nchBen

Hi kneimond, Ok it sounds like you're facing a few issues here. The out-of-sync recording should *never* happen with the fast-track and it convinces me that it has a driver compatibility issue with El Capitan. However, there is a work around which may help in this instance. In the "General" tab of the Options dialog, there is a field called "Offset Recordings". You can enter a millisecond value here which will automatically apply an offset to your recordings to get them to correctly sync with your other tracks. If the latency is consistent then you might be able to get a decent result by figuring out how far to offset - it could take a bit of experimentation. As for upgrading to El Capitan, you are only one of many people I have heard complaining about unexpected breakages after the update. My laptop runs Mac, and after I upgraded to El Capitan I have noticed a *massive* decrease in speed Regards, Ben

Hi EdWheeler, Are you trying to burn the whole project to disk? Or just the music? A MixPad project can be *much* larger than just the audible clips you have on the screen. The original resources for each clip you add to MixPad are kept in their entirety so that you are able to re-edit clips from the originals and keep your project quality high. This can quickly add up to a large size. Are you wanting to move the project to somewhere else? Regards, Ben

Hi Kevin, Ok so in the context of audio hardware, latency actually refers to the internal buffer size that the audio hardware uses to record/playback music. The smaller you make that number the smaller the buffer size is and the smaller the latency. The lower the latency, the more processing your computer has to do because it has to swap buffers more often which is computationally expensive. If your computer is unable to keep up with swapping these buffers often enough you will start to experience issues like audio looping and skipping and jerky playback and recording. A higher latency is easier for your computer to handle, but it comes at the cost of adding a time lag to your playback. What this means is that when you are playing your track and you make some changes to something like the volume or pan or an effect, you will not hear those changes until the next buffer swap. If I recall correctly, the maximum latency setting on the fast track is only around 250ms or something like that. So even with it set to the maximum you are unlikely to experience many serious problems. Really low latency is most important for live performance products like DJ programs. As a note, it doesn't matter what your latency settings are in regard to having correctly sync'd recording and playback. That should work regardless. So we know we're on the same page here, can you describe the latency problem you are having? Regards, Ben

Hi AaronA, Currently grace notes are not sounded when playing back, but this will shortly be added. As for the layout issues you're experiencing, I will ask someone to have a look at this to see how it can be improved. Regards, Ben

Hi desander, There is no way to have this automatically done in Crescendo, but can I ask for clarification on why this feature is needed? Generally speaking, if you are composing a song in a certain key then you do not need to specify accidental (b or #) symbols for the notes which are affected by the key as this is implied by the key signature itself. Regards, Ben

Hi Gizmo, There are quite a lot of steps ahead of you, but you are always welcome to post here for advice along the way. The first step in this process is you need an internet connection. Without that you won't be able to stream anything. The second step is to read through the Broadwave help section and try to set up a basic streaming server that you can test. If you have any trouble following the instructions in the help section then post a specific question and someone will be able to help you. Regards, Ben

Hi ypdave, Has this problem just started for the first time? If so, have you added/removed any hardware or changed any hardware settings recently? Especially anything related to your soundcard? Or perhaps you may have done a driver update? Regards, Ben

Hi kneimond, I suspect the issue is mainly to do with the upgrade to El Capitan. The latest core audio drivers for the fasttrack don't mention support for El Capitan and the latency issue you describe is very likely to be driver related. When you change the latency settings on the fasttrack are you getting any changes at all? Regards, Ben

Hi Crazybiker, You can only apply effects to MIDI clips if you are using a virtual instrument (VSTi) to synthesize the MIDI into audio. Virtual instruments are modules that will take MIDI input and turn it into a standard audio signal that we can hear. Any MixPad tracks which are assigned to use a VSTi will also have any effects applied. If you are not using VSTi plugins to synthesize your MIDI, then you are likely to be using Windows' built in synthesiser (the MS Wavetable Synth). This synthesizer exists outside of MixPad's audio chain and consequently MixPad can't apply effects to this audio. There are plenty of places where you can download free VST instruments (just Google) if you would like to give this method a try. Regards, Ben

Hi ColeE07, Switch is only able to convert *from* dss formats. You can't output to this format. Regards, Ben

Hi petewatterschatsworth, In answer to the second part of your question, yes, both of those formats are supported. Regards, Ben

Hi Mitchell, This appears to actually be a bug. I have raised the issue in our system and it should be fixed in the next release. In the meantime the only workaround I can suggest is that you manually split the 16 hour file into slightly smaller (say 4 hours) blocks and then run the auto-split function on each of these. Regards, Ben

Hi Pascal, WavePad can only show the instantaneous spectrum so I'm afraid it can't do what you are asking. Additionally you are unable to display the grid values other than by using the mouse to hover over the area of interest. I've added your suggestions to our list of todo items for consideration in future releases. Regards, Ben

Hi vapors, Ok, so I think what has happened is you have installed the *trial* version of WavePad. This is different to the *free* version. The free version will always work and never expire although there are some features missing and you have to click a button at the beginning stating that you acknowledge that this version is for non-commercial use only. It should certainly never prevent you from saving your work. The trial or demo version will restrict you from saving after you have saved a number of files (I think it's about 10). Once this happens you still have the chance to save your work as a WavePad project, but you won't be able to export the audio to a specific format like mp3 or wav. Sorry for the confusion and I will have some discussions with people to see how we can make the fact that you are using the demo vs free versions a little bit clearer in future releases. Regards, Ben

Hi MayoPardo, Very sorry that the Lowpass filter is not in that list. It absolutely should be and I'll make sure it is there for the next release. If using the highpass filter is working for you then there is no real reason not to use it. At the end of the day what matters is that the result sounds how you want it. Regards, Ben

Hi kk4mds, iPhone ringtones are generally saved as .m4r files. You can do this in WavePad by choosing the .m4r file type when saving your file. Once you have saved the file you will need to import that file into iTunes. How you do this will depend a little bit on the options you have selected in your iTunes setup, but generally it can be accomplished by dragging and dropping your m4r file into the iTunes window. The final step is to sync your phone with your iTunes the same way you would normally do so, but making sure that the new ringtone is marked for syncing. You may need to do some additional reading on syncing ringtones in iTunes as the process can vary between different versions of iTunes. Regards Ben

Hi Twync, Sorry to hear you having trouble with the Mac version of WavePad. I agree, there are a lot of behavioural differences between the two. We are actually slowly changing the Mac version to be more like the windows version so that over time there should be very little difference. However, this is a slow process with incremental changes. I've made a note of the issues you have described, but if there are any other major problems then please PM me and I will make sure they get a priority boost! Regards, Ben

Hi Winfried, The closest thing WavePad has to this is to use its regions feature. It is quite simple and I think should do what you need. Here is the process: 1. Select a region you want to keep 2. Press Ctrl-R 3. Just accept the default region name and press OK 4. Repeat until all the sections you want have been processed 5. Go to the region list dialog and right click and select Assemble... 6. Select all the regions and click "Add" 7. Click Assemble. 8. This will create a new file with all the regions placed sequentially Note you can use the Options button in the Assemble dialog to choose the size of the gap which is placed between each region. Regards, Ben

Hi Andy the fiddler, A compressor or normalize is probably not going to help here. Your best bet will be to apply a notch filter at the frequency of the hum. Step 1: Establish the frequency of the hum To do this choose a section of the recording where the background hum is obvious and there is not much other audio being recorded in the foreground. Place your cursor in this section and then select Tools->FFT. A spectral graph should be displayed. The hum should be represented as a peak in the graph. Mouse over the peak to find out what the frequency is. Step 2: Remove the frequency Select the range of audio from which you want to remove the hum. Click on the Levels tab and click the Equalize button. In the Eq dialog select the "Visual" tab. Select "Notch" from the Preset pull down list. In the options box which should be displayed select the start and end of the frequency range you would like to remove. You may need to experiment with the numbers a little bit to get the best result possible and you it is possible that you will also notice undesirable effects occurring to the rest of your track. Unfortunately this can be hard to avoid if the frequency of the hum happens to be spectrally significant to the audio you are recording. Hope this helps. Regards, Ben

Hi djteoli, Crescendo can do this! Just open the midi file in Crescendo and then "Save As..." a cdo project. Regards, Ben

Hi livinlife000, The main problem with separating these two audio sources will be that they are potentially very similar in nature, and in addition, the TV source is actually in the foreground and a much louder signal. I would say that your best bet would be to open WavePad's equalizer and see if you can isolate and boost the frequencies at which the background conversation is occurring. You will have to spend a fair bit of time experimenting. Looking at the spectral content of the conversation may also help (do this using WavePad's FFT tool). Other than that I'm afraid I can't be much more help. It is a technically extremely difficult task, and I don't know of any other software which specialises in this type of thing. Regards, Ben

Hi livinlife000, Completely separating two sources is very difficult and in many cases basically impossible. You may be able to process the audio so that you can reduce the unwanted noise a bit, but this may also affect the audio you want to keep. Can you give me a better idea of what you have recorded and how? In your OP, you say that the TV audio is much higher. Do you mean that the volume is much louder? What kind of TV audio is it? Regards, Ben

Ok, Sorry I can't get this resolved for you. SoundTap should run on XP without trouble, but obviously in this case there is something not working right. The only other thing I would suggest is to make sure you have the latest drivers for your soundcard. Regards, Ben

Hi Ray, Currently playback is relatively limited in its ability and all dynamics are ignored. However, we are currently working on improving playback to include a lot of new features such as dynamic volume. Regards, Ben

Hi Ray, I understand what you are trying to do. I typically handle this case by doing the following: Does this solve the issue? Regards, Ben