takev

Oops, I missed a part of Darran's post. Having the Sonosax be the master and the MOTU a slave to the SPDIF from the 744T, and the SMPTE from the 744T into an audio input for Boom Recorder, should work as well. Cheers, Take

Hi, Yip, you can monitor using Boom Recorder, although with some latency. It is indeed a good way to hear if there are clocking issues. But if two clocks are really close together it may take more than 30 seconds to hear clicks, does it not? I guess you will also need to clock the Sonosax to the 744T, then everything is clocked to the 744T so there should be no clocking issues, I think. Cheers, Take

Hello everyone, I also wanted to mention that Dolby TrueHD (http://en.wikipedia.org/wiki/Dolby_TrueHD) format itself is 24bit/96kHz, this is the format as how it is now presented at home on a HD DVD player. This format also uses a lossless compression, so that you will not get compression artifacts like with mp3 or dolby digital. If the destination format is 24bit/96kHz would you not want to record in this format as well? Cheers, Â Â Take

Hi, Ok, I'll have a guess. Is it one of these headphones (more neckphones really), that transfer the audio through your jaw bone? My seconds guess would be the other way around, a microphone that listens to your jaw bone, for communication in a high noise environment. Cheers, Take

Hello vin, Maybe taking a synthesizer and hooking it up to a oscilloscope to show the wave form. You start by changing the volume, the wave becomes higher/lower. Then you press an other note, to make the wave form tighter Pick other sounds, to see a different wave. Setting the oscilloscope to very slow, you could show ADSR (Attack Decay Sustain Release). etc. picture tells a thousand words and such. Cheers, Take

Hello everyone, Someone asked me to comment on how word clock and timecode signals can change the sample rate. So here I am doing a brain dump, I hope I will be coherent. Every AD converter uses an oscillator, the early AD converters 16 bit/48 kHz oscillate once for each sample. For each pulse it measures the voltage on the input and converts it into a PCM value of 16 bits. You could switch between the internal oscillator, or the word clock driving the AD converter directly. These days we want 24 bit samples, but simply measuring the voltage doesn't work anymore, 24 bit AD converters are usually delta-sigma (or sigma-delta) converters (or hybrids between the old and delta-sigma). A delta-sigma doesn't directly measure the input voltage, but does this indirectly by measuring the time for a bucket to fill. It needs a very high frequency oscillator for this, mostly in the high Mhz range. You can no longer directly drive these AD converters with a word clock, we come to that later. The precision of these delta-sigma AD converters is directly related to the stability of the oscillator. With stability I mean that each pulse of the oscillator takes exactly the same amount of time, low jitter. If the oscillator jitters, the noise floor of the AD converter raises. Crystal oscillators by itself are not stable enough to reach the full 24 bit of precision, therefore they make them temperature controlled (putting the oscillator in a small oven with a thermostat), and may contain multiple crystals, or whatever. The frequency of the oscillator, of course, controls the sample rate. But the frequency of the oscillator is much higher than the sample rate. The frequency of the oscillator can be tuned, this can be done by a PLL (Phase Locked Loop). The frequency of the oscillator is divided to the sample rate frequency and then compared to the incoming word clock, the frequency of the oscillator is then tuned up or down, until the two are locked together. This is a pretty fluid process, so if the word clock has jitter this is averaged away and does not increase the noise floor of the AD converter, but it could add a lower frequency modulation. The same PLL trick can also be done using the timecode signal, which has a frequency itself. The speed of the motor of an analogue tape deck is controlled in a similar way. The timecode signal also contains the current time stamp and is used to say when a sample is recorded. An BWF audio file only shows the time stamp for the first sample in the audio file. Which is why it is important to run the timecode at the same speed as the sample rate; either by locking the sample rate to the timecode signal, or by locking the sample rate to a word clock that runs at the same speed of the timecode signal. Some house clocks can deliver a word clock, a timecode signal, tri-level sync and black burst. The simplest way to design one of these would be to start with a stable high frequency oscillator to make a black burst signal, then you would use frequency dividers to create the lower frequency word clock, tri-level sync and a timecode signal from this. I've seen many house clocks that have a black burst input to generate the other three, you can also get separate black burst generators. Cheers, Take

Hello everyone, I've just released Boom Recorder 7.5. Someone had asked me for Boom Recorder 7.4 a function to start a new recording (take) while he was recording, there was a special option in the File menu for this. I have modified this function, so you can start a new recording by pressing the "rec" button during recording (you will have to enable this in the preferences). If one overrides the position of the file number with %f (useful when importing mono files in a program that sorts them alphabetical), the numbering started with 0 instead of 1, this is now fixed in version 7.5. I removed the number of folders and number of files limitations from the Lite version, because it was confusing. I've done some more work on the soundlog, it now adds a soundlog.xsl. If you open the soundlog.xml file in Safari it will now shows a table of all the recordings in this session. This is still a useless feature until Boom Recorder can automatically import the audio files from the previous session. Cheers, Take Vos

Hello Noah, I believe that most NL edit software and Boom Recorder count the number of samples to calculate the current timecode. So it is very important to have the sample rate of the audio interface to be very accurate to have the calculations yield an exact time. Well, I am lying, the sample rate need not to be exact in the real world, but it needs to be in complete sync with the camera; as here as well a film-edit applications count how many frames has passed to calculate the time. So if you don't want your audio to drift compared to the camera, one would need both a word clock and a timecode signal to be generated from the same device (or two devices that are calibrated to the same speed). Film cameras may lock to a timecode signal, but video cameras often need a blackburst signal or tri-level sync in combination with a timecode signal. Some audio interfaces like from MOTU can also lock their sample rate to a timecode signal without a word clock. But all of this is not needed when you just do a feature film the takes for dialogue is often too short to notice any drifting, because at the start of each take both the audio recorder and camera uses the current timecode at that moment. But if you are recording concerts or beauty pageants, where you need to record audio continuously for many hours and have multiple cameras this becomes an issue.

Hello everyone, I've just released version 7.4. There are three changes: - I noticed when one resized the preferences window some of the labels where not moving with it. So I fixed that. - Someone had asked for an "instant file split" feature. When you press option-command-T or select "New Take" from the File menu, the file is stopped and immediately started again with an incremented take number. The two files will be seamless if there is a little bit pre record buffer. - I am also working on an XML sound report. I simply have put the iXML of the recordings between BWFXML-LIST tags. Anyway, it will be possible using XSLT to show a nice looking sound report as a web-page, or possibly print it out and all other kind of funky things. Cheers, Take

Hello everyone, I've just released Boom Recorder 7.3. With this version, Boom Recorder Pro now has the same limitations as the Boom Recorder Studio variant. I am discontinuing Boom Recorder Studio. As Final Cut Pro sorts mono audio files alphabetical according to filename, someone has asked me to move the file-number before the channel name. You may now use the %f tag in the filename template to place the file-number in an other location than the default. You can download Boom Recorder from: http://www.vosgames.nl/downloads/ Cheers, Take Vos

Hi there, Dithering is one of the things I still have to add to Boom Recorder. But I already did some research on this topic. Dither means adding white noise to the audio signal, so that the least significant bit of the 16 bit signal will twiddle, after adding this noise the 24-bits are truncated to 16-bits. Basically you are trading in distortion for a higher noise floor. And everyone agrees the distortion sounds worse. But if your noise floor (or even room tone) is already above this least significant bit of the 16 bit signal you do not have to add dithering noise on top of this. Cheers, Take

Thank you all very much, I hope I can get the funds together to do some advertising. Cheers, Take

Hello everyone, As you know I am the developer of Boom Recorder. I am thinking of advertise my product in a magazine. Could you tell me which magazines you are reading. I am also looking for a good general film production magazine for me to just read, as I am trying to make my own independed science fiction movie. And as it will be the first movie I will write, direct and finance I need all the help I can get. Oh, and a couple of weeks ago I had my first Boom Operating job, that thing is quite heavy after a while, and the mixer wasn't that light either. I just hope the sound is ok. Cheers, Take

Hello everyone, I've just released a new version of Boom Recorder. You are now able to change the maximum file size; where Boom Recorder splits the recording and continues into a new audio file. No other changes or bug fixes have been made. Cheers, Take

Hello Larry, No problem, I found an easy way to implement the last selected interface, by remembering the name of the interface. This is actually the wrong way to do it, Boom Recorder should get the unique interface id and remember that, but I am a little bit lazy. Cheers, Take

Hello everyone, I just released version 7.1.2 in which the following issues where solved: - Boom Recorder 7.1 and 7.1.1 crashes when there are no audio interface with inputs connected to the computer. - The new buffer size slider and input box, introduced in version 7.1, moved accross the window on resize. As always you can download this version from: http://www.vosgames.nl/downloads/ Greetings from Amsterdam, Take

Hello Ron, Am I correct, that after you press stop, it takes a while for Boom Recorder to show you it stopped until the external DVD writer stops writing? But eventually it will stop? After you press stop, Boom Recorder should show the wait-flower to the right of the tape-transport-buttons. Boom Recorder first makes sure the audio file(s) are completed before putting the transport buttons back to "stop". I hope this is what happens in your case. Have a nice trip, and lots of fun. Cheers, Take

Hello Jeff, I do not know what people do with Boom Recorder, but because Ron is having difficulties, I made sure I asked. It seems that he is writing to an external firewire hard disk, not to an external firewire DVD, so he has some other issue. If OS X is able to write directly to a DVD disk without the use of burning software, then Boom Recorder should be able to write directly to such a DVD disk. Of course you are right that writing to a DVD might be slow, however with today speeds of DVD recorders this may not be a problem for everyone (depending on the number of channels). You can see how well the recording is doing by looking at the ring buffer circle of Boom Recorder. Cheers, Take

Hello Ron, With the Boom Recorder "stop" problem you are having, are you writing directly to the DVD RAM drive from within Boom Recorder? In case you are not receiving my reply from your email, could you tell me exactly is happening when you can not stop Boom Recorder is using a personal message via this forum? Take

Hello everyone, I've just released Boom Recorder 7.1.1. I've removed the pull-up option from the Hardware Preferences and added a "Stamp sample rate as" to the Metadata preferences. I was worried that people and even myself didn't understand what was happing with the pull-up option and, I've read some things that showed me that it wasn't complete anyway. Now you can select three ways to stamp the sample rate in the audio file: the same as the actual sample rate, 0.1 % below the actual sample rate, 0.1 % above the actual sample rate. I think this is much more clear to everyone. Cheers, Take

Hello everyone, I've been talking to the author of BWF widget about the audio files Boom Recorder creates, and I've incorporated some of his suggestions in this new version, I've changed the description in the bext chunk so that each field is prefixed with a 'b'. I also added some room after the iXML data, so that programs could modify the iXML data without moving the chunks around. A couple of people have experienced "Computer to slow" messages, which may be caused by to small or to large audio interface buffers, I've added an new control in the hardware tab of the preferences panel where you can select this buffer size. Somehow it resets to a default value every time you start Boom Recorder, very strange. On the same hardware tab, I've made it so it will select the last selected interface when you restart Boom Recorder, or connect the audio interface, something a lot of people have asked me to fix. I've added the possibility to drag and drop old audio files in the sound log window again, however a lot of metadata is not yet read from this audio file. I made some fixes with the tape transport controls, there where bugs where you could not start a recording right in the middle of playback, and Boom Recorder would actually hang on it. The following is something to try and make the user interface a little bit more user friendly for people who have never seen a patch bay matrix before, it may even be handy for the more experienced user. I've added a few buttons in the patch-bay preferences tab, for automatically assigning the right number of files, and channel->file->folder assignments, depending on how machine channels and folders you have selected. Selecting 64 channels and 8 folders, and then pressing the mono button may not be smart, I've waited a couple of minutes without anything happening. Although if you have to assign that many channels to so many folders, it may still be faster to actually wait an half an hour than doing it manually (it would create 512 files). Cheers, Take Vos

Hello everyone, Now that some of you have tried the new version. I have a little bit of a feeling that I've made it slightly to complicated. I know a lot of people here are pretty comfortable with a patch-bay/audio-routing matrix. But people who are seeing Boom Recorder for the first time may think: ?. I wouldn't like it for my program not to be usable by someone not reading the manual. (where that three negatives, sorry) Cheers, Take

Hello Sergio. I am glad it works for you know. This is indeed mighty weird, I'll will try and investigate myself what it is with these buffer sizes. I am very busy implementing the buffer size myself. I will have to wait and see. The only thing that I can think off is that the larger buffer sizes means that the CPU can not cache all the values, making the calculations significantly slower. Or I made a bug that makes it do the wrong thing with bigger buffer sizes. Take

Hello Sergio, Processor overload, means that the deadline of the I/O cycle wasn't met. This is maybe caused by having a very small device buffer, this is the same buffer that causes (monitoring) latency. You can not yet configure this buffer size in Boom Recorder, but you can change the size of this buffer in other applications like ProTools or other DAWs. Could you try and make this buffer larger, in an other application and try again? In the mean time I will try and add a buffer configuration to Boom Recorder. Cheers, Take

Hello Sergio, I am very sorry that your sound card doesn't work in Boom Recorder. Could you tell me if Boom Recorder does detect the digigram vx pocket and if it does what exactly isn't working? I've one customer who uses the Digigram VXpocket 2, he had some driver related problems, something to do with not being able to change the input levels? I hope we can solve your problem. Cheers, Take