I don't think many people are using the Lime mini with a PC. I am assuming Windows rather than Linux. At some point, SDR Television may support the Lime but not yet. Most are using the Pluto, which is a much easier solution these days.
You should be able to do what you want with SDR Angel or Gnu Radio. I think your problem is probably not getting the transport stream rate right. You could use DATV Easy but it is probably better to use FFMPEG to generate a transport stream with the correct rate and right number of nulls. This gives you more control, but requires more thought. I see you are using 1/2 FEC and 333ks. That means the target bit rate for the transport stream will be 330 kb/s.
Download the FFMPEG executable ffmpeg.exe from
https://ffmpeg.org/download.html and put it somewhere useful, e.g. c:\ffmpeg\bin
Here is an example batch file. Copy and paste it into notepad, save as .bat and click on it. It should generate a test mpeg2 transport stream at 330kb/s and send it to localhost port 1234.
c:\ffmpeg\bin\ffmpeg -f lavfi -i testsrc2=640x384:rate=15 -f lavfi -i "sine=f=512" ^
-vcodec libx264 -s 640x384 -bf 0 -pix_fmt yuv420p -r 15 -preset slow -profile:v main ^
-x264-params "nal-hrd=cbr" -b:v 220k -minrate 220k -maxrate 220k -bufsize 100k ^
-acodec aac -aac_coder twoloop -ar 48000 -ac 1 -b:a 32k ^
-f mpegts -muxrate 330k -streamid 0:256 -streamid 1:257 ^
-metadata service_provider="Mycall" -metadata service_name="MyName" ^
-pcr_period 40 -pat_period 0.4 "udp://127.0.0.1:1234?pkt_size=1316"
pause
You can test this with ffplay. From a command line:
C:\ffmpeg\bin>ffplay udp://127.0.0.1:1234
Then send it to SDR Angel.
Breaking it down:
c:\ffmpeg\bin\ffmpeg -f lavfi -i testsrc2=640x384:rate=15 -f lavfi -i "sine=f=512" ^
This is your input. Normally it would be a camera and a microphone, or perhaps the OBS virtual camera and virtual microphone.
e.g. I am using c:\ffmpeg\bin\ffmpeg -f dshow -i video="OBS-Camera" -thread_queue_size 512 -f dshow -i audio="OBS-Audio"
these will typically be direct show inputs, hence the -f dshow which indicates the directshow filter.
The next part is the video encoding - this can be a lot more complex and call on nvidia or lightspeed hardware encoders. The example is H264 but many are using H265 if the CPU is up to it because the results are much better. Some are even using H266.
-vcodec libx264 -s 640x384 -bf 0 -pix_fmt yuv420p -r 15 -preset slow -profile:v main ^
-x264-params "nal-hrd=cbr" -b:v 220k -minrate 220k -maxrate 220k -bufsize 100k ^
As we are sending over a link with a fixed bit rate, we want the encoder to work at a constant rate. It is a low rate, so the frame size is small and the frame rate is only 15 frame per second. The second line is telling the encoder to use a constant bit rate. The buffer is to allow it to do this within a window. A large buffer makes this easier, but also increases latency.
This encodes the audio, with the advanced audio codec encoding to 48k (vs 44.1k) and at a bitrate of 32 kb/s.
-acodec aac -aac_coder twoloop -ar 48000 -ac 1 -b:a 32k ^
You can reduce that audio rate or even not send audio at very low symbol rates. If you have the bandwidth, you can have high quality audio in stereo, but you don't, so don't as the image will suffer.
finally, this is creating the mpeg transport stream at a rate of 33ks/s for your 333ks FEC1/2. The metadata is normally your callsign and name.
-f mpegts -muxrate 330k -streamid 0:256 -streamid 1:257 ^
-metadata service_provider="Mycall" -metadata service_name="MyName" ^
-pcr_period 40 -pat_period 0.4 "udp://127.0.0.1:1234?pkt_size=1316"
Mike