AI: Add utest to cover rtsp module.

trunk/src/app/srs_app_rtsp_conn.cpp

@@ -347,6 +347,14 @@ void SrsRtspPlayStream::set_all_tracks_status(bool status)
     srs_trace("RTSP: Init tracks %s ok", merged_log.str().c_str());
 }
 
+ISrsRtspConnection::ISrsRtspConnection()
+{
+}
+
+ISrsRtspConnection::~ISrsRtspConnection()
+{
+}
+
 SrsRtspConnection::SrsRtspConnection(ISrsResourceManager *cm, ISrsProtocolReadWriter *skt, std::string cip, int port)
 {
     manager_ = cm;

trunk/src/app/srs_app_rtsp_conn.hpp

@@ -88,7 +88,24 @@ public:
     void set_all_tracks_status(bool status);
 };
 
-class SrsRtspConnection : public ISrsResource, public ISrsDisposingHandler, public ISrsExpire, public ISrsCoroutineHandler, public ISrsStartable
+// The handler for RTSP connection send packet.
+class ISrsRtspConnection
+{
+public:
+    ISrsRtspConnection();
+    virtual ~ISrsRtspConnection();
+
+public:
+    virtual srs_error_t do_send_packet(SrsRtpPacket *pkt) = 0;
+};
+
+// A RTSP session, client request and response with RTSP.
+class SrsRtspConnection : public ISrsResource, // It's a resource.
+                          public ISrsDisposingHandler,
+                          public ISrsExpire,
+                          public ISrsCoroutineHandler,
+                          public ISrsStartable,
+                          public ISrsRtspConnection
 {
 private:
     bool disposing_;

trunk/src/app/srs_app_rtsp_source.cpp

@@ -250,6 +250,9 @@ SrsRtspSource::SrsRtspSource()
     req_ = NULL;
 
     stream_die_at_ = 0;
+
+    stat_ = _srs_stat;
+    circuit_breaker_ = _srs_circuit_breaker;
 }
 
 SrsRtspSource::~SrsRtspSource()
@@ -266,6 +269,9 @@ SrsRtspSource::~SrsRtspSource()
     if (cid.empty())
         cid = _pre_source_id;
     srs_trace("free rtc source id=[%s]", cid.c_str());
+
+    stat_ = NULL;
+    circuit_breaker_ = NULL;
 }
 
 // CRITICAL: This method is called AFTER the source has been added to the source pool
@@ -437,8 +443,7 @@ srs_error_t SrsRtspSource::on_publish()
         return srs_error_wrap(err, "source id change");
     }
 
-    SrsStatistic *stat = _srs_stat;
+    stat_->on_stream_publish(req_, _source_id.c_str());
-    stat->on_stream_publish(req_, _source_id.c_str());
 
     return err;
 }
@@ -457,8 +462,7 @@ void SrsRtspSource::on_unpublish()
     }
     _source_id = SrsContextId();
 
-    SrsStatistic *stat = _srs_stat;
+    stat_->on_stream_close(req_);
-    stat->on_stream_close(req_);
 
     // Destroy and cleanup source when no publishers and consumers.
     if (consumers_.empty()) {
@@ -475,7 +479,7 @@ srs_error_t SrsRtspSource::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_->hybrid_dying_water_level()) {
         _srs_pps_aloss2->sugar_ += (int64_t)consumers_.size();
         return err;
     }
@@ -531,6 +535,8 @@ SrsRtspRtpBuilder::SrsRtspRtpBuilder(ISrsRtpTarget *target, SrsSharedPtr<SrsRtsp
     // Lazy initialization flags
     audio_initialized_ = false;
     video_initialized_ = false;
+
+    config_ = _srs_config;
 }
 
 SrsRtspRtpBuilder::~SrsRtspRtpBuilder()
@@ -538,6 +544,8 @@ SrsRtspRtpBuilder::~SrsRtspRtpBuilder()
     srs_freep(format_);
     srs_freep(meta_);
     srs_freep(video_builder_);
+
+    config_ = NULL;
 }
 
 srs_error_t SrsRtspRtpBuilder::initialize_audio_track(SrsAudioCodecId codec)
@@ -679,7 +687,7 @@ srs_error_t SrsRtspRtpBuilder::initialize(ISrsRequest *r)
     }
 
     // Setup the SPS/PPS parsing strategy.
-    format_->try_annexb_first_ = _srs_config->try_annexb_first(r->vhost_);
+    format_->try_annexb_first_ = config_->try_annexb_first(r->vhost_);
 
     srs_trace("RTSP bridge from RTMP, try_annexb_first=%d", format_->try_annexb_first_);
 
@@ -997,7 +1005,7 @@ srs_error_t SrsRtspRtpBuilder::consume_packets(vector<SrsRtpPacket *> &pkts)
     return err;
 }
 
-SrsRtspSendTrack::SrsRtspSendTrack(SrsRtspConnection *session, SrsRtcTrackDescription *track_desc, bool is_audio)
+SrsRtspSendTrack::SrsRtspSendTrack(ISrsRtspConnection *session, SrsRtcTrackDescription *track_desc, bool is_audio)
 {
     session_ = session;
     track_desc_ = track_desc->copy();
@@ -1031,7 +1039,7 @@ std::string SrsRtspSendTrack::get_track_id()
     return track_desc_->id_;
 }
 
-SrsRtspAudioSendTrack::SrsRtspAudioSendTrack(SrsRtspConnection *session, SrsRtcTrackDescription *track_desc)
+SrsRtspAudioSendTrack::SrsRtspAudioSendTrack(ISrsRtspConnection *session, SrsRtcTrackDescription *track_desc)
     : SrsRtspSendTrack(session, track_desc, true)
 {
 }
@@ -1071,7 +1079,7 @@ srs_error_t SrsRtspAudioSendTrack::on_rtp(SrsRtpPacket *pkt)
     return err;
 }
 
-SrsRtspVideoSendTrack::SrsRtspVideoSendTrack(SrsRtspConnection *session, SrsRtcTrackDescription *track_desc)
+SrsRtspVideoSendTrack::SrsRtspVideoSendTrack(ISrsRtspConnection *session, SrsRtcTrackDescription *track_desc)
     : SrsRtspSendTrack(session, track_desc, false)
 {
 }

trunk/src/app/srs_app_rtsp_source.hpp

@@ -26,6 +26,10 @@ class SrsRtcSourceDescription;
 class SrsResourceManager;
 class SrsRtspConnection;
 class SrsRtpVideoBuilder;
+class ISrsStatistic;
+class ISrsCircuitBreaker;
+class ISrsAppConfig;
+class ISrsRtspConnection;
 
 // The RTSP stream consumer, consume packets from RTSP stream source.
 class SrsRtspConsumer
@@ -118,6 +122,10 @@ extern SrsResourceManager *_srs_rtsp_manager;
 // A Source is a stream, to publish and to play with, binding to SrsRtspPlayStream.
 class SrsRtspSource : public ISrsRtpTarget
 {
+private:
+    ISrsStatistic *stat_;
+    ISrsCircuitBreaker *circuit_breaker_;
+
 private:
     // For publish, it's the publish client id.
     // For edge, it's the edge ingest id.
@@ -201,6 +209,9 @@ public:
 // Convert AV frame to RTSP RTP packets.
 class SrsRtspRtpBuilder
 {
+private:
+    ISrsAppConfig *config_;
+
 private:
     ISrsRequest *req_;
     ISrsRtpTarget *rtp_target_;
@@ -264,10 +275,10 @@ public:
 
 protected:
     // The owner connection for this track.
-    SrsRtspConnection *session_;
+    ISrsRtspConnection *session_;
 
 public:
-    SrsRtspSendTrack(SrsRtspConnection *session, SrsRtcTrackDescription *track_desc, bool is_audio);
+    SrsRtspSendTrack(ISrsRtspConnection *session, SrsRtcTrackDescription *track_desc, bool is_audio);
     virtual ~SrsRtspSendTrack();
 
 public:
@@ -284,7 +295,7 @@ public:
 class SrsRtspAudioSendTrack : public SrsRtspSendTrack
 {
 public:
-    SrsRtspAudioSendTrack(SrsRtspConnection *session, SrsRtcTrackDescription *track_desc);
+    SrsRtspAudioSendTrack(ISrsRtspConnection *session, SrsRtcTrackDescription *track_desc);
     virtual ~SrsRtspAudioSendTrack();
 
 public:
@@ -294,7 +305,7 @@ public:
 class SrsRtspVideoSendTrack : public SrsRtspSendTrack
 {
 public:
-    SrsRtspVideoSendTrack(SrsRtspConnection *session, SrsRtcTrackDescription *track_desc);
+    SrsRtspVideoSendTrack(ISrsRtspConnection *session, SrsRtcTrackDescription *track_desc);
     virtual ~SrsRtspVideoSendTrack();
 
 public:

trunk/src/app/srs_app_srt_conn.cpp

@@ -531,7 +531,6 @@ srs_error_t SrsMpegtsSrtConn::do_playing()
 
     srs_assert(consumer_raw);
     SrsUniquePtr<ISrsSrtConsumer> consumer(consumer_raw);
-    SrsSrtConsumer *consumer_impl = dynamic_cast<SrsSrtConsumer *>(consumer_raw);
 
     // TODO: FIXME: Dumps the SPS/PPS from gop cache, without other frames.
     if ((err = srt_source_->consumer_dumps(consumer.get())) != srs_success) {

trunk/src/utest/srs_utest_app12.hpp

@@ -12,9 +12,10 @@
 */
 #include <srs_utest.hpp>
 
-#include <srs_utest_app6.hpp>
+#include <srs_app_rtsp_conn.hpp>
 #include <srs_app_stream_bridge.hpp>
 #include <srs_protocol_rtmp_stack.hpp>
+#include <srs_utest_app6.hpp>
 
 // Mock frame target for testing SrsSrtFrameBuilder
 class MockSrtFrameTarget : public ISrsFrameTarget
@@ -107,5 +108,52 @@ public:
     void reset();
 };
 
-#endif
+// Forward declaration
+class SrsRtspConsumer;
 
+// Mock RTSP source for testing SrsRtspConsumer
+class MockRtspSource
+{
+public:
+    int on_consumer_destroy_count_;
+
+public:
+    MockRtspSource();
+    virtual ~MockRtspSource();
+    void on_consumer_destroy(SrsRtspConsumer *consumer);
+    void reset();
+};
+
+// Mock RTP target for testing SrsRtspRtpBuilder
+class MockRtspRtpTarget : public ISrsRtpTarget
+{
+public:
+    int on_rtp_count_;
+    SrsRtpPacket *last_rtp_;
+    srs_error_t rtp_error_;
+
+public:
+    MockRtspRtpTarget();
+    virtual ~MockRtspRtpTarget();
+    virtual srs_error_t on_rtp(SrsRtpPacket *pkt);
+    void set_rtp_error(srs_error_t err);
+    void reset();
+};
+
+// Mock RTSP connection for testing SrsRtspSendTrack
+class MockRtspConnection : public ISrsRtspConnection
+{
+public:
+    int do_send_packet_count_;
+    SrsRtpPacket *last_packet_;
+    srs_error_t send_error_;
+
+public:
+    MockRtspConnection();
+    virtual ~MockRtspConnection();
+    virtual srs_error_t do_send_packet(SrsRtpPacket *pkt);
+    void set_send_error(srs_error_t err);
+    void reset();
+};
+
+#endif