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Commit 3b371676 authored by Baptiste Coudurier's avatar Baptiste Coudurier
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cosmetics, reindent 

Originally committed as revision 13541 to svn://svn.ffmpeg.org/ffmpeg/trunk
parent 1cb4d12c
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with 134 additions and 134 deletions
ffserver.c
+ 134
− 134
View file @ 3b371676
......@@ -2043,152 +2043,152 @@ static int http_prepare_data(HTTPContext *c)
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);
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;
}
}
/* 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 {
/* 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;
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);
}
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);
#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;
}
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;
codec->frame_number++;
if (len == 0) {
/* find RTP context */
c->packet_stream_index = pkt.stream_index;
ctx = c->rtp_ctx[c->packet_stream_index];
if(!ctx) {
av_free_packet(&pkt);
goto redo;
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;
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;
}
av_free_packet(&pkt);
}
av_free_packet(&pkt);
}
}
break;
default:
case HTTPSTATE_SEND_DATA_TRAILER:
......
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