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Fix error about TestRtcPublish_HttpFlvPlay. v7.0.36 (#4363)
In the scenario of converting WebRTC to RTMP, this conversion will not
proceed until an SenderReport is received; for reference, see:
https://github.com/ossrs/srs/pull/2470.
Thus, if HTTP-FLV streaming is attempted before the SR is received, the
FLV Header will contain only audio, devoid of video content.
This error can be resolved by disabling `guess_has_av` in the
configuration file, since we can guarantee that both audio and video are
present in the test cases.
However, in the original regression tests, the
`TestRtcPublish_HttpFlvPlay` test case contains a bug:
5a404c089b/trunk/3rdparty/srs-bench/srs/rtc_test.go (L2421-L2424)
The test would pass when `hasAudio` is true and `hasVideo` is false,
which is actually incorrect. Therefore, it has been revised so that the
test now only passes if both values are true.
---------
Co-authored-by: Copilot <175728472+Copilot@users.noreply.github.com>
Co-authored-by: winlin <winlinvip@gmail.com>
This commit is contained in:
6
.github/workflows/test.yml
vendored
6
.github/workflows/test.yml
vendored
@@ -81,8 +81,10 @@ jobs:
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run: |
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WORKDIR=$(cygpath -u $SRS_WORKSPACE) && export PATH=/usr/bin:/usr/local/bin && cd ${WORKDIR} &&
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pwd && rm -rf /usr/local/srs-cache && mkdir -p /usr/local/srs-cache/srs/trunk && ls -lh &&
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tar xf objs.tar.bz2 -C /usr/local/srs-cache/srs/trunk/ && du -sh /usr/local/srs-cache/srs/trunk/* &&
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cd ${WORKDIR}/trunk && ./configure --h265=on --gb28181=on --utest=on && ls -lh && du -sh * && du -sh objs/* &&
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tar xf objs.tar.bz2 -C /usr/local/srs-cache/srs/trunk/ &&
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du -sh /usr/local/srs-cache/srs/trunk/* && du -sh /usr/local/srs-cache/srs/trunk/objs/* &&
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cd ${WORKDIR}/trunk && ./configure --h265=on --gb28181=on --utest=on &&
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ls -lh && du -sh * && du -sh objs/* &&
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cd ${WORKDIR}/trunk && make utest && ./objs/srs_utest
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runs-on: windows-latest
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