diff --git a/trunk/src/app/srs_app_rtc_codec.cpp b/trunk/src/app/srs_app_rtc_codec.cpp
index eb2d66613f..c923deb5a9 100644
--- a/trunk/src/app/srs_app_rtc_codec.cpp
+++ b/trunk/src/app/srs_app_rtc_codec.cpp
@@ -197,7 +197,7 @@ srs_error_t SrsAudioTranscoder::init_enc(SrsAudioCodecId dst_codec, int dst_chan
     enc_->channel_layout = av_get_default_channel_layout(dst_channels);
     enc_->bit_rate = dst_bit_rate;
     enc_->sample_fmt = codec->sample_fmts[0];
-    enc_->time_base = {1, 1000};
+    enc_->time_base.num = 1; enc_->time_base.den = 1000; // {1, 1000}
     if (dst_codec == SrsAudioCodecIdOpus) {
         //TODO: for more level setting
         enc_->compression_level = 1;
diff --git a/trunk/src/app/srs_app_rtc_conn.cpp b/trunk/src/app/srs_app_rtc_conn.cpp
index 547ef0aa60..8b5d97d07f 100644
--- a/trunk/src/app/srs_app_rtc_conn.cpp
+++ b/trunk/src/app/srs_app_rtc_conn.cpp
@@ -999,6 +999,7 @@ srs_error_t SrsRtcPublishStream::initialize(SrsRequest* r, SrsRtcStreamDescripti
     source->set_publish_stream(this);
 
     // Bridge to rtmp
+#if defined(SRS_RTC) && defined(SRS_FFMPEG_FIT)
     bool rtc_to_rtmp = _srs_config->get_rtc_to_rtmp(req->vhost);
     if (rtc_to_rtmp) {
         SrsSource *rtmp = NULL;
@@ -1019,6 +1020,7 @@ srs_error_t SrsRtcPublishStream::initialize(SrsRequest* r, SrsRtcStreamDescripti
 
         source->set_bridger(bridger);
     }
+#endif
 
     return err;
 }
diff --git a/trunk/src/app/srs_app_rtmp_conn.cpp b/trunk/src/app/srs_app_rtmp_conn.cpp
index 4cf58610f8..d6fe9d6307 100644
--- a/trunk/src/app/srs_app_rtmp_conn.cpp
+++ b/trunk/src/app/srs_app_rtmp_conn.cpp
@@ -983,7 +983,7 @@ srs_error_t SrsRtmpConn::acquire_publish(SrsSource* source)
     }
 
     // Bridge to RTC streaming.
-#ifdef SRS_RTC
+#if defined(SRS_RTC) && defined(SRS_FFMPEG_FIT)
     if (rtc) {
         SrsRtcFromRtmpBridger *bridger = new SrsRtcFromRtmpBridger(rtc);
         if ((err = bridger->initialize(req)) != srs_success) {