mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-11-26 19:01:44 +02:00
cosmetics, reindent
Originally committed as revision 13541 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Baptiste Coudurier
parent
1cb4d12c72
commit
3b37167691
266
ffserver.c
266
ffserver.c
@ -2043,152 +2043,152 @@ static int http_prepare_data(HTTPContext *c)
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break;
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case HTTPSTATE_SEND_DATA:
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/* find a new packet */
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/* read a packet from the input stream */
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if (c->stream->feed)
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ffm_set_write_index(c->fmt_in,
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c->stream->feed->feed_write_index,
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c->stream->feed->feed_size);
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/* read a packet from the input stream */
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if (c->stream->feed)
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ffm_set_write_index(c->fmt_in,
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c->stream->feed->feed_write_index,
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c->stream->feed->feed_size);
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if (c->stream->max_time &&
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c->stream->max_time + c->start_time - cur_time < 0)
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/* We have timed out */
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c->state = HTTPSTATE_SEND_DATA_TRAILER;
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else {
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AVPacket pkt;
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redo:
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if (av_read_frame(c->fmt_in, &pkt) < 0) {
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if (c->stream->feed && c->stream->feed->feed_opened) {
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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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c->state = HTTPSTATE_WAIT_FEED;
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return 1; /* state changed */
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if (c->stream->max_time &&
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c->stream->max_time + c->start_time - cur_time < 0)
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/* We have timed out */
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c->state = HTTPSTATE_SEND_DATA_TRAILER;
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else {
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AVPacket pkt;
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redo:
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if (av_read_frame(c->fmt_in, &pkt) < 0) {
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if (c->stream->feed && c->stream->feed->feed_opened) {
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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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c->state = HTTPSTATE_WAIT_FEED;
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return 1; /* state changed */
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} else {
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if (c->stream->loop) {
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av_close_input_file(c->fmt_in);
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c->fmt_in = NULL;
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if (open_input_stream(c, "") < 0)
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goto no_loop;
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goto redo;
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} else {
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if (c->stream->loop) {
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av_close_input_file(c->fmt_in);
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c->fmt_in = NULL;
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if (open_input_stream(c, "") < 0)
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goto no_loop;
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goto redo;
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} else {
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no_loop:
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/* must send trailer now because eof or error */
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c->state = HTTPSTATE_SEND_DATA_TRAILER;
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no_loop:
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/* must send trailer now because eof or error */
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c->state = HTTPSTATE_SEND_DATA_TRAILER;
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}
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}
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} else {
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/* update first pts if needed */
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if (c->first_pts == AV_NOPTS_VALUE) {
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c->first_pts = av_rescale_q(pkt.dts, c->fmt_in->streams[pkt.stream_index]->time_base, AV_TIME_BASE_Q);
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c->start_time = cur_time;
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}
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/* send it to the appropriate stream */
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if (c->stream->feed) {
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/* if coming from a feed, select the right stream */
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if (c->switch_pending) {
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c->switch_pending = 0;
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for(i=0;i<c->stream->nb_streams;i++) {
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if (c->switch_feed_streams[i] == pkt.stream_index)
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if (pkt.flags & PKT_FLAG_KEY)
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do_switch_stream(c, i);
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if (c->switch_feed_streams[i] >= 0)
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c->switch_pending = 1;
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}
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}
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for(i=0;i<c->stream->nb_streams;i++) {
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if (c->feed_streams[i] == pkt.stream_index) {
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pkt.stream_index = i;
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if (pkt.flags & PKT_FLAG_KEY)
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c->got_key_frame |= 1 << i;
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/* See if we have all the key frames, then
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* we start to send. This logic is not quite
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* right, but it works for the case of a
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* single video stream with one or more
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* audio streams (for which every frame is
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* typically a key frame).
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*/
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if (!c->stream->send_on_key ||
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((c->got_key_frame + 1) >> c->stream->nb_streams))
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goto send_it;
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}
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}
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} else {
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/* update first pts if needed */
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if (c->first_pts == AV_NOPTS_VALUE) {
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c->first_pts = av_rescale_q(pkt.dts, c->fmt_in->streams[pkt.stream_index]->time_base, AV_TIME_BASE_Q);
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c->start_time = cur_time;
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}
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/* send it to the appropriate stream */
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if (c->stream->feed) {
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/* if coming from a feed, select the right stream */
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if (c->switch_pending) {
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c->switch_pending = 0;
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for(i=0;i<c->stream->nb_streams;i++) {
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if (c->switch_feed_streams[i] == pkt.stream_index)
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if (pkt.flags & PKT_FLAG_KEY)
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do_switch_stream(c, i);
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if (c->switch_feed_streams[i] >= 0)
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c->switch_pending = 1;
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}
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}
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for(i=0;i<c->stream->nb_streams;i++) {
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if (c->feed_streams[i] == pkt.stream_index) {
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pkt.stream_index = i;
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if (pkt.flags & PKT_FLAG_KEY)
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c->got_key_frame |= 1 << i;
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/* See if we have all the key frames, then
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* we start to send. This logic is not quite
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* right, but it works for the case of a
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* single video stream with one or more
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* audio streams (for which every frame is
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* typically a key frame).
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*/
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if (!c->stream->send_on_key ||
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((c->got_key_frame + 1) >> c->stream->nb_streams))
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goto send_it;
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}
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}
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} else {
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AVCodecContext *codec;
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AVCodecContext *codec;
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send_it:
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/* specific handling for RTP: we use several
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output stream (one for each RTP
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connection). XXX: need more abstract handling */
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if (c->is_packetized) {
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AVStream *st;
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/* compute send time and duration */
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st = c->fmt_in->streams[pkt.stream_index];
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c->cur_pts = av_rescale_q(pkt.dts, st->time_base, AV_TIME_BASE_Q);
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if (st->start_time != AV_NOPTS_VALUE)
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c->cur_pts -= av_rescale_q(st->start_time, st->time_base, AV_TIME_BASE_Q);
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c->cur_frame_duration = av_rescale_q(pkt.duration, st->time_base, AV_TIME_BASE_Q);
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send_it:
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/* specific handling for RTP: we use several
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output stream (one for each RTP
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connection). XXX: need more abstract handling */
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if (c->is_packetized) {
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AVStream *st;
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/* compute send time and duration */
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st = c->fmt_in->streams[pkt.stream_index];
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c->cur_pts = av_rescale_q(pkt.dts, st->time_base, AV_TIME_BASE_Q);
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if (st->start_time != AV_NOPTS_VALUE)
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c->cur_pts -= av_rescale_q(st->start_time, st->time_base, AV_TIME_BASE_Q);
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c->cur_frame_duration = av_rescale_q(pkt.duration, st->time_base, AV_TIME_BASE_Q);
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#if 0
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printf("index=%d pts=%0.3f duration=%0.6f\n",
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pkt.stream_index,
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(double)c->cur_pts /
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AV_TIME_BASE,
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(double)c->cur_frame_duration /
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AV_TIME_BASE);
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printf("index=%d pts=%0.3f duration=%0.6f\n",
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pkt.stream_index,
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(double)c->cur_pts /
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AV_TIME_BASE,
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(double)c->cur_frame_duration /
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AV_TIME_BASE);
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#endif
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/* find RTP context */
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c->packet_stream_index = pkt.stream_index;
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ctx = c->rtp_ctx[c->packet_stream_index];
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if(!ctx) {
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av_free_packet(&pkt);
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break;
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}
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codec = ctx->streams[0]->codec;
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/* only one stream per RTP connection */
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pkt.stream_index = 0;
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} else {
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ctx = &c->fmt_ctx;
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/* Fudge here */
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codec = ctx->streams[pkt.stream_index]->codec;
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}
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if (c->is_packetized) {
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int max_packet_size;
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if (c->rtp_protocol == RTSP_PROTOCOL_RTP_TCP)
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max_packet_size = RTSP_TCP_MAX_PACKET_SIZE;
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else
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max_packet_size = url_get_max_packet_size(c->rtp_handles[c->packet_stream_index]);
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ret = url_open_dyn_packet_buf(&ctx->pb, max_packet_size);
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} else {
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ret = url_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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}
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if (pkt.dts != AV_NOPTS_VALUE)
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pkt.dts = av_rescale_q(pkt.dts,
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c->fmt_in->streams[pkt.stream_index]->time_base,
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ctx->streams[pkt.stream_index]->time_base);
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if (pkt.pts != AV_NOPTS_VALUE)
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pkt.pts = av_rescale_q(pkt.pts,
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c->fmt_in->streams[pkt.stream_index]->time_base,
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ctx->streams[pkt.stream_index]->time_base);
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if (av_write_frame(ctx, &pkt))
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c->state = HTTPSTATE_SEND_DATA_TRAILER;
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len = url_close_dyn_buf(ctx->pb, &c->pb_buffer);
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c->cur_frame_bytes = len;
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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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codec->frame_number++;
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if (len == 0) {
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/* find RTP context */
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c->packet_stream_index = pkt.stream_index;
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ctx = c->rtp_ctx[c->packet_stream_index];
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if(!ctx) {
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av_free_packet(&pkt);
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goto redo;
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break;
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}
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codec = ctx->streams[0]->codec;
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/* only one stream per RTP connection */
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pkt.stream_index = 0;
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} else {
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ctx = &c->fmt_ctx;
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/* Fudge here */
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codec = ctx->streams[pkt.stream_index]->codec;
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}
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if (c->is_packetized) {
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int max_packet_size;
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if (c->rtp_protocol == RTSP_PROTOCOL_RTP_TCP)
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max_packet_size = RTSP_TCP_MAX_PACKET_SIZE;
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else
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max_packet_size = url_get_max_packet_size(c->rtp_handles[c->packet_stream_index]);
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ret = url_open_dyn_packet_buf(&ctx->pb, max_packet_size);
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} else {
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ret = url_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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}
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if (pkt.dts != AV_NOPTS_VALUE)
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pkt.dts = av_rescale_q(pkt.dts,
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c->fmt_in->streams[pkt.stream_index]->time_base,
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ctx->streams[pkt.stream_index]->time_base);
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if (pkt.pts != AV_NOPTS_VALUE)
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pkt.pts = av_rescale_q(pkt.pts,
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c->fmt_in->streams[pkt.stream_index]->time_base,
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ctx->streams[pkt.stream_index]->time_base);
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if (av_write_frame(ctx, &pkt))
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c->state = HTTPSTATE_SEND_DATA_TRAILER;
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len = url_close_dyn_buf(ctx->pb, &c->pb_buffer);
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c->cur_frame_bytes = len;
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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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codec->frame_number++;
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if (len == 0) {
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av_free_packet(&pkt);
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goto redo;
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}
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av_free_packet(&pkt);
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}
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av_free_packet(&pkt);
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}
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}
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break;
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default:
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case HTTPSTATE_SEND_DATA_TRAILER:
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