_FEED;
+ return 1; /* state changed */
} else {
- /* update first pts if needed */
- if (c->first_pts == AV_NOPTS_VALUE) {
- c->first_pts = av_rescale_q(pkt.dts, c->fmt_in->streams[pkt.stream_index]->time_base, AV_TIME_BASE_Q);
- c->start_time = cur_time;
+ 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;
}
- /* send it to the appropriate stream */
- if (c->stream->feed) {
- /* if coming from a feed, select the right stream */
- if (c->switch_pending) {
- c->switch_pending = 0;
- for(i=0;i<c->stream->nb_streams;i++) {
- if (c->switch_feed_streams[i] == pkt.stream_index)
- if (pkt.flags & PKT_FLAG_KEY)
- do_switch_stream(c, i);
- if (c->switch_feed_streams[i] >= 0)
- c->switch_pending = 1;
- }
- }
+ }
+ } else {
+ /* update first pts if needed */
+ if (c->first_pts == AV_NOPTS_VALUE) {
+ c->first_pts = av_rescale_q(pkt.dts, c->fmt_in->streams[pkt.stream_index]->time_base, AV_TIME_BASE_Q);
+ c->start_time = cur_time;
+ }
+ /* send it to the appropriate stream */
+ if (c->stream->feed) {
+ /* if coming from a feed, select the right stream */
+ if (c->switch_pending) {
+ c->switch_pending = 0;
for(i=0;i<c->stream->nb_streams;i++) {
- if (c->feed_streams[i] == pkt.stream_index) {
- pkt.stream_index = i;
+ if (c->switch_feed_streams[i] == pkt.stream_index)
if (pkt.flags & PKT_FLAG_KEY)
- c->got_key_frame |= 1 << i;
- /* See if we have all the key frames, then
- * we start to send. This logic is not quite
- * right, but it works for the case of a
- * single video stream with one or more
- * audio streams (for which every frame is
- * typically a key frame).
- */
- if (!c->stream->send_on_key ||
- ((c->got_key_frame + 1) >> c->stream->nb_streams))
- goto send_it;
- }
+ do_switch_stream(c, i);
+ if (c->switch_feed_streams[i] >= 0)
+ c->switch_pending = 1;
}
- } else {
- AVCodecContext *codec;
+ }
+ for(i=0;i<c->stream->nb_streams;i++) {
+ if (c->feed_streams[i] == pkt.stream_index) {
+ pkt.stream_index = i;
+ if (pkt.flags & PKT_FLAG_KEY)
+ c->got_key_frame |= 1 << i;
+ /* See if we have all the key frames, then
+ * we start to send. This logic is not quite
+ * right, but it works for the case of a
+ * single video stream with one or more
+ * audio streams (for which every frame is
+ * typically a key frame).
+ */
+ if (!c->stream->send_on_key ||
+ ((c->got_key_frame + 1) >> c->stream->nb_streams))
+ goto send_it;
+ }
+ }
+ } else {
+ AVCodecContext *codec;
- send_it:
- /* specific handling for RTP: we use several
- 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 = av_rescale_q(pkt.dts, st->time_base, AV_TIME_BASE_Q);
- if (st->start_time != AV_NOPTS_VALUE)
- c->cur_pts -= av_rescale_q(st->start_time, st->time_base, AV_TIME_BASE_Q);
- c->cur_frame_duration = av_rescale_q(pkt.duration, st->time_base, AV_TIME_BASE_Q);
+ send_it:
+ /* specific handling for RTP: we use several
+ 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 = av_rescale_q(pkt.dts, st->time_base, AV_TIME_BASE_Q);
+ if (st->start_time != AV_NOPTS_VALUE)
+ c->cur_pts -= av_rescale_q(st->start_time, st->time_base, AV_TIME_BASE_Q);
+ c->cur_frame_duration = av_rescale_q(pkt.duration, st->time_base, AV_TIME_BASE_Q);
#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);
+ 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;
- } else {
- ctx = &c->fmt_ctx;
- /* Fudge here */
- codec = ctx->streams[pkt.stream_index]->codec;
+ /* 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;
+ } else {
+ ctx = &c->fmt_ctx;
+ /* Fudge here */
+ codec = ctx->streams[pkt.stream_index]->codec;
+ }
- if (c->is_packetized) {
- 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);
- }
- if (ret < 0) {
- /* XXX: potential leak */
- return -1;
- }
- if (pkt.dts != AV_NOPTS_VALUE)
- pkt.dts = av_rescale_q(pkt.dts,
- c->fmt_in->streams[pkt.stream_index]->time_base,
- ctx->streams[pkt.stream_index]->time_base);
- if (pkt.pts != AV_NOPTS_VALUE)
- pkt.pts = av_rescale_q(pkt.pts,
- c->fmt_in->streams[pkt.stream_index]->time_base,
- ctx->streams[pkt.stream_index]->time_base);
- if (av_write_frame(ctx, &pkt))
- c->state = HTTPSTATE_SEND_DATA_TRAILER;
+ if (c->is_packetized) {
+ 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);
+ }
+ if (ret < 0) {
+ /* XXX: potential leak */
+ return -1;
+ }
+ if (pkt.dts != AV_NOPTS_VALUE)
+ pkt.dts = av_rescale_q(pkt.dts,
+ c->fmt_in->streams[pkt.stream_index]->time_base,
+ ctx->streams[pkt.stream_index]->time_base);
+ if (pkt.pts != AV_NOPTS_VALUE)
+ pkt.pts = av_rescale_q(pkt.pts,
+ c->fmt_in->streams[pkt.stream_index]->time_base,
+ ctx->streams[pkt.stream_index]->time_base);
+ if (av_write_frame(ctx, &pkt))
+ c->state = HTTPSTATE_SEND_DATA_TRAILER;
- 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;
+ 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) {
- av_free_packet(&pkt);
- goto redo;
- }
+ codec->frame_number++;
+ if (len == 0) {
+ av_free_packet(&pkt);
+ goto redo;
}
- av_free_packet(&pkt);
}
+ av_free_packet(&pkt);
}
+ }
break;
default:
case HTTPSTATE_SEND_DATA_TRAILER:
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