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The interface is divided into six sections:
- Top Panel
- Routing Panel
- Offline Processor
The top panel contains overall controls and utilities for enabling the connection to the server.
- The email and password are the website account credentials.
- The server menu shows the list of available servers in the cloud. Choose the server that corresponds to the equipment that you reserved on the website. This is where you currently select Access Analog or Robot Lemon servers.
- The connect/disconnect button is used to establish the connection with the server once the email and password are entered. When not connected, the plugin will pass through audio unaltered. When connected, the audio will stream to the server. You may connect to the server at any time and stream unprocessed audio, even without a reservation.
- The buffer setting can be used to increase the sample buffers if dropouts become recurring or frequent. See the Buffering/Latency tab on this website for more information. Higher buffer values have a better chance of reducing buffer underflows, but they cause more latency between adjusting a front panel setting and subsequently hearing that change. This latency is approximately half of the total buffer size value.
- The transmit format is used to set the format that will be used for the bi-directional audio stream across the internet. The uncompressed formats require a more stable connection between the client computer and the server. The compressed formats can be used if connection quality is limited. See the separate tab on Format for more information.
- The buffer graph displays the size over time of the sample buffers used to handle internet disruptions. The client and server buffers are overlayed on top of each other and a solid orange color indicates both have samples and audio should be streaming. See the Buffering/Latency tab for more details.
- The mix knob is used to mix the dry audio passing through the plugin, with the resulting wet audio that has been processed in the hardware rack.
The sidebar on the left side of the plugin lists all of the equipment located at the server to which you are connected. After connecting to the server with the proper credentials in the top panel, this list will show all of the equipment that is located at that server. Each piece of hardware can be in one of five state
- Unavailable – the equipment is currently in use by another user, or it has been taken offline for repair and maintenance.
- Available – the equipment is not in use and available to reserve or use on demand if you have more than 1 credit. Dragging the hardware into the routing panel will start an on demand use after a 45 second preview, and your credits will be deducted accordingly after that at a rate that is equivalent to the hourly reservation rate.
- Reserved – the equipment is currently reserved for you and available to be used in the routing panel. Your credits will not be reduced when using reserved equipment since it has already been purchased on the website.
- In Rack – the equipment is currently being actively used by you in the routing panel or in another instance of the plugin.
- Update Plugin – the equipment was added to the service after your version of the plugin was released. Go to the Downloads page on the website and install the latest plugin.
You can hover the mouse over any equipment and a tooltip that shows the rate at which credits will be deducted when using the equipment in on demand mode. It will also show any upcoming reservations for the equipment so that you can plan your on demand use.
The second tab of the sidebar shows the factory and user chain presets. These presets store the position of each piece of equipment in the chain as well as all control settings for each device. Double click a preset to instantly populate the routing panel and switch between entire chains while playing audio. You can create folders and presets in the user tree by using the right-click menu.
The routing panel is used to show the path of mono and stereo audio through the system, build analog chains with multiple pieces of equipment, and select equipment so that you can use the front panel in the equipment section. The source audio coming into the plugin is represented on the left side of the panel. The left audio channel is the top row and the right audio channel is the bottom row.
- The first piece of equipment of your analog chain is inserted into the left column of the panel by dragging your reserved equipment from the sidebar. Dragging equipment to subsequent columns to the right will insert them later in the analog chain and audio will flow through them in series from left to right. Clicking on an equipment name will highlight it and show the corresponding equipment faceplate in the equipment panel so that you can adjust the controls. A stereo device will occupy both the left and right boxes of a column.
- Mono pieces can be inserted on either the left or right audio channel. When two mono pieces are inserted in the same column, they are in a “dual mono” configuration where the left audio goes through one device and the right audio goes through the second device.
- If both pieces of equipment are the same model, they can be linked with the link button. When linked, control changes to the device on the left channel (using the equipment panel described later) will be duplicated to the box on the right. When linked, control changes to the box on the right channel can be adjusted independently of the left channel to compensate for minor hardware differences between the two devices.
- A gain slider is provided for each box in the analog chain. This gain adjusts the level of the signal being sent to the box. A mix knob is also provided to mix the signal from the box with a dry signal that bypasses that box.
- The “In” button gives the ability to easily insert and remove a box from the audio path while audio is playing. Similarly, the solo button will insert the box into the audio path while removing all other boxes.
- The final gain slider adjusts the level of the signal after the last box in the analog chain and just after the final A/D conversion. This effectively adjusts the output of the plugin and will always be the last slider on the right, regardless of the number of boxes in the chain.
- Meters (0 dBFS) show the RMS signal level at all stages of the audio path. If a sample reaches full scale, the clip indicator will light. Click on the meter to clear the clip indication. The clipping is handled by the A/D converter and can sometimes be desirable at moderate levels depending on your specific application.
Equipment can be removed from the chain by clicking the ‘x’ to the top right of the equipment name. Any columns that are empty will simply pass through audio unchanged. You can drag and drop equipment from one box to another inside the equipment panel in real time with audio playing to compare different analog chain equipment sequences with ease. Devices will swap positions or move to new points in the analog chain and the hardware will immediately be re-routed at the server to reflect the new configuration.
The equipment panel shows the likeness of the equipment currently highlighted in the routing panel.
- If you currently have the equipment reserved, the remaining time shows the time left in the current reservation that you made on the website. If you are using the equipment on demand, the remaining time shows the time until the next credit is deducted.
- The title shows the equipment name which indicates the model and instance. If there is more than one device of a particular model at the server, they are numbered #1, #2, etc.
- The preset menu contains a list of factory presets for the device, followed by user presets. Add a user preset by setting the controls of the front panel as desired, then click the ‘+’ button and name the preset. User presets can be removed by selected them and then clicking the ‘-‘ button.
- The equipment controls occupy the main area of the window. These control the robotics at the server. Use the equipment manual online (links on our equipment page) if you want to dive deep and understand how these controls work.
The offline processor is located at the bottom of the plugin and can be used to capture a clip of audio locally at the plugin, upload it to the server, process it at the server, and collect the full resolution 24 bit samples back at the plugin. The final processed audio sample rate is selected in the menu at the left of the processor. The nature of this process makes it unaffected by internet interruptions. This method can be used on any internet connection to safely capture the final processed audio for your reservation. The real time streaming format at the top of the plugin does not affect offline processing, which is always processed at the user selected sample rate in lossless 24 bit sample format. To process audio in your DAW, you can follow this sequence of steps:
- Start with playback idle and locate your DAW play head at the start of the audio you want to process.
- Select the desired sample rate of the final processed wav file using the selection on the left (1).
- Setup the active rack with the hardware you want to use. Use it in real time as many times as you want to dial in the settings. Do not worry if you use a compressed audio format during this time, the final audio will be full resolution as noted above.
- The arm button (2) is used to arm the processor to record audio, so click that next.
- Start playback in your DAW. The offline processor will begin uploading the audio to the server.
- When you get to the end of the audio you want to process, stop playback. The processor will finish uploading and you will see a representation of your unprocessed audio in the window, now stored at the server.
- If everything looks right, click process (3) to send the audio to the server for processing, or click cancel (3) and return to step 1. When you click process, the offline processing interface will display “Processing…”. The server will begin sending the audio through the hardware and the processed samples will be collected back at the plugin at the sample rate you specified. During this processing time you can continue working locally, and the plugin will simply pass through the local audio unchanged, because the server is busy processing your stored audio.
- Once the server finishes processing, the interface will display the audio and the “Processing…” message will go away.
- At this point you can click save (4) to save the processed audio to a wav file at the sample rate you specified, and it can be subsequently imported into your DAW. Alternately you can click cancel (3) to return to step 1.
If you have a wav file outside of your DAW that you want to process, you can follow a similar procedure to process it. Instead of arming the processor, you use the load button to load the wav file into the plugin which immediately beings uploading the audio. Then proceed with the process button as described above.
If you have more than one wav file to process with the same hardware at the same settings (like all stems in a project), then you can use the batch button:
Choose the output directory for the processed files at the top of the window. Processed filenames will be appended with “-aa” and can coexist with the unprocessed files. Use the add button to add one or more wav files to the list. When you click process, each wav file will be uploaded to the server, processed, and saved as a new wav file with the “-aa” extension. Each wav file will go through the offline processing sequence described above. During this time, you can close the plugin window and work on other things.
- Check this box to show or hide the equipment thumbnails in the sidebar.
- This checkbox can be used to automatically process audio after uploading when using the offline processor. When this is enabled, there is no need to click the process button once audio has finished uploading. Instead, processing at the server will automatically begin.
- This checkbox can override the address of the server. This is reserved for future use and should not be checked under normal opration.
- This checkbox can turn on some debug logging for the plugin. Use this to create a log file that can be emailed to technical support when issues arise.
You will need recording software that uses the AAX, VST, VST3, or AU plugin formats on either the Mac (10.10+, 64 bit) or Windows (7+, 64 bit) platform.
You will need a robust internet connection. In general, wifi is not ideal for streaming low latency audio completely error free, but if you have a strong signal then it is usually fine. Since this is a professional audio application, we cannot lose even one sample of audio. So, the requirements for this streaming application are a bit more stringent than something like Netflix, which can drop a frame here or there without any noticeable effect to the application.
We transmit audio across the internet in one of the following formats, selectable by the user:
- Lossless, 44.1 kHz, 24 bit
- Lossless, 48 kHz, 24 bit
- Lossless, 88.2 kHz, 24 bit
- Lossless, 96 kHz, 24 bit
- Lossless, 44.1 kHz, 16 bit
- Lossless, 48 kHz, 16 bit
- Lossless, 88.2 kHz, 16 bit
- Lossless, 96 kHz, 16 bit
- Compressed, 48 kHz, 512 kBit/sec
- Compressed, 48 kHz, 256 kBit/sec
- Compressed, 48 kHz, 128 kBit/sec
The compressed formats reduce the requirements on the internet connection but may not be desirable for professional results. If this is the only choice for a given user internet connection, the offline processing tool is provided to allow real time system usage with the compressed audio, but the final bounce or save can be done offline at lossless, 96 kHz, 24 bit resolution. See the user manual offline processing topic for more details.
You can measure your connection bandwidth at speedtest.net.
The system allows the user to select the audio format that will be used to transmit the data across the internet and back. This format can be chosen to match the sample rate of the user session, or it can be adjusted to match the quality and bandwidth of the internet connection.
When the sample rate on both sides of the SRC are the same, then no conversion takes place. Thus, when the session sample rate and transmit sample rate are the same, there is no sample rate conversion at the plugin. Similarly, when the transmit sample rate is 96 kHz, there is no conversion at the server.
The transmit sample rate should not be set higher than the DAW session sample rate, as that would use more bandwidth and result in no benefit.
The hardware A/D converter is set to 96 kHz.
When offline bounce is used, the audio is transmitted from the plugin to the server in lossless format at the session sample rate. The samples are converted to 96k at the server when sent to the hardware. You can pick the sample rate of the final wav file in the offline processor. The server converts the processed samples to this sample rate and transmits them losslessly back to the plugin. Thus, offline bounce looks like this:
The system employs two buffers to manage the inconsistencies of the internet. There is one buffer located at the server which receives samples from the client plugin. There is another buffer located in the plugin that receives samples back from the server. The user can select the entire system buffer size at the top of the plugin, between 300ms and 2500ms, and this is split between the two buffers.
The buffer graph is located at the top of the plugin. This shows the size of both the plugin and server buffer relative to the total system buffer setting. The two graphs are overlayed on top of each other and show the fill percentage of both buffers over time. For best streaming, each buffer should be approximately half full, resulting in a bright orange graph that is half of the height of the graph, as shown. If one graph is much larger than the other, you can use the reset button below the graph to clear up the issue.
Inserting the Analog Matrix plugin on the master track of a session is always possible at any buffer size setting because there are no parallel tracks to the master track and compensation is not necessary.
For individual tracks, each DAW provides plugin delay compensation. This can be used to compensate for the Analog Matrix buffer size setting and keep the track in sync with the session. Analog Matrix provides a selectable buffer size setting that represents the total buffer size, or delay, through the system. Each DAW has it’s own maximum delay compensation detailed below. The Analog Matrix plugin can be used on any individual track as long as the buffer size is less than the maximum delay compensation for that DAW, and delay compensation is enabled. If the Analog Matrix plugin requires larger buffer sizes for stable audio streaming, and the buffer size is larger than the maximum delay compensation, the track with the plugin will be out of sync with parallel tracks in the session. If you are processing the master track, then there are no parallel tracks, and so this doesn’t affect you.
If you are processing an individual track, you can compensate for this by manually moving the audio of the track earlier in the timeline. If delay compensation is not enabled, you would move the audio earlier by the buffer amount. If delay compensation is enabled, you would move the audio earlier by the buffer size minus the maximum delay compensation value.
How can I try out this service?
- Register a free account here.
- Download the free plugin here.
- In the plugin, connect with your account credentials and then drag in a free device from the left panel.
What should I know as a new user?
- If your connection is limited, use a 2500 ms buffer size and compressed 128k transmit format.
- This service uses TCP port 47300. If you cannot connect, you may need to allow this port in your firewall.
- If you are using Reaper, disable “Anticipative FX processing” under Preferences -> Buffering.
- The best way to save your final processed audio is to use the Offline Processor at the bottom of the plugin. This will upload all of your audio to the server first, thereby avoiding any chance of a connection dropout from affecting your audio. It also uses lossless, 24-bit audio, regardless of your transmit format setting.
- You can reserve a specific time on the website. The corresponding device will show as “Reserved” for you in the plugin at the specified time.
- You can buy and use credits. You can use any “Available” device in the plugin directly and your credits will be charged only as long as you are using it.
- The one hour reservation rate and the credit deduction rate are the same.
What buffer size should I use?
See the separate tab – “Buffering/Latency” – for a discussion of buffer size and delay compensation.
What should I do if I consistently see “buffer underflow”?
We recommend getting an accurate measurement of your bandwidth. Use speedtest.net, but make sure to “Change Server” and set it to “Nextlight Longmont, CO”.
- If your bandwidth is less than ~5 Mbit, then you will want to set the “Transmit Format” to a compressed format and set your “Buffer Size” to 2500ms.
- You may still have an inconsistent connection even if your bandwidth is adequate. Some providers intentionally time slice your connection with other users.
- Always consider using ethernet, or locate yourself near the wifi router. Packet errors cause streaming dropouts, and many errors are caused by wifi.
- Sometimes plugins before or after the Analog Matrix will try to run audio faster than real time, which is not possible in the analog domain. So, try removing plugins before or after the Analog Matrix to see if it has any effect.
- Sometimes our software has trouble balancing your buffering. Our system “learns” your connection each time you start/stop audio playback. Give the connection some time to learn, and try start/stop a few times.
- The “reset” button at the top right of the plugin also helps the system learn and adapt to your connection. After starting playback, try the reset button to restart your streaming.
- You should disable any VPN that might be active. Most VPN’s are not optimized for bi-directional low latency streaming.
- In rare cases, the system might not have an accurate value of your DAW sample rate, and therefore it is trying to stream audio at the wrong speed. In this case, the streaming usually lasts a few seconds before dropping, and it never comes back. The orange buffer graph at the top right of the plugin will also show linear increasing/decreasing buffer sizes (i.e. a ramp up/down). Use the DAW to set the session sample rate to a different value, and then return to your original value. This should update the plugin. Usually, the problem does not return after this.
Do I need an account?
Yes, an account on this website is needed to reserve equipment or purchase credits. Use the email and password from the website account inside the plugin to connect to the equipment rack.
Are there any special requirements when using Cubase?
You possibly need to set the Audio Pre-Record Time to 0.
Are there any special requirements when using Ableton Live?
Live has a feature to warp audio clips and adjust the timing to match the Live session. This can cause problems with your final bounced audio. After you use the plugin offline processor to capture your final bounce into a wav file, you may want to drag that bounced audio back into Live on a new track. In order to make the new bounced audio line up perfectly with the original track, you must manually disable “Warp” on the bounced audio clip.
Are there any special requirements when using Reaper?
If you are using Reaper, disable “Anticipative FX processing” under Preferences -> Buffering.
Are there any special requirements when using Logic?
Logic has a maximum delay compensation of 1000ms. Use a buffer size of 1000ms or less in the plugin if you are processing individual tracks where there are parallel tracks that require synchronization. If you are processing the mix/master bus, you can probably use any buffer size, because there are no parallel tracks.
Are there any special requirements when using Pro Tools?
Pro Tools has a maximum delay compensation of 341.3125 ms. Use a buffer size of 340 ms or less in the plugin if you are processing individual tracks where there are parallel tracks that require synchronization. If you are processing the mix/master bus, you can probably use any buffer size, because there are no parallel tracks. A 340 ms buffer size will require a strong internet connection.
If you must use a larger buffer than 340 ms on individual tracks, you can manually correct for the timing mismatch by sliding the audio in the track. As an example, if you use a 2500 ms buffer size in the plugin, you would then select all audio in the track that has the plugin inserted, then pick Edit->Shift, then put in (2500 – 341.3125) = 2158.6875 ms, and shift the audio to the left (or “earlier”) by that amount.
What format should I use?
You can first try the 24 bit format that matches the sample rate of your session. If the streaming is intermittent, you can drop the sample rate and/or use the 16 bit format. If the streaming is still intermittent, you should use a compressed format for real time streaming. When you are ready to capture your processed audio, you can use of the offline processor described in the user manual to get full resolution (lossless, 24-bit) 96 kHz audio.
How is my digital audio being delivered to analog hardware?
We employ real time A/D and D/A conversion at the server using Lynx Aurora(n) and Antelope Audio Galaxy 64 converters. Different devices are connected to each converter. In the future, we will provide the information about which converter is connected to each device on our website.
How can I minimize any noise through the system?
How do I use two mono units in stereo?
We refer to this as “Dual Mono”. We have two devices for most of our mono units for this reason. To use them in stereo, use the “Dual Mono” submenu when selecting devices in the equipment popup menu. If you want to reserve them, you will need to reserve both devices for the same time slot on the website. Once the devices are inserted, you can switch back and forth between controlling left and right by using the L/R button at the bottom. If you want the left and right settings to stay in sync, then set Link to “Absolute’ and the controls will be robotically linked. You can then control the left unit and the right unit will follow.
Has the Analog Hardware Been Modified?
Never. We use a wide range of cutting edge robotics to control the hardware from the front panel, just like you would if you were interacting with it manually. All plugin meters reflect the physical needles and LED’s of the actual equipment. We do not replace or change any analog component of the units.
Can I Create My Own Presets?
Yes! You can use the ‘+’ button at the top of each individual equipment panel to create presets for that piece of equipment. You can also use the chain preset panel in the left menu to create presets that store the entire routing panel plus all of the control values for each equipment. Use right click on the User folder to create your presets.
Can Anyone Steal or Listen To My Audio?
Your audio is never routed anywhere on our server except through the analog unit that you are using. So, we can’t hear your audio, and it is no more vulnerable to theft than any other unencrypted data going across the internet. Encryption increases the load on the internet connection, so we do not offer it at this time, but it could be added in the future.
Can I Put In a Request For a Specific Piece of Gear?
Of course! We want to hear what you are interested in. Please email firstname.lastname@example.org and let us know.
Why doesn’t the plugin work with Plugin Doctor?
Plugin Doctor has a speed setting that defaults to “Ultra”. Thus, it is trying to send audio through the plugin much faster than real time. The hardware converter at our server can only operate in real time. So, you just need to change the Plugin Doctor speed setting to “Real Time” and it should work normally.
What is a loopback cable?
We have two loopback XLR cables connecting two outputs back into two inputs of each converter, and these cables are available at no cost. When you reserve these and/or drag them into your plugin rack, you will stream audio through the cables just as if they were an analog unit. These cables allow you to learn the system and fine tune your streaming parameters without having to spend money. You can also use them to analyze the conversion quality.
Can’t find what your looking for? Need additional information or help?
You can usually chat with us using the icon at the bottom right of this page. If that doesn’t work, please click a link below related to your specific question or visit our contact page for additional inquiries.