Livestreaming

score can stream audio and video directly through its GStreamer and FFmpeg devices. A separate encoder application is not required. OBS remains useful when you want to combine score with other sources.

• Livestreaming
  • Streaming with FFmpeg
  • Streaming with GStreamer
    • RTP video
    • Audio and video in one pipeline
  • Livestreaming with OBS Studio
    • Windows
    • macOS
    • Linux
• Configuring Jitsi for high-quality streams

Streaming with FFmpeg

FFmpeg is built into score. Its device is documented under Libav.

To send an H.264/AAC stream to an RTMP service:

1. Add an FFmpeg device in the Device explorer and set Direction to Output.
2. Enter the service’s ingest URL, including the stream key, in Path / URL. For a local RTMP server this might be rtmp://localhost/live/stream.

3. Set the following output fields:

   Field	Value
   Width	1280
   Height	720
   Rate	30
   Audio Channels	2
   Muxer	flv
   Video Encoder	libx264
   Audio Encoder	aac
   Pixel Format	yuv420p
   Sample Format	fltp

4. Enter these Options, one per line:

   preset=veryfast
   tune=zerolatency
   g=60
   

   These are the settings used by score’s RTMP output recipe. g=60 sets a 60-frame keyframe interval: two seconds at 30 fps. Use Show options… to inspect the encoder and muxer options and adapt them to your streaming service.

5. Create the device. Assign FFmpeg:/Video to the last texture outlet in your video effect chain. Assign FFmpeg:/Audio to the audio outlet carrying the mix to stream. Replace FFmpeg with your device name if you changed it.
6. Start playback and check the stream at the receiving service.

The audio route is separate from the video route: selecting the video destination does not capture score’s main audio output. Set Audio Channels to 0 for video-only output. The Audio address is created when both an audio encoder and a positive channel count are selected.

Width and Height set the encoded image size; Rate is frames per second. Input Transfer describes the incoming texture: keep sRGB for ordinary SDR content, or choose the appropriate transfer for your source. See Video formats and color.

Streaming with GStreamer

score loads GStreamer dynamically. Install GStreamer on your machine, including the plug-ins used by your pipeline. Unlike FFmpeg, the GStreamer runtime is not built into score.

Add a GStreamer device in the Device explorer. Enter a pipeline in its Pipeline field, without the gst-launch-1.0 command. Output pipelines receive score’s video through appsrc name=video and audio through appsrc name=audio.

RTP video

For an MJPEG stream over RTP, enter:

appsrc name=video ! videoconvert ! video/x-raw,format=I420 ! jpegenc ! rtpjpegpay ! udpsink host=127.0.0.1 port=5000 sync=false

Set Width to 1280, Height to 720, Rate to 30 and Audio Channels to 0. Change host to the receiver’s IP address when streaming to another machine.

Create the device and assign GStreamer:/Video to the final texture outlet. Start playback to feed the pipeline.

You can receive the stream in another score instance with a GStreamer device using:

udpsrc port=5000 caps="application/x-rtp,media=video,encoding-name=JPEG,payload=26,clock-rate=90000" ! rtpjpegdepay ! jpegdec ! videoconvert ! video/x-raw,format=RGBA ! appsink name=video sync=false

On this input device, GStreamer:/video is a texture source. Assign it to a texture inlet and route the process’s output to a window. The lowercase video comes from the appsink name; output devices expose the uppercase Video address.

Audio and video in one pipeline

To send H.264 video and AAC audio to an RTMP service, use:

appsrc name=video ! queue ! videoconvert ! video/x-raw,format=I420 ! x264enc bitrate=6000 tune=zerolatency key-int-max=60 ! h264parse ! queue ! mux.
appsrc name=audio ! queue ! audioconvert ! audioresample ! audio/x-raw,rate=48000,channels=2 ! avenc_aac ! aacparse ! queue ! mux.
flvmux name=mux streamable=true ! rtmpsink location="rtmp://SERVER/app/STREAM_KEY"

Replace the URL with the service’s ingest address. Install the x264enc, avenc_aac (from gst-libav), parser, FLV muxer and RTMP sink plug-ins. voaacenc is another AAC encoder you can use in place of avenc_aac if your GStreamer installation provides it.

Set Rate to 30 and Audio Channels to 2. Route the final texture outlet to GStreamer:/Video and your audio mix to GStreamer:/Audio, then start playback. bitrate=6000 is the x264 encoder’s video bitrate in kbit/s; key-int-max=60 limits the keyframe interval to two seconds at this rate.

score supplies floating-point audio at the audio engine’s sample rate. The audioresample element converts it before the pipeline requests 48 kHz. Setting Audio Channels alone does not connect an audio source: the Audio address must also receive the mix.

Livestreaming with OBS Studio

OBS Studio can combine score’s output with cameras, desktop capture and other sources.

Windows

1. Install the OBS Spout2 plug-in.
2. In score, add a Spout output and assign it to the final texture outlet.
3. In OBS, add a Spout2 Input source and select score’s output.
4. For audio, use score’s WASAPI output and an Audio Output Capture source in OBS. ASIO output may not be available through loopback capture.

macOS

Add a Syphon output in score and route the final texture outlet to it. Select that server through an OBS Syphon input plug-in or a compatible bridge such as Syphon Virtual Webcam. Route audio separately.

Linux

For receivers that accept PipeWire video nodes, use score’s PipeWire Video Output and connect the nodes.

For an OBS Video Capture Device source, create a virtual camera with v4l2loopback:

sudo modprobe v4l2loopback video_nr=10 card_label="score" exclusive_caps=1

Then create a GStreamer device in score with this Pipeline:

appsrc name=video ! videoconvert ! video/x-raw,format=YUY2 ! v4l2sink device=/dev/video10

Set Audio Channels to 0, assign GStreamer:/Video to your texture outlet and start playback. Select /dev/video10 in OBS. This sends score’s video to the virtual camera without a separate GStreamer process.

For audio, use JACK or PipeWire routing to connect score to OBS’s JACK Input Client source.

Configuring Jitsi for high-quality streams

If you use Jitsi, by default the sound and video quality may be low and optimized for talking rather than music.

You can open your Jitsi room with the following set of parameters after the room name to increase the available video and audio quality, and enable stereo sound (if the browsers used support it):

https://meet.jit.si/<JITSI_ROOM_NAME>#config.disableAP=true&config.disableAEC=true&config.disableNS=true&config.disableAGC=true&config.disableHPF=true&config.stereo=true&config.enableLipSync=false&config.p2p.enabled=false&config.prejoinPageEnabled=false&config.resolution=1080

Note that for these parameters to be taken into account, the room must not have been created yet (that is, no one must have joined https://meet.jit.si/<JITSI_ROOM_NAME> already).