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>

.github/workflows/test.yml

@@ -81,8 +81,10 @@ jobs:
         run: |
           WORKDIR=$(cygpath -u $SRS_WORKSPACE) && export PATH=/usr/bin:/usr/local/bin && cd ${WORKDIR} &&
           pwd && rm -rf /usr/local/srs-cache && mkdir -p /usr/local/srs-cache/srs/trunk && ls -lh &&
-          tar xf objs.tar.bz2 -C /usr/local/srs-cache/srs/trunk/ && du -sh /usr/local/srs-cache/srs/trunk/* && 
-          cd ${WORKDIR}/trunk && ./configure --h265=on --gb28181=on --utest=on && ls -lh && du -sh * && du -sh objs/* &&
+          tar xf objs.tar.bz2 -C /usr/local/srs-cache/srs/trunk/ && 
+          du -sh /usr/local/srs-cache/srs/trunk/* && du -sh /usr/local/srs-cache/srs/trunk/objs/* && 
+          cd ${WORKDIR}/trunk && ./configure --h265=on --gb28181=on --utest=on && 
+          ls -lh && du -sh * && du -sh objs/* &&
           cd ${WORKDIR}/trunk && make utest && ./objs/srs_utest
     runs-on: windows-latest
 

.vscode/README.md

@@ -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:

README.md

@@ -66,6 +66,10 @@ To learn more about RTMP, HLS, HTTP-FLV, SRT, MPEG-DASH, WebRTC protocols, clust
 HTTP API, DVR, and transcoding, please check the documents in [English](https://ossrs.io) 
 or [Chinese](https://ossrs.net).
 
+If you want to use an IDE, VSCode is recommanded. VSCode supports macOS, Linux, and Windows 
+platforms. The settings are ready. All you need to do is open the folder with VSCode and 
+enjoy the efficiency brought by the IDE. See [VSCode README](.vscode/README.md) for details.
+
 ## Sponsor
 
 Would you like additional assistance from us? By becoming a sponsor or backer of SRS, we can provide you 

trunk/3rdparty/srs-bench/README.md

@@ -14,7 +14,7 @@ git clone -b feature/rtc https://github.com/ossrs/srs-bench.git &&
 cd srs-bench && make
 ```
 
-> Note: 依赖Go编译工具，建议使用 Go 1.17 及以上的版本。
+> Note: 依赖Go编译工具，建议使用 Go 1.21 及以上的版本。
 
 编译会生成下面的工具：
 

trunk/3rdparty/srs-bench/srs/rtc_test.go

@@ -2418,7 +2418,7 @@ func TestRtcPublish_HttpFlvPlay(t *testing.T) {
 				}
 				logger.Tf(ctx, "got %v tag, %v %vms %vB", nnVideo+nnAudio, tagType, timestamp, len(tag))
 
-				if audioPacketsOK, videoPacketsOK := !hasAudio || nnAudio >= 10, !hasVideo || nnVideo >= 10; audioPacketsOK && videoPacketsOK {
+				if audioPacketsOK, videoPacketsOK := hasAudio && nnAudio >= 10, hasVideo && nnVideo >= 10; audioPacketsOK && videoPacketsOK {
 					logger.Tf(ctx, "Flv recv %v/%v audio, %v/%v video", hasAudio, nnAudio, hasVideo, nnVideo)
 					cancel()
 				}

trunk/conf/regression-test-for-clion.conf

@@ -50,6 +50,8 @@ vhost __defaultVhost__ {
     }
     http_remux {
         enabled     on;
+        # Disabling 'guess_has_av' as it is not required for the current test setup.
+        guess_has_av    off;
         mount       [vhost]/[app]/[stream].flv;
         drop_if_not_match on;
     }

trunk/conf/regression-test.conf

@@ -50,8 +50,10 @@ vhost __defaultVhost__ {
         atc         on;
     }
     http_remux {
-        enabled     on;
-        mount       [vhost]/[app]/[stream].flv;
+        enabled         on;
+        # Disabling 'guess_has_av' as it is not required for the current test setup.
+        guess_has_av    off;
+        mount           [vhost]/[app]/[stream].flv;
     }
     ingest livestream {
         enabled on;

trunk/doc/CHANGELOG.md

@@ -7,6 +7,7 @@ The changelog for SRS.
 <a name="v7-changes"></a>
 
 ## SRS 7.0 Changelog
+* v7.0, 2025-05-29, Merge [#4363](https://github.com/ossrs/srs/pull/4363): Fix error about TestRtcPublish_HttpFlvPlay. v7.0.36 (#4363)
 * v7.0, 2025-05-29, Merge [#4362](https://github.com/ossrs/srs/pull/4362): Update VSCode launch configuration to support GDB on Linux and LLDB on macOS. v7.0.35 (#4362)
 * v7.0, 2025-05-26, Merge [#4359](https://github.com/ossrs/srs/pull/4359): update pion/webrtc to v4. v7.0.34 (#4359)
 * v7.0, 2025-05-13, Merge [#4289](https://github.com/ossrs/srs/pull/4289): rtmp2rtc: Support RTMP-to-WebRTC conversion with HEVC. v7.0.33 (#4289)

trunk/src/core/srs_core_version7.hpp

@@ -9,6 +9,6 @@
 
 #define VERSION_MAJOR       7
 #define VERSION_MINOR       0
-#define VERSION_REVISION    35
+#define VERSION_REVISION    36
 
 #endif