diff --git a/ffserver.c b/ffserver.c index 04e0d2a782..bf7b5caf7d 100644 --- a/ffserver.c +++ b/ffserver.c @@ -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;istream->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;istream->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;istream->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: