AI: Add utests to cover app rtc module. (#4498)

Co-authored-by: OSSRS-AI <winlinam@gmail.com>

trunk/configure

@@ -237,8 +237,8 @@ fi
 MODULE_ID="CORE"
 MODULE_DEPENDS=()
 ModuleLibIncs=(${SRS_OBJS})
-MODULE_FILES=("srs_core" "srs_core_version" "srs_core_version7" "srs_core_autofree" "srs_core_performance"
+MODULE_FILES=("srs_core" "srs_core_version" "srs_core_version7" "srs_core_autofree"
-		"srs_core_time" "srs_core_platform" "srs_core_deprecated")
+		"srs_core_time" "srs_core_platform" "srs_core_deprecated" "srs_core_performance")
 CORE_INCS="src/core"; MODULE_DIR=${CORE_INCS} . $SRS_WORKDIR/auto/modules.sh
 CORE_OBJS="${MODULE_OBJS[@]}"
 #
@@ -248,10 +248,10 @@ MODULE_DEPENDS=("CORE")
 ModuleLibIncs=(${SRS_OBJS} ${LibSSLRoot})
 MODULE_FILES=("srs_kernel_error" "srs_kernel_log" "srs_kernel_buffer"
         "srs_kernel_utility" "srs_kernel_flv" "srs_kernel_codec" "srs_kernel_io"
-        "srs_kernel_consts" "srs_kernel_aac" "srs_kernel_mp3" "srs_kernel_ts" "srs_kernel_ps"
+        "srs_kernel_consts" "srs_kernel_aac" "srs_kernel_mp3" "srs_kernel_ts"
         "srs_kernel_stream" "srs_kernel_balance" "srs_kernel_mp4" "srs_kernel_file"
         "srs_kernel_kbps" "srs_kernel_rtc_rtp" "srs_kernel_rtc_rtcp" "srs_kernel_packet"
-        "srs_kernel_st" "srs_kernel_factory" "srs_kernel_hourglass"
+        "srs_kernel_st" "srs_kernel_factory" "srs_kernel_hourglass" "srs_kernel_ps"
         "srs_kernel_pithy_print" "srs_kernel_rtc_queue" "srs_kernel_resource")
 KERNEL_INCS="src/kernel"; MODULE_DIR=${KERNEL_INCS} . $SRS_WORKDIR/auto/modules.sh
 KERNEL_OBJS="${MODULE_OBJS[@]}"
@@ -261,8 +261,8 @@ MODULE_ID="PROTOCOL"
 MODULE_DEPENDS=("CORE" "KERNEL") 
 ModuleLibIncs=(${SRS_OBJS} ${LibSTRoot} ${LibSSLRoot})
 MODULE_FILES=("srs_protocol_amf0" "srs_protocol_io" "srs_protocol_conn" "srs_protocol_rtmp_handshake"
-        "srs_protocol_rtmp_stack" "srs_protocol_utility" "srs_protocol_rtmp_msg_array" "srs_protocol_stream"
+        "srs_protocol_rtmp_stack" "srs_protocol_utility" "srs_protocol_rtmp_msg_array"
-        "srs_protocol_raw_avc" "srs_protocol_http_stack" "srs_protocol_json"
+        "srs_protocol_raw_avc" "srs_protocol_http_stack" "srs_protocol_json" "srs_protocol_stream"
         "srs_protocol_format" "srs_protocol_log" "srs_protocol_st" "srs_protocol_http_client"
         "srs_protocol_http_conn" "srs_protocol_rtmp_conn" "srs_protocol_protobuf"
         "srs_protocol_http_stack_llhttp" "srs_protocol_http_stack_llhttpapi" 
@@ -289,17 +289,17 @@ if [[ $SRS_FFMPEG_FIT == YES ]]; then
     ModuleLibIncs+=("${LibFfmpegRoot[*]}")
 fi
 MODULE_FILES=("srs_app_server" "srs_app_rtmp_conn" "srs_app_rtmp_source"
-        "srs_app_refer" "srs_app_hls" "srs_app_forward" "srs_app_encoder" "srs_app_http_stream"
+        "srs_app_refer" "srs_app_hls" "srs_app_forward" "srs_app_encoder"
         "srs_app_st" "srs_app_log" "srs_app_config" "srs_app_stream_bridge"
         "srs_app_reload" "srs_app_http_api" "srs_app_http_conn" "srs_app_http_hooks"
         "srs_app_ingest" "srs_app_ffmpeg" "srs_app_utility" "srs_app_edge"
         "srs_app_heartbeat" "srs_app_http_client" "srs_app_http_static"
         "srs_app_recv_thread" "srs_app_security" "srs_app_statistic" "srs_app_hds"
         "srs_app_mpegts_udp" "srs_app_listener" "srs_app_async_call"
-        "srs_app_caster_flv" "srs_app_latest_version" "srs_app_process" "srs_app_ng_exec"
+        "srs_app_caster_flv" "srs_app_latest_version" "srs_app_process"
-        "srs_app_dash" "srs_app_fragment" "srs_app_dvr"
+        "srs_app_dash" "srs_app_fragment" "srs_app_dvr" "srs_app_ng_exec"
         "srs_app_coworkers" "srs_app_circuit_breaker" "srs_app_factory"
-        "srs_app_stream_token")
+        "srs_app_stream_token" "srs_app_http_stream")
 # Always include SRT app modules
 MODULE_FILES+=("srs_app_srt_server" "srs_app_srt_listener" "srs_app_srt_conn" "srs_app_srt_source")
 MODULE_FILES+=("srs_app_rtc_conn" "srs_app_rtc_dtls" "srs_app_rtc_network"
@@ -378,10 +378,11 @@ if [[ $SRS_UTEST == YES ]]; then
     MODULE_FILES=("srs_utest" "srs_utest_amf0" "srs_utest_kernel" "srs_utest_core"
         "srs_utest_config" "srs_utest_rtmp" "srs_utest_http" "srs_utest_avc" "srs_utest_reload"
         "srs_utest_mp4" "srs_utest_service" "srs_utest_app" "srs_utest_app2" "srs_utest_rtc" "srs_utest_config2"
-        "srs_utest_config3" "srs_utest_config4" "srs_utest_protocol" "srs_utest_protocol2" "srs_utest_kernel2" "srs_utest_protocol3"
+        "srs_utest_config3" "srs_utest_config4" "srs_utest_protocol" "srs_utest_protocol2" "srs_utest_kernel2"
         "srs_utest_st" "srs_utest_rtc2" "srs_utest_rtc3" "srs_utest_fmp4" "srs_utest_source_lock"
         "srs_utest_stream_token" "srs_utest_rtc_recv_track" "srs_utest_st2" "srs_utest_hevc_structs"
-        "srs_utest_coworkers" "srs_utest_pithy_print" "srs_utest_kernel3" "srs_utest_protocol4")
+        "srs_utest_coworkers" "srs_utest_pithy_print" "srs_utest_kernel3" "srs_utest_protocol4" 
+        "srs_utest_protocol3")
     # Always include SRT utest
     MODULE_FILES+=("srs_utest_srt")
     if [[ $SRS_GB28181 == YES ]]; then

trunk/src/app/srs_app_rtc_source.cpp

@@ -400,6 +400,7 @@ SrsRtcSource::SrsRtcSource()
 #ifdef SRS_FFMPEG_FIT
     frame_builder_ = NULL;
 #endif
+    circuit_breaker_ = _srs_circuit_breaker;
 
     pli_for_rtmp_ = pli_elapsed_ = 0;
     stream_die_at_ = 0;
@@ -793,7 +794,7 @@ srs_error_t SrsRtcSource::on_rtp(SrsRtpPacket *pkt)
     srs_error_t err = srs_success;
 
     // If circuit-breaker is dying, drop packet.
-    if (_srs_circuit_breaker->hybrid_dying_water_level()) {
+    if (circuit_breaker_ && circuit_breaker_->hybrid_dying_water_level()) {
         _srs_pps_aloss2->sugar_ += (int64_t)consumers_.size();
         return err;
     }

trunk/src/app/srs_app_rtc_source.hpp

@@ -44,6 +44,7 @@ class SrsRtcFrameBuilder;
 class SrsLiveSource;
 class SrsRtpVideoBuilder;
 class ISrsRtcConsumer;
+class ISrsCircuitBreaker;
 
 // Firefox defaults as 109, Chrome is 111.
 const int kAudioPayloadType = 111;
@@ -219,6 +220,8 @@ public:
 class SrsRtcSource : public ISrsFastTimer, public ISrsRtcSourceForConsumer
 {
 private:
+    // Circuit breaker for protecting server resources.
+    ISrsCircuitBreaker *circuit_breaker_;
     // For publish, it's the publish client id.
     // For edge, it's the edge ingest id.
     // when source id changed, for example, the edge reconnect,

trunk/src/utest/srs_utest.cpp

@@ -329,8 +329,23 @@ srs_error_t SrsHttpTestServer::start()
 {
     srs_error_t err = srs_success;
 
-    if ((err = srs_tcp_listen(ip_, port_, &fd_)) != srs_success) {
+    // Retry up to 3 times with different random ports if listen fails
-        return srs_error_wrap(err, "listen %s:%d", ip_.c_str(), port_);
+    for (int retry = 0; retry < 3; retry++) {
+        if ((err = srs_tcp_listen(ip_, port_, &fd_)) == srs_success) {
+            break;
+        }
+
+        // If this is not the last retry, generate a new random port and try again
+        if (retry < 2) {
+            srs_freep(err);
+            port_ = srs_rand_integer(30000, 60000);
+            srs_trace("HTTP test server listen failed on %s:%d, retry %d with new port %d",
+                      ip_.c_str(), port_, retry + 1, port_);
+        }
+    }
+
+    if (err != srs_success) {
+        return srs_error_wrap(err, "listen %s:%d after 3 retries", ip_.c_str(), port_);
     }
 
     return trd_->start();
@@ -418,8 +433,23 @@ srs_error_t SrsHttpsTestServer::start()
 {
     srs_error_t err = srs_success;
 
-    if ((err = srs_tcp_listen(ip_, port_, &fd_)) != srs_success) {
+    // Retry up to 3 times with different random ports if listen fails
-        return srs_error_wrap(err, "listen %s:%d", ip_.c_str(), port_);
+    for (int retry = 0; retry < 3; retry++) {
+        if ((err = srs_tcp_listen(ip_, port_, &fd_)) == srs_success) {
+            break;
+        }
+
+        // If this is not the last retry, generate a new random port and try again
+        if (retry < 2) {
+            srs_freep(err);
+            port_ = srs_rand_integer(30000, 60000);
+            srs_trace("HTTPS test server listen failed on %s:%d, retry %d with new port %d",
+                      ip_.c_str(), port_, retry + 1, port_);
+        }
+    }
+
+    if (err != srs_success) {
+        return srs_error_wrap(err, "listen %s:%d after 3 retries", ip_.c_str(), port_);
     }
 
     if ((err = trd_->start()) != srs_success) {
@@ -527,8 +557,23 @@ srs_error_t SrsRtmpTestServer::start()
 {
     srs_error_t err = srs_success;
 
-    if ((err = srs_tcp_listen(ip_, port_, &fd_)) != srs_success) {
+    // Retry up to 3 times with different random ports if listen fails
-        return srs_error_wrap(err, "listen %s:%d", ip_.c_str(), port_);
+    for (int retry = 0; retry < 3; retry++) {
+        if ((err = srs_tcp_listen(ip_, port_, &fd_)) == srs_success) {
+            break;
+        }
+
+        // If this is not the last retry, generate a new random port and try again
+        if (retry < 2) {
+            srs_freep(err);
+            port_ = srs_rand_integer(30000, 60000);
+            srs_trace("RTMP test server listen failed on %s:%d, retry %d with new port %d",
+                      ip_.c_str(), port_, retry + 1, port_);
+        }
+    }
+
+    if (err != srs_success) {
+        return srs_error_wrap(err, "listen %s:%d after 3 retries", ip_.c_str(), port_);
     }
 
     return trd_->start();
@@ -835,9 +880,23 @@ srs_error_t SrsUdpTestServer::start()
         return err;
     }
 
-    // Create UDP socket
+    // Create UDP socket - retry up to 3 times with different random ports if listen fails
-    if ((err = srs_udp_listen(host_, port_, &lfd_)) != srs_success) {
+    for (int retry = 0; retry < 3; retry++) {
-        return srs_error_wrap(err, "udp listen %s:%d", host_.c_str(), port_);
+        if ((err = srs_udp_listen(host_, port_, &lfd_)) == srs_success) {
+            break;
+        }
+
+        // If this is not the last retry, generate a new random port and try again
+        if (retry < 2) {
+            srs_freep(err);
+            port_ = 30000 + (srs_rand_integer() % (60000 - 30000 + 1));
+            srs_trace("UDP test server listen failed on %s:%d, retry %d with new port %d",
+                      host_.c_str(), port_, retry + 1, port_);
+        }
+    }
+
+    if (err != srs_success) {
+        return srs_error_wrap(err, "udp listen %s:%d after 3 retries", host_.c_str(), port_);
     }
 
     // Get the actual port that was assigned

trunk/src/utest/srs_utest_rtc2.cpp

@@ -1516,19 +1516,19 @@ VOID TEST(KernelRTC2Test, SrsRtcFrameBuilderVideoFrameDetectorNullPacketHandling
 }
 
 // Mock bridge for testing SrsRtcFrameBuilder
-class MockStreamBridge : public ISrsStreamBridge
+class MockRtcFrameBuilderBridge : public ISrsStreamBridge
 {
 public:
     srs_error_t last_error;
     int frame_count;
 
-    MockStreamBridge()
+    MockRtcFrameBuilderBridge()
     {
-        last_error = srs_success;
+        last_error = NULL;
         frame_count = 0;
     }
 
-    virtual ~MockStreamBridge()
+    virtual ~MockRtcFrameBuilderBridge()
     {
         srs_freep(last_error);
     }
@@ -1577,7 +1577,7 @@ VOID TEST(KernelRTC2Test, SrsRtcFrameBuilderPacketVideoRtmpNullPointerCrash)
     // but the start packet (100) is missing. With the fix, the function should use
     // packet 101 (first available) instead of crashing on the missing packet 100.
     if (true) {
-        MockStreamBridge bridge;
+        MockRtcFrameBuilderBridge bridge;
         SrsRtcFrameBuilder frame_builder(&bridge);
 
         // Skip initialization and directly set up the test scenario