mirror of
https://github.com/librempeg/librempeg
synced 2024-11-23 11:39:47 +00:00
Revert "avserver: fix build after the next bump."
This temporarily (until 0.8 is released) reverts commit
8e1340abc3
. That commit breaks shared
builds because of symbol hiding. Reverting it will enable shared builds
for 0.8
This commit is contained in:
parent
d2a0041c20
commit
97e3f94b61
22
avserver.c
22
avserver.c
@ -26,16 +26,13 @@
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#include <string.h>
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#include <stdlib.h>
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#include "libavformat/avformat.h"
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// FIXME those are internal headers, avserver _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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#include "libavformat/rtpdec.h"
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#include "libavformat/rtsp.h"
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// XXX for ffio_open_dyn_packet_buffer, to be removed
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#include "libavformat/avio_internal.h"
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#include "libavformat/internal.h"
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#include "libavformat/url.h"
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#include "libavutil/avstring.h"
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#include "libavutil/lfg.h"
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#include "libavutil/dict.h"
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@ -870,7 +867,7 @@ static void close_connection(HTTPContext *c)
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}
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h = c->rtp_handles[i];
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if (h)
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ffurl_close(h);
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url_close(h);
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}
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ctx = &c->fmt_ctx;
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@ -2251,6 +2248,7 @@ static int http_prepare_data(HTTPContext *c)
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* Default value from Libav
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* Try to set it use configuration option
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*/
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c->fmt_ctx.preload = (int)(0.5*AV_TIME_BASE);
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c->fmt_ctx.max_delay = (int)(0.7*AV_TIME_BASE);
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if (avformat_write_header(&c->fmt_ctx, NULL) < 0) {
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@ -2369,7 +2367,7 @@ static int http_prepare_data(HTTPContext *c)
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if (c->rtp_protocol == RTSP_LOWER_TRANSPORT_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 = c->rtp_handles[c->packet_stream_index]->max_packet_size;
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max_packet_size = url_get_max_packet_size(c->rtp_handles[c->packet_stream_index]);
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ret = ffio_open_dyn_packet_buf(&ctx->pb, max_packet_size);
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} else {
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ret = avio_open_dyn_buf(&ctx->pb);
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@ -2522,8 +2520,8 @@ static int http_send_data(HTTPContext *c)
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} else {
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/* send RTP packet directly in UDP */
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c->buffer_ptr += 4;
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ffurl_write(c->rtp_handles[c->packet_stream_index],
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c->buffer_ptr, len);
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url_write(c->rtp_handles[c->packet_stream_index],
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c->buffer_ptr, len);
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c->buffer_ptr += len;
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/* here we continue as we can send several packets per 10 ms slot */
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}
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@ -3406,10 +3404,10 @@ static int rtp_new_av_stream(HTTPContext *c,
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"rtp://%s:%d", ipaddr, ntohs(dest_addr->sin_port));
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}
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if (ffurl_open(&h, ctx->filename, AVIO_FLAG_WRITE, NULL, NULL) < 0)
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if (url_open(&h, ctx->filename, AVIO_FLAG_WRITE) < 0)
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goto fail;
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c->rtp_handles[stream_index] = h;
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max_packet_size = h->max_packet_size;
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max_packet_size = url_get_max_packet_size(h);
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break;
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case RTSP_LOWER_TRANSPORT_TCP:
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/* RTP/TCP case */
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@ -3432,7 +3430,7 @@ static int rtp_new_av_stream(HTTPContext *c,
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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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ffurl_close(h);
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url_close(h);
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av_free(ctx);
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return -1;
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}
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@ -3469,7 +3467,7 @@ static AVStream *add_av_stream1(FFStream *stream, AVCodecContext *codec, int cop
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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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av_set_pts_info(fst, 33, 1, 90000);
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fst->sample_aspect_ratio = codec->sample_aspect_ratio;
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stream->streams[stream->nb_streams++] = fst;
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return fst;
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