Hi Ann,   Thank you for the reply. Sennheisers are in my future, but for now the Airpods are what I have. They're Bluetooth, so they work with any BT enabled device (and work great with my laptop). It's only when I have to manually direct PP to make them the output that I'm having issues with. Definitely a software issue on Adobe's part. By   Sorry MB, but the Apple support forums seem to imply that you require Apple hardware and/or Apple OS for these to function properly. It's a strange answer though, as the support person states that they 'see you want to use your Airpods with a Dell PC - I'm happy to help' before stating the following   Use your AirPods -- Apple AirPods will work with the devices below. Before you begin You need at least one of these: iPhone, iPad, or iPod touch with iOS 10 or later Apple Watch with watchOS 3 or later Apple TV with tvOS 11 or later Mac with macOS Sierra or later ... View more

How are your outputs assigned in the Audio Mixer? ... View more

To everyone. Please, please when posting can you give the following details: Operating System and build number This is absolutely vital, otherwise we simply do not have a clue where to start with helping. ... View more

Can you please elaborate on what you mean when you say 'progressive in time'? What - exactly - is the effect you are attempting to achieve please? (I ask as in all my years I have never once heard that phrase with regard to a reverb effect and honestly do not understand what you mean)   Tip: Turn OFF the current mode you have got your audio set to (rectified waveforms) and you will have a far better visual representation of the waveform (see attached screen edit) de-selecting rectified waveform ... View more

I would go as far as to say that hard panning to left/right with 2 people talking at the same time will be an extremely uncomfortable listen, if not outright irritating. Can I ask what the logic is here please, out of sheer interest?   ... View more

Firstly can I please echo what Ann says?   Secondly, I would always create a separate marker track for edits by not putting them directly into the clips, but instead above the clips. This allows you to move things as necessary, and once you are happy with everything then put markers in the video file for the chapter points on export - but this is the way I do things, and it may not work for you. See the screenshot - I initially put the markers (green ones) above the footage in the ruler, and only added Chapter Markers when this was all correct and checked. ... View more

Okey Dokey. First up, what is your OS, version of Premiere and what type of NR are you trying to do with what tools please? I would also highly recommend using something like Audition to do this type of thing, as it has much better tools & is designed for this and you can easily re-import the cleaned audio back into Premiere afterwards. The right tool for the job will always help matters - honestly. Do the mix first, then export the completed stereo file and bring this into AA, clean & massage it & re-import the results back into a new audio track in premiere - you will find life much easier. ... View more

It certainly sounds like a problem with the plugin itself to me. Have you tried contacting iZotope at all? ... View more

You really do not need to maintain the 32-bit float for the audio files at all - simply export to 24-bit fixed, as Adobe's use of floating point is a bit weird as you have options for both 16-bit fixed with a 16-bit mantissa or 24-bit fixed with an 8-bit mantissa. All DAW workstations will run happily at 24-bit fixed source files, and internally they will use 32-bit floating point anyway. Saving any audio to 32-bit floating point is simply a waste of hdd space - honestly it is. I have been doing this for decades now, and believe me it is the 24-bit that is important. ... View more

I know this is no help at all, but my money here is on Windows 10 causing all the trouble. I refuse point blank to put that dog of an OS on my working DAW system and this is why: Recently I had to buy a new NLE editing setup as the old system had a motherboard fail after almost 10 years, and getting a new W7 system proved to be impossible because it is allegedly 'EOL & Unsupported' (despite the fact you can purchse extended support direct from Microsoft) and I reluctantly went for W10 Professional because it was supposedly capable of allowing you to NOT install 'updates' - that turned out to be an Urban Myth however, as after only 2 weeks I was forced to do a full OS update because it would not defer any longer. On restarting I found my £2500 machine was effectively now a Boat Anchor. I called the system builders (this was not an off-the-shelf setup from a box shifter) who told me that this is common, and when MS do an OS update approximately 20% of the time it will fail and turn your PC into a boat anchor, and for that reason there was a little thumb drive in the case with the OS installation files. I had to open it up, take out the little drive & completely wipe my HDD and reinstall everything - including all the motherboard drivers and hope like crazy I did not forget anything. It took HOURS.   Other people I know have had their PC go into Mac Mode, as I call it - the OS 'update' breaks a load of 3rd party applications, or severely reduces their functionality & time is then wasted on the phone to various support people trying to find out how to get software I have paid for working again after MS broke it in the update.   So I am really sorry but my guess is the OS update killed you - but just in case, can you please let me know just what audio plugins are causing the crash? What type - VST2 or VST3? AAX32-bit or AAX64-bit? Have you tried deleting the unnecessary AAX plugins (assuming you do not use ProTools)? Where are the install paths for your audio plugins? The more info you can give me the more chance I havce of finding a possible answer for you ... View more

I am sorry to sound like a broken record, but can you please tell us the fiollowing: Operating System. Project settings Assets in use Export settings Anything else to help us try to isolate the cause of the problem, such as your sequence settings & assets in use would also help. ... View more

32/192 Apple Audio is snake oil.     ... View more

No, but what you can do is output with a CBR AAC-LC file (say 256kbps) from a lossless master. This is why whenever any edits are being rendered for the first time you should use either a ProRes MOV or a DnxHD MXF with PCM sound as your master. Not only can this then have any necessary post production work carried out on it before creating streaming platform masters such as Apple TV, Amazon Prime, Discovery Channel or YouTube or whatever is required. We never, ever render a streamable file from the edit project. What we will do is create the edit project from all the different clips, and get this done first. That will be exported as DnxHD MXF (I do not like to use ProRes as there can be gamma issues if you go cross-platform with quicktime ProRes, which is why Apple have deprecated it, but sometimes a client will insist on a ProRes reference for signing off purposes) for approval, and once that is done (we upload the full original to a private Vimeo account, each file password protected, & let the client view from there) we will create a new project for making all the deliverables. I guarantee to you that nothing will get bounced by streaming platforms if you use this method. ... View more

Apparently, 2244.mov was recorded with only the left audio channel.   In the effects panel search for "Fill right with left" (or the other one) and apply it to the clip.   Alternatively, you can change this in Clip > Modify > Audio Channels and then change the checkmark from right to left. By @Richard van den Boogaard   Not just the 2244.mov clip either, by the look of the screenie above 15 out of the 19 clips on the sequence audio track are left channel only, so there is either a problem with the recorder not correctly set up for stereo operation (either it is not capable of stereo recording or else it is set for mono) or else incorrect interpretation of the clips in Premiere, or something else entirely. My best guess - looking at the waveforms (and a zoomed-in screenshot would help immensely) is that you have plugged one mic into the recorder but left it in 'stereo' mode of operation and as such are only recording the left channel.  What unit is doing the recording please - make & model? Let's see if this can be set up in a more user-friendly manner, because what you do not really want to do is either A - duplicate content on right channel aka 'fill right with left' (this will double your percieved volume and also at the same time double your noise floor) and additionally this is now dual mono, not stereo - although if dual mono is what you want..... B - split the content across both channels, effectively panning to centre (this will cut your levels by 50% because of Pan Law etc - although the exact amount of reduction will depend on setup)   If all clips are being recorded through the same recorder with the same mic using the same sockets etc then you have an intermittent issue, possibly with one of the cables being dry jointed at a soldering point..... ... View more

I typically use a Multiband Compressor with the Broadcast preset on my audio clips in the Audio Track Mixer (using the twirl down at the top) and adjust the waveforms audio levels on the clips to further boost softer passages.   Hope this helps. By @Richard van den Boogaard On each clip? Why? Multiband compression - if used at all - should be used on the master buss, not the tracks and treated with the same care you would take with a loaded shotgun! These are seriously dangerous tools and can utterly wreck the sound especially if used additively........ ... View more

Hey Neil,   Thanks for your reply! I tried to use the 'normalize' function to make all of my sound levels go below red, although I found it didn't help very much. I'm working on projects that contain a lot of sound such as music, dialogue and sound effects and I'm trying to find the best way to balance all sounds as to not make it hit red. I find Premiere really hard to use when it comes to balancing/controlling my sound levels and I'm not sure what the best way to do this. Any tips would be greatly apperciated!    Thank you 🙂 By @Billy544645   Hi Billy. This is to be expected really, as normalization will never actually help you to improve matters - all it does is moves the baseline. This is because Peak Normalization (the usual sort) works to the highest transient (I refuse to use the word 'loudest' here because when talking about peak values, loudness does not really describe things properly simply because when we talk about 'loudness' we usually mean 'perceived volume' which is based more on RMS values, but that is a subject for a thread all of it's own)  and as such, peak normalization has very little to do with actual perceived volume. Let me try to explain why this is: If our highest peak in a waveform (these are what are known as 'transients' in sound mixing, and they represent an extremely momentary peak level of 1/48,000 per second) is already at -3dBFS (decibels Full Scale) then the highest we can raise the entire file using peak normalization is +3dB - now, this sounds like quite a lot especially when you realise that decibels are not a linear scale but a logarithmic one and a raising of 6dB doubles the 'volume' - but with peak levels we are talking about that 1/48,000th of a second at this 'volume' so in almost all cases your ears will not actually hear the difference at all because our ears wotk on a very close approximation of RMS, which is an average level rather than a transient, or momentary one. I'll try & explain this better in a mo. Additionally, peak normalization will also bring UP the noise floor by the same amount that it raises the level of the 'loudest' transient (it would be better if we did not actually think in terms of transients being 'loud' rather instead we should perhaps think of them as being 'momentary peaks' - this is where digital peak metering causes more problems these days than it is worth, and we should instead consider going back to VU meters as these represent the averaged level or strength of a signal - indeed, VU metering is actually an analogue form of Loudness Metering in LUFS! The only difference to my mind is that LUFS can be strictly calibrated whereas a VU meter can be set to many different standards in order to accommodate the enormously variable range of analogue equipment's ideal operating levels. (we love our standards in audio & video, which is perhaps why we have so bloody many of them?) I hope this has not caused too much confusion - if it has please let me know & I will reword as required.   In the meantime, rather than using normalization you should learn instead to either use a compressor (in order to reduce the dynamic range of the audio clip in question, which will make the peaks quieter and the overall level can then be turned up with either a fader or make-up gain bearing in mind that again this will also bring up the noise floor at the same time), or else an expander (also known as upwards compression). These operations will affect dynamic range but keeping in mind that the dB scale is logarithmic, adding 6dB of compression with the associated make-up gain will effectively double the perceived level but at the same time you will make any noise present louder too. There are always payoffs and paybacks to consider.   The best way to get great sound is to record it well in the first place though, and this begins by not treating sound as an afterthought but instead as 50% of your production. I can offer some pointers if you would like me to - all I need to ask of you in return is that you can tell me - in detail if possible - exactly what you are needing to do, as well as the specifics of the source material as far as possible. This is because there are no set procedures when mixing sound - none at all, as no 2 recordings are quite the same so they all have specific needs. ... View more

Just doubt if app store launch my video successfully with a approx. bitrate, it's a task for work & another department help with the launching & judge me with the specification ...so let's see By @ju22259367ig3g Your problem here is that anything using a variable bitrate can only ever provide an approximate bitrate to the app store. You need fixed bitrates with audio - this is yet another reason why lossy audio is bad for your general wellbeing & state of mind. What you need to do is extract the audio from these Hero devices and turn it into something nice & sensible - PCM. This will give a fixed bitrate, and you will find a lot of audio problems, glitches & upsets will melt away like the spring snow on a sunny morning. You will also have less stress in your life. The best place to start is with the devices themselves - tell them to use PCM WAVE files, not MP3, AAC-LC or any other lossy garbage. Honestly. ... View more

I feel your pain, I really do. Glad it seems to be sorted out now though ... View more

I'll dig into the Key Commands section but I suspect this will be quite beyond Premiere, and this type of thing is exactly why my preferential workflow is to do all my audio work in a dedicated DAW as Premiere's audio handling is a bit 'last century' with the clunkiness of it.  But I will go look - are you on a Mac or a PC? (Quick suggestion - whenever you are posting any request for help or assistance it is a good idea - no, a Great idea to provide as a minimum your OS details & the version of PPro you are running otherwise we don't know how to begin to help you out) ... View more

This is a common flaw with Adobe Software - it also happens in After Effects on Mac. If you do not regularly purge the media cache then you will run into all manner of problems. ... View more

Hey Kenny Sadly these screenshots confirm our worst thoughts, and the Hero8 & Hero10 are both recording your audio as extremely lossy AAC-LC whereas the GH5 is recording as PCM. What you are effectively trying to do is edit lossy sound in an AAC container, then re-render it as lossy AAC again. This is honestly a very bad idea. The best workaround I can think of is to do as you have done - export the audio as a WAV file - then re-import that back into the project to create the final output, save this under a new name and then remove the other audio tracks. This way you'll have a 'master sequence' with the audio as a WAV file and this should work.   I hope this helps ... View more

Try a far better AVC/H.264 encoder than the stock Adobe one. The Adobe encoder is based on the MainConcept engine, whereas the TMPG one is based on the far superior x264 engine instead. The difference is huge, and well worth the cost of the plugin! It does have a learning curve if you want to tweak/fine tune the settings via the 'advanced' tab but the effort you put in will be well repaid, I can assure you.   ... View more

Can you please provide the following information so we can try to answer? 1 - What is your OS - Mac or PC, what version? 2 - Can you post your sequence settings? 3 - How are you creating your sequence - automatically (new sequence from clip) or manually (File / New / Sequence)? ... View more

Have you by any chance updated Windows 10 recently? I think if I were you I would completely uninstall Premiere and then do a registry sweep, clean up any lingersing files & do a cold reboot and then reinstall from scratch. This is a serious error that basically means 'this system cannot cope any longer but we do not know why'   see https://www.windowsbbs.com/threads/what-causes-exception-code-c0000005-access_violation.40538/ ... View more

Personally, I would not even attempt to do both video edit & audio edit at the same time. That's really asking for trouble - try what Jeff says, export without pre-rendering and listen carefully to the output.   ... View more

(please delete this post)     ... View more

Hi Billy. Please, I beg you, do not use 'normalize' - ever! It is not a good process. What it is actually doing is adding a whole layer of utterly superfluous DSP to a file when there are far better ways to go about achieving the same result. If you could please describe the reason you think this is necessary (and I am an audio engineer, so please go into detail) I am certain I can show you a far better way to do the same thing but without using an automated algorithm that should frankly be avoided. ... View more