AI: Support VP9 codec for WebRTC

Author: ossrs-ai

trunk/src/utest/srs_utest_manual_mock.cpp

@@ -306,6 +306,8 @@ ISrsRequest *MockRtcAsyncCallRequest::as_http()
 MockRtcSource::MockRtcSource()
 {
     on_rtp_count_ = 0;
+    rtp_audio_count_ = 0;
+    rtp_video_count_ = 0;
 }
 
 MockRtcSource::~MockRtcSource()
@@ -315,6 +317,14 @@ MockRtcSource::~MockRtcSource()
 srs_error_t MockRtcSource::on_rtp(SrsRtpPacket *pkt)
 {
     on_rtp_count_++;
+
+    // Count audio and video packets separately
+    if (pkt->frame_type_ == SrsFrameTypeAudio) {
+        rtp_audio_count_++;
+    } else if (pkt->frame_type_ == SrsFrameTypeVideo) {
+        rtp_video_count_++;
+    }
+
     return SrsRtcSource::on_rtp(pkt);
 }
 

trunk/src/utest/srs_utest_manual_mock.hpp

@@ -162,6 +162,8 @@ class MockRtcSource : public SrsRtcSource
 {
 public:
     int on_rtp_count_;
+    int rtp_audio_count_;
+    int rtp_video_count_;
 
 public:
     MockRtcSource();

trunk/src/utest/srs_utest_workflow_rtc_conn.cpp

@@ -81,25 +81,7 @@ ISrsRtcPublishStream *MockAppFactoryForRtcConn::create_rtc_publish_stream(ISrsEx
     return publisher;
 }
 
-MockRtcSourceForRtcConn::MockRtcSourceForRtcConn()
-{
-    rtp_audio_count_ = 0;
-    rtp_video_count_ = 0;
-}
-
-MockRtcSourceForRtcConn::~MockRtcSourceForRtcConn()
-{
-}
 
-srs_error_t MockRtcSourceForRtcConn::on_rtp(SrsRtpPacket *pkt)
-{
-    if (pkt->frame_type_ == SrsFrameTypeAudio) {
-        rtp_audio_count_++;
-    } else if (pkt->frame_type_ == SrsFrameTypeVideo) {
-        rtp_video_count_++;
-    }
-    return srs_success;
-}
 
 // This test is used to verify the basic workflow of the RTC connection.
 // It's finished with the help of AI, but each step is manually designed
@@ -121,7 +103,7 @@ VOID TEST(BasicWorkflowRtcConnTest, ManuallyVerifyForPlayer)
     mock_dtls_certificate->fingerprint_ = "test-fingerprint";
     mock_app_factory->rtc_sources_ = mock_rtc_sources.get();
     mock_app_factory->mock_protocol_utility_ = new MockProtocolUtilityForRtcConn("192.168.1.100");
-    MockRtcSourceForRtcConn *mock_rtc_source = new MockRtcSourceForRtcConn();
+    MockRtcSource *mock_rtc_source = new MockRtcSource();
     mock_rtc_sources->mock_source_ = SrsSharedPtr<SrsRtcSource>(mock_rtc_source);
 
     // Create a real ISrsRtcConnection using _srs_app_factory_
@@ -272,7 +254,7 @@ VOID TEST(BasicWorkflowRtcConnTest, ManuallyVerifyForPublisher)
     mock_dtls_certificate->fingerprint_ = "test-fingerprint";
     mock_app_factory->rtc_sources_ = mock_rtc_sources.get();
     mock_app_factory->mock_protocol_utility_ = new MockProtocolUtilityForRtcConn("192.168.1.100");
-    MockRtcSourceForRtcConn *mock_rtc_source = new MockRtcSourceForRtcConn();
+    MockRtcSource *mock_rtc_source = new MockRtcSource();
     mock_rtc_sources->mock_source_ = SrsSharedPtr<SrsRtcSource>(mock_rtc_source);
 
     // Create a real ISrsRtcConnection using _srs_app_factory_

trunk/src/utest/srs_utest_workflow_rtc_conn.hpp

@@ -66,18 +66,4 @@ class MockAppFactoryForRtcConn : public SrsAppFactory
     virtual ISrsRtcPublishStream *create_rtc_publish_stream(ISrsExecRtcAsyncTask *exec, ISrsExpire *expire, ISrsRtcPacketReceiver *receiver, const SrsContextId &cid);
 };
 
-class MockRtcSourceForRtcConn : public SrsRtcSource
-{
-public:
-    int rtp_audio_count_;
-    int rtp_video_count_;
-
-public:
-    MockRtcSourceForRtcConn();
-    virtual ~MockRtcSourceForRtcConn();
-
-public:
-    virtual srs_error_t on_rtp(SrsRtpPacket *pkt);
-};
-
 #endif