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:
Haibo Chen(陈海博)
2025-05-29 23:21:15 +08:00
committed by winlin
parent 9c559dcb48
commit 33b0a0fe7d
9 changed files with 32 additions and 7 deletions

14
.vscode/README.md vendored
View File

@@ -67,6 +67,20 @@ Then you will discover all the unit testcases from the `View > Testing` panel. Y
open utest source file like `trunk/src/utest/srs_utest.cpp`, then click the `Run Test` or `Debug Test`
on each testcase such as `FastSampleInt64Test`.
## macOS: SRS Regression Test
Follow the [srs-bench](../trunk/3rdparty/srs-bench/README.md) to setup the environment.
Open the test panel by clicking `View > Testing`, run the regression tests under:
```
+ Go
+ github.com/ossrs/srs-bench
+ blackbox
+ gb28181
+ srs
```
## macOS: Proxy
Install the following extensions: