Codex: Fix potential issues with memory leak.

AGENTS.md

@@ -3,29 +3,38 @@
 This is the guide for the OpenAI Codex agent.
 
 ## General repository layout
-- Source code: trunk/src/ contains the C++ code for the SRS server.
-  - trunk/src/core and trunk/src/kernel contain the common definitions.
-  - trunk/src/protocol contains the media streaming protocol implementations.
-  - trunk/src/app contains the application-level implementations.
-  - trunk/src/main contains the main entry points for the programs.
-- Source code: proxy/ contains the proxy server in Go for RTMP/SRT/WebRTC proxying.
+- Source code: 
+  - `trunk/src/` contains the major C++ code for the SRS server.
+  - `trunk/src/core` and trunk/src/kernel contain the common definitions.
+  - `trunk/src/protocol` contains the media streaming protocol implementations.
+  - `trunk/src/app` contains the application-level implementations.
+  - `trunk/src/main` contains the main entry points for the programs.
+  - `proxy/` contains the proxy server in Go for RTMP/SRT/WebRTC proxying.
+  - `trunk/src/core/srs_core_autofree.hpp` defines the smart pointer; you should review it before fixing memory issues.
 - Configuration:
-  - trunk/conf/full.conf contains all supported configurations.
-  - trunk/src/app/srs_app_config.cpp parses and checks the configuration file.
-  - trunk/conf/*.conf contains other example configuration files.
-- UTests: trunk/src/utest contains the unit tests using gtest.
-- Third-party dependencies: trunk/3rdparty contains the dependency libraries. You may refer to these codes, but never attempt to change or update them.
-  - trunk/3rdparty/st-srs is the coroutine library (state-threads).
-  - trunk/3rdparty/srs-bench is the integration test tool for blackbox tests.
-  - trunk/3rdparty/gtest-fit is the gtest framework used for unit tests.
-  - trunk/3rdparty/ffmpeg-4-fit is the codec library for transcoding audio streams such as AAC with Opus.
-  - trunk/3rdparty/openssl-1.1-fit is used for RTMP handshake and WebRTC DTLS handshake.
-  - trunk/3rdparty/libsrtp-2-fit is used for SRTP in WebRTC.
-  - trunk/3rdparty/srt-1-fit is the SRT implementation.
+  - `trunk/conf/full.conf` contains all supported configurations.
+  - `trunk/src/app/srs_app_config.cpp` parses and checks the configuration file.
+  - `trunk/conf/*.conf` contains other example configuration files.
+- Tests: 
+  - `trunk/src/utest` contains the unit tests using gtest.
+  - `trunk/3rdparty/srs-bench` is the integration test tool for blackbox tests.
+- Third-party dependencies: `trunk/3rdparty` contains the dependency libraries. You may refer to these codes, but never attempt to change or update.
+  - `trunk/3rdparty/st-srs` is the coroutine library (state-threads).
+  - `trunk/3rdparty/gtest-fit` is the gtest framework used for unit tests.
+  - `trunk/3rdparty/ffmpeg-4-fit` is the codec library for transcoding audio streams such as AAC with Opus.
+  - `trunk/3rdparty/openssl-1.1-fit` is used for RTMP handshake and WebRTC DTLS handshake.
+  - `trunk/3rdparty/libsrtp-2-fit` is used for SRTP in WebRTC.
+  - `trunk/3rdparty/srt-1-fit` is the SRT implementation.
 - Documentation:
-  - README.md is a brief introduction to this project.
-  - trunk/doc/CHANGELOG.md contains the changelog history.
+  - `README.md` is a brief introduction to this project.
+  - `trunk/doc/CHANGELOG.md` contains the changelog history.
+  - `trunk/doc/Dockers.md` is the guide for building Docker images.
+- Build:
+  - `trunk/configure` is the script used to configure the project; after running it, you can use make to build.
+  - `Dockerfile` is the main Docker file for building the image.
+  - `trunk/Dockerfile.test` is used to build the image for testing.
+  - `trunk/Dockerfile.builds` is used to verify builds on different target platforms.
 
 ## Testing Instructions
-- Run CI tests defined in .github/workflows/test.yml file.
+- Run CI tests defined in `.github/workflows/test.yml` file.
 - Add or update tests for the code you change, even if nobody asked.

trunk/src/app/srs_app_caster_flv.cpp

@@ -95,6 +95,12 @@ SrsAppCasterFlv::~SrsAppCasterFlv()
 {
     srs_freep(http_mux);
     srs_freep(manager);
+
+    std::vector<ISrsConnection*>::iterator it;
+    for (it = conns.begin(); it != conns.end(); ++it) {
+        ISrsConnection* conn = *it;
+        srs_freep(conn);
+    }
 }
 
 srs_error_t SrsAppCasterFlv::initialize(SrsConfDirective* c)

trunk/src/app/srs_app_server.cpp

@@ -1275,6 +1275,7 @@ srs_error_t SrsServer::do_on_tcp_client(ISrsListener* listener, srs_netfd_t& stf
     conn_manager->add(resource);
 
     // If connection is a resource to start, start a coroutine to handle it.
+    // Note that conn is managed by conn_manager, so we don't need to free it.
     ISrsStartable* conn = dynamic_cast<ISrsStartable*>(resource);
     srs_assert(conn);
     if ((err = conn->start()) != srs_success) {

trunk/src/app/srs_app_srt_server.cpp

@@ -236,6 +236,7 @@ srs_error_t SrsSrtServer::accept_srt_client(srs_srt_t srt_fd)
     // directly enqueue, the cycle thread will remove the client.
     conn_manager_->add(resource);
 
+    // Note that conn is managed by conn_manager, so we don't need to free it.
     ISrsStartable* conn = dynamic_cast<ISrsStartable*>(resource);
     if ((err = conn->start()) != srs_success) {
         return srs_error_wrap(err, "start srt conn coroutine");