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https://github.com/nyanmisaka/ffmpeg-rockchip.git
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Merge remote branch 'qatar/master'
* qatar/master: fate: fix partial run when no samples path is specified ARM: NEON fixed-point forward MDCT ARM: NEON fixed-point FFT lavf: bump minor version and add an APIChanges entry for avio changes avio: simplify url_open_dyn_buf_internal by using avio_alloc_context() avio: make url_fdopen internal. avio: make url_open_dyn_packet_buf internal. avio: avio_ prefix for url_close_dyn_buf avio: avio_ prefix for url_open_dyn_buf avio: introduce an AVIOContext.seekable field ac3enc: use generic fixed-point mdct lavfi: add fade filter Change yadif to not use out of picture lines. lavc: deprecate AVCodecContext.antialias_algo lavc: mark mb_qmin/mb_qmax for removal on next major bump. Conflicts: doc/filters.texi libavcodec/ac3enc_fixed.h libavcodec/ac3enc_float.h libavfilter/Makefile libavfilter/allfilters.c libavfilter/vf_fade.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
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@@ -41,7 +41,7 @@
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buffer to 'rtp_write_packet' contains all the packets for ONE
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frame. Each packet should have a four byte header containing
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the length in big endian format (same trick as
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'url_open_dyn_packet_buf')
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'ffio_open_dyn_packet_buf')
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*/
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static RTPDynamicProtocolHandler ff_realmedia_mp3_dynamic_handler = {
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@@ -264,7 +264,7 @@ int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
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return -1;
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s->last_octet_count = s->octet_count;
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if (url_open_dyn_buf(&pb) < 0)
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if (avio_open_dyn_buf(&pb) < 0)
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return -1;
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// Receiver Report
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@@ -321,7 +321,7 @@ int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
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}
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avio_flush(pb);
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len = url_close_dyn_buf(pb, &buf);
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len = avio_close_dyn_buf(pb, &buf);
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if ((len > 0) && buf) {
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int result;
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av_dlog(s->ic, "sending %d bytes of RR\n", len);
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@@ -339,7 +339,7 @@ void rtp_send_punch_packets(URLContext* rtp_handle)
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int len;
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/* Send a small RTP packet */
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if (url_open_dyn_buf(&pb) < 0)
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if (avio_open_dyn_buf(&pb) < 0)
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return;
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avio_w8(pb, (RTP_VERSION << 6));
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@@ -349,13 +349,13 @@ void rtp_send_punch_packets(URLContext* rtp_handle)
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avio_wb32(pb, 0); /* SSRC */
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avio_flush(pb);
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len = url_close_dyn_buf(pb, &buf);
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len = avio_close_dyn_buf(pb, &buf);
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if ((len > 0) && buf)
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url_write(rtp_handle, buf, len);
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av_free(buf);
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/* Send a minimal RTCP RR */
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if (url_open_dyn_buf(&pb) < 0)
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if (avio_open_dyn_buf(&pb) < 0)
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return;
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avio_w8(pb, (RTP_VERSION << 6));
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@@ -364,7 +364,7 @@ void rtp_send_punch_packets(URLContext* rtp_handle)
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avio_wb32(pb, 0); /* our own SSRC */
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avio_flush(pb);
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len = url_close_dyn_buf(pb, &buf);
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len = avio_close_dyn_buf(pb, &buf);
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if ((len > 0) && buf)
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url_write(rtp_handle, buf, len);
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av_free(buf);
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