openharmony/third_party_ffmpeg
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Revert "avserver: fix build after the next bump."

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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:
Anton Khirnov 2012-01-21 08:40:47 +01:00
parent d2a0041c20
commit 97e3f94b61
1 changed files with 10 additions and 12 deletions
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22
avserver.c
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@ -26,16 +26,13 @@
#include <string.h>
#include <stdlib.h>
#include "libavformat/avformat.h"
// FIXME those are internal headers, avserver _really_ shouldn't use them
#include "libavformat/ffm.h"
#include "libavformat/network.h"
#include "libavformat/os_support.h"
#include "libavformat/rtpdec.h"
#include "libavformat/rtsp.h"
// XXX for ffio_open_dyn_packet_buffer, to be removed
#include "libavformat/avio_internal.h"
#include "libavformat/internal.h"
#include "libavformat/url.h"
#include "libavutil/avstring.h"
#include "libavutil/lfg.h"
#include "libavutil/dict.h"
@ -870,7 +867,7 @@ static void close_connection(HTTPContext *c)
}
h = c->rtp_handles[i];
if (h)
ffurl_close(h);
url_close(h);
}
ctx = &c->fmt_ctx;
@ -2251,6 +2248,7 @@ static int http_prepare_data(HTTPContext *c)
* Default value from Libav
* Try to set it use configuration option
*/
c->fmt_ctx.preload = (int)(0.5*AV_TIME_BASE);
c->fmt_ctx.max_delay = (int)(0.7*AV_TIME_BASE);
if (avformat_write_header(&c->fmt_ctx, NULL) < 0) {
@ -2369,7 +2367,7 @@ static int http_prepare_data(HTTPContext *c)
if (c->rtp_protocol == RTSP_LOWER_TRANSPORT_TCP)
max_packet_size = RTSP_TCP_MAX_PACKET_SIZE;
else
max_packet_size = c->rtp_handles[c->packet_stream_index]->max_packet_size;
max_packet_size = url_get_max_packet_size(c->rtp_handles[c->packet_stream_index]);
ret = ffio_open_dyn_packet_buf(&ctx->pb, max_packet_size);
} else {
ret = avio_open_dyn_buf(&ctx->pb);
@ -2522,8 +2520,8 @@ static int http_send_data(HTTPContext *c)
} else {
/* send RTP packet directly in UDP */
c->buffer_ptr += 4;
ffurl_write(c->rtp_handles[c->packet_stream_index],
c->buffer_ptr, len);
url_write(c->rtp_handles[c->packet_stream_index],
c->buffer_ptr, len);
c->buffer_ptr += len;
/* here we continue as we can send several packets per 10 ms slot */
}
@ -3406,10 +3404,10 @@ static int rtp_new_av_stream(HTTPContext *c,
"rtp://%s:%d", ipaddr, ntohs(dest_addr->sin_port));
}
if (ffurl_open(&h, ctx->filename, AVIO_FLAG_WRITE, NULL, NULL) < 0)
if (url_open(&h, ctx->filename, AVIO_FLAG_WRITE) < 0)
goto fail;
c->rtp_handles[stream_index] = h;
max_packet_size = h->max_packet_size;
max_packet_size = url_get_max_packet_size(h);
break;
case RTSP_LOWER_TRANSPORT_TCP:
/* RTP/TCP case */
@ -3432,7 +3430,7 @@ static int rtp_new_av_stream(HTTPContext *c,
if (avformat_write_header(ctx, NULL) < 0) {
fail:
if (h)
ffurl_close(h);
url_close(h);
av_free(ctx);
return -1;
}
@ -3469,7 +3467,7 @@ static AVStream *add_av_stream1(FFStream *stream, AVCodecContext *codec, int cop
}
fst->priv_data = av_mallocz(sizeof(FeedData));
fst->index = stream->nb_streams;
avpriv_set_pts_info(fst, 33, 1, 90000);
av_set_pts_info(fst, 33, 1, 90000);
fst->sample_aspect_ratio = codec->sample_aspect_ratio;
stream->streams[stream->nb_streams++] = fst;
return fst;