takev

Hello everyone, I am currently renaming my company VOSGAMES to Pokitec. I will be developing more products and I wanted a name that was less linked to "games". Cheers, Take Vos

Physically I find firewire to make a much better electrical connection. The USB-C connectors that I've seen just moving them up and down, will break the electrical connection, or just a slight pull will completely disconnect. It could be that Apple USB-C connectors are extremely bad.

I am not sure about this connector. It looks like it is not very reliable to work with live audio. My power adapter seems to cut-in-and-out with the connector on my MacBook. Does anyone have any experience, or a solution?

Hi, I just released Boom Recorder 8.7.0. I think many people here have been waiting to save and load preferences for different projects. I Also made it possible to have a larger ring buffer, for people who record at 384 kHz Cheers, Take Vos

I've been reading up and watching Apple's WWDC videos. I think I will start from scratch with my new application, not sure how to call it though. I am thinking of making a document-based application; meaning that each project you do is a separate document that will be stored with all the audio files. You can simply open this document to start Boom Recorder and continue where you have left off when you were working on that project. A document is a a bunch of mixer/input/output/audiounits/metadata/timecode/recording-widgets that you can place free form on a canvas, similar to the omni-graffle drawing program. You will be able to open two documents at once and copy a set of widgets from one document to another. But only one document can be active at a time, i.e. only one document can be connected to the audio interface(s). You will also be able to save a template of pre configured widgets. This is useful for saving a certain setup that is shared between multiple projects. When you change settings of the widgets that are in the template, it will share these changes in all the documents that use the template. One last thing, I will also add auxiliary windows, which are extra windows that show a view of the canvas, you can place these windows on other screens, so we will have proper multi-(full)-screen support.

Yea, I got a LG 21:9 2650x1080 monitor it is really nice for gaming as it adds some peripheral vision. Also because I play lots of MMO games which have lots of windows, the windows can go on the side and you can still play the actual game at the same time. Which is also true for 3D modelling where all the tools go to the side.

Sinnlicht, Boom Recorder can already record to multiple locations at the same time. Select the number of folders you want, add more files and assign them to the other folders, and assign channels to these files. I am actually thinking of removing the compatibility table from my website; I have not seen incompatible hardware for about a decade. Tom, your request about being able to align strips to physical dimensions of the mixer has made me rethink the user interface for this to be a more freeform canvas to put down the mixing elements.

For gaming I now have an Ultra Wide Screen monitor, 50% more horizontal space. Yea, I guess using a trackpad it is rather annoying to bumping the fader. I guess I should make that a configuration option, I think scrollwheel could be nice if you actually have a scrollwheel. MIDI control would be a lot better for this.

Hi Philip, I am not sure about latency from MIDI when Boom Recorder is working hard. But I believe I implemented the MIDI in a separate thread from everything else, so unless the computer itself is grinding to a halt I think the latency should be quite good. Of course, I have no idea what will happen when you are using hundreds of channels in Boom Recorder, only a few customers are using over a hundred channels. Since I know a few customers are using so many channels, I will try to limit the amount of screen real estate per channel very low. One thing I could offer is a simple indicator shown as a single dot where I can through colour (and shape) give information about the loudness of a channel. I've been thinking on how to control channels with a mouse. How does using the scroll wheel of the mouse (or touchpad) to change the value of a parameter sound? I am not entirely sure, because you may expect that you scroll when using the scroll wheel, not to change a value.

Did you know in Version 8 there is already MIDI control for tape transport?

Hi Jeff, Since I will be hosting AudioUnits, I will no longer pursue putting echo cancellation in Boom Recorder, instead it will come for free. In fact almost any AudioUnit you can find or buy will hopefully work correctly with Boom Recorder, and you get to decide how you want to use Boom Recorder. I agree that Boom Recorder's user interface needs to stay elegant. However the current patch bay gui is not user friendly at all especially when using a large number of channels. But, things are going to change significantly, I hope you guys will follow me. Which is why I want a lively discussion on this topic, as I read myself into the development documentation about how AudioUnits actually work.

Hello Al, I personally looked for an echo cancellation for my Twitch channel (Twitch is a live broadcast gaming website). I don't like wearing headphones, so I would play my game sound through my speakers. My microphone would pick up the game sound, and both the direct game sound plus everything picked up on my microphone would be send to Twitch. An echo cancellation unit would subtract the direct game sound (after automatically finding the delay, volume and possibly normalising EQ) from the microphone input. Eventually I found a plugin for the OBS (broadcasting software) to do echo cancellation, I am quite happy with that. You could possible use this if you play music in a dance scene, and subtracting the music from the microphone picking up the dialogue. Or maybe you can put a microphone near a noisy air-conditioner and subtract that sound from your boom microphone. However I am not sure what the quality is of the default Echo Cancellation AudioUnit that comes with OS X, it may be rather good, or not. But Boom Recorder is used by many customers including ones that don't use it to record any kind of entertainment. Someone may find it useful.

Hello Philip, Don't worry, I will not try to make Boom Recorder look like a simulation of a mixing deck, instead I will try to make the user interface very clean according to the Apple style guidelines. In my new vision you are allowed to create several windows of channel strips. You can, on your main window include only a set of meters that will give you pretty much the same layout as the current Boom Recorder. But I think you want to add some extra features on the main window, I think I can design the user interface in such a way that each channel will actually take even less room then in Boom Recorder 8, since you can group channels together. However I am not sure if I will maintain both a vertical and a horizontal layout, I may decide on a vertical only layout.

Thank you CraigF for pointing me to double M/S, I haven't heard of that before. I notice there is an AudioUnit plugin for processing Double M/S. Which decides it for me, I am going to support AudioUnits in Boom Recorder as first class citizens. I believe there are quite a few people using Dante Audinate with Boom Recorder, since you can buy the Dante Virtual Soundcard, which is a CoreAudio driver. Caleymw I will indeed allow you to listen to solo, and create taps at any place you like. I just thought of the possibility to bypass any AudioUnit for solo so you can audition an AudioUnit isolated, not sure how to handle this. Maybe solo is a very special path. The user interface is starting to form in my mind. I am thinking of an empty grid, where you can create a (multi-)channel strip by drag-and-drop small AudioUnit widgets underneath each other. You can open multiple windows of those grids, so you can use multiple screens. The top widget allows you to select audio-interface-input(s), or an aux input(s), The bottom widget will route the audio to an audio-file-track, aux or an audio-interface-output. Other widgets will allow you to tap the audio signal(s) to an aux, track or output in the middle of the strip, or insert a signal(s) into the strip. There will also be blank widgets, so you can align the strips, including blank widgets that automatically resize. All controls on all the widgets can optionally be controlled by MIDI. I am hoping I can also send feedback back into a controller to control leds, etc.

Hi, I am seriously thinking about doing some major development for Boom Recorder and start working toward version 9. I am thinking of getting rid of the patch bay and creating a better user interface for this function; routing inputs to outputs and files. There are several reasons for getting rid of the patch-bay and maybe what is shown on the main window: The current interface is cumbersome when working with over a hundred channels. The storage of the patch-bay data is large and is hampering the possibility of having multiple configuration files. I want to increase the number of channels above and beyond 256. The CPU usage is higher than it needs to be (even though it is not that much) I want to add effects: echo-cancelation, M/S decoding, compression, cut off filters and equaliser I want to control these effects and other things using MIDI. Including the ability to mix the audio live. As a replacement I am thinking of a more traditional channel strips like they have on mixers. The channel strips will have the ability to group multiple inputs. At the moment I have the following in mind: Strip name An input selection, either an input/buffer. (multiple inputs. buffers are like aux ports that can feed back into another strip) Peak and hold indicator on the input. Trim/Gain, both attenuation or amplification. Invert. Volume indicator post trim/gain. Delay. Echo cancellation auto duck mid/side. nr of dBs to compress. low cut, high cut, mid cut frequency and strength. low, mid, high EQ attenuation or amplification. solo mode: off/isolated/processed/equalised output on/off/privacy output bus/buffer output fader output balance (equal power, only for stereo) There will be separate user interface elements to set up monitoring outputs and file assignments. Do you have any suggestions?

Sadly it doesn't solve my confusion. It sound like Boom Recorder would somehow put corrupt audio data into an audio file after a while. The code for doing this hasn't changed for many many years, and I have customers who record continues for months for wildlife, and for days on reality shows.

Fauzan, Do you mean you just stop and start a new recording Boom Recorder and that fixes the issue. That is really odd, because nothing changes with the communication between Boom Recorder and the audio interface when stopping or starting a recording. Boom Recorder is capturing audio at all times, whetter recording or not, this is how pre-roll works. I am really confused right now, this information has to sink in before I comment on this further. Cheers, Take

That is funny, I was working on the exact same implementation; a pre record buffer that jumps to seconds during idle, and stopping on a second as well. However the pre-record buffer is running at the word clock speed, so I still need to do the calculations for non integer frame rates and pull downed sample rates. I don't use floating point calculations, I do them in integer that way it remains accurate, but it is a bit more difficult.

I've been thinking about it for a while now. Simply stated this is a bug in AVID that cannot be completely solved by an audio recorder. We can do our best to get it correct in most cases, but it is mathematically impossible to always start a recording at a frame boundary. It is pretty simple for the standard 24,25,30 fps and recording at 48000Hz to start the audio file at a timestamp that is a multiple of 48000 samples. Starting the audio file at a timestamp that is a multiple of the sample rate, which would start a recording at exactly the start of a second. However if you would put this 48000 Hz file on a 29.98 fps time line 'actual seconds' do not align with frames boundaries anymore. The problem is that a BWF file has no notion of the frame rate of the movie. The second problem is that the timestamp is recorded as number of samples since midnight, so it is not possible to calculate the start of a frame (or second) without knowing the frame rate of the movie. In the newest versions of Boom Recorder you can already specify the sample rates and frame rates how the file is recorded on the set and how the file will be used in post production. It will be important to set those correctly (it didn't use to matter a lot, because the post-production-frame-rate is not recorded in a BWF file) for the record-on-frame-boundary feature to work. Still, for drop-frame-rates I can probably never get it to work. Also because of how Boom Recorder works, I don't think I can make it that the file ends at a frame boundary.

It is going to be difficult to add this to the BWF standard, I believe that the specification for how audio recorders should interpret a start of frame or even a start of a second will be 2 times larger than the BWF standard itself The biggest problem is that the timestamp in a BWF is recorded as "number of samples since midnight" and that the frame rate is not recorded. It is pretty easy to know where to start a second with 24, 25 or 30 fps, however it is going to be a lot more interesting when recording at 29.98 and don't mention on what to do on drop-frame-rates?

Boom Recorder does not start or stop at exact start or end of a frame. I will have to investigate if I can make Boom Recorder do this. For your information the BWF standard does not specify that the recording needs to start and end at frame edges. In fact the timestamp in the header is sample accurate.

Hi, I have had a few customers where an audio edit application would corrupt Boom Recorder audio files during import. Always make a copy of audio files before importing them into any application. The audio edit application would move the audio in the file to earlier as it created a shorter audio header. If the audio edit application would crash or the import would fail half way through this process, the audio file would get corrupt. The header would be incorrect and therefor audio applications are not able to read in the file anymore. Also the audio in the file would get chopped up half way through and the two audio segments may also have a different alignment. Therefor even when importing the file as raw audio could only retrieve half of the audio. My new Audio Rescue application would open a file. - If it is a wav file it may be able to get some information from the header. - You then select the sample format, sample rate and number of channels. - When you press rescue, the application will search through the complete file for audio segments and guess its alignment. You then get a choice to save each segment. The resulting audio files are really basic, none of the metadata that was in the original file is maintained. You can download the application here: http://www.vosgames.nl/downloads/ I hope this application will one day make someones day less rotten. P.S. This application is different from BR File Fixer, which would completely fix a corrupt Boom Recorder audio file due to a power failure or god forbid a crash with Boom Recorder. Cheers, Take

Thank you for the information. The newest version of Boom Recorder has a slightly different IOCycle so I was worried that it may have been the cause. But 7.25.3 has an IOCycle that has done its job for many years. What I found interesting was the audio sounded like it was mixing audio from slightly earlier, just a few miliseconds. A sort of reverb was clearly audible. It is like somehow audio buffers where partially cleared and partly reused and mixed-in with new audio. Make sure you have downloaded the latest CoreAudio drivers for your audio interface, and you may want to contact Yamaha if they have heard about this issue before.

Hello Richard, It may be possible (not all CoreAudio drivers support this) to make an aggregate audio device from the Yamaha+Apple Buildin devices. You can make an aggregate device using the Audio MIDI Setup utility. Boom Recorder can record and listen to timecode from an aggregate device. However because the clocks of the Yamaha and the Build-in device are not in sync, one of the devices will need to be resampled to follow the other (the aggregate device will do this automatically) you will need to select the master device. I am not sure how well resampling works, it may cause weird problems.

Hi, May I know the exact version of Boom Recorder you are using? You can see this from the Boom Recorder menu, by selecting About. Cheers, Take Vos