Of course not?  Adobe is now using a modeled pro-res 422 encoder (based on some FFMPEG stuff).  Unfortunately, you usually have to have the quicktime player with the bundles that go with it in order for any of the software to even recognize it. Use the Quicktime preset from the list, then, down below,  you'll see "encoder", select pro-res from that.  You can then set quality, and compression.  However, Pro-res is only a stop-gap format anymore.  A lot of colorists swear by it for moving between systems because they bought quicktime and have the full encoder, with the presets installed. Most colorists are Moving to DNxHD, as it is very close to Pro-Res in color standard and it lacks the overly vibrant or blown out reds.  Cineform is another great alternative, but is primarily for quicktime based edits.  IF you started with an H.264 file in a windows format, you're best off going to DNxHD and handing that in.  You can usually set the extension to MOV on your own afterward, and it should still be read much the same, through the quicktime codecs. There is also an FFMPEG bundle that has a Pro-Res 422 preset that you can target.  However, you will have to look that up, and download the packaging, then run through all the extensive setup for tying it to your encoder list in AME, which should also make it available in other apps (most of them draw on the list from AME). ... View more

Unfortunately, working across versions can have a bug or two.  The source range modifier (I'm assuming in export dialogue as that is where Source Range is defined) is connected to your cached previews and work area in some way, I don't recall exactly how.  If you move from one version up to another, you'll have to: 1. resave the file as the new format file 2. Relink the datafiles so that the new version can grab them properly 3. save the project in the new version at least once and make sure the work areas are set If you're talking about subcliping in the source window, you are doing a source selection range, and this is accomplished with the "i" key for in and "O" key for out points.  You should try using those first.  You can clear them also, check the menus for CLIP etc.  It will show you the keypress for clearing the selections. ... View more

Sorry I've been a little absent lately.  Just moved over to the Windows side... ..Sorry everybody... ...Had to go cost effective for a superior power machine. However, encore still works, and quite well.  The downside... ...You have to render out compliant video, you have to conform it to 22gb or less as blu-ray is not actually 25gb unless you get a specially made disc or higher quality media made by a decent factory.  Remember, Blu-ray out of encore is not AVCHD compatible.  You have to render it out to older standards, of lesser bandwidth.  If you want to create an AVCHD compatible disc, you sure can.  It's possible.  Just don't burn the thing with encore.  Send it out to a disc image, then burn it with your favorite burning app. Encore worked great but slow on my old Macbook pro2009, which still functions by the way--the screens blown, but the rest of the machinery's still intact--and while it took a day, sometimes two, to render a disc, while it spit out cup-coaster after cup-coaster, I still used it.  I found out about the disc image trick by playing around with it.  Burning with my system was unstable at best, so I had to move it to another computer or use a burning software to get it done.  Roxio was great.  But even some of the free burn utilities can do the job.  I suggest you look up the standard you are trying to match, then match the settings, render in media encoder to video only, and to audio only, then link them in encore and you shouldn't have a problem.  The only downside is that it won't link to timelines anymore in premiere.  Though, you might be able to fake it, if you output to an xml format cs6 can read, then open it in premiere there, and then export to encore, but that's more work than it's worth. ... View more

Open the BDMV file in quicktime.  You'll get a box asking which to open, pick one, then file and SHARE to another format, you'll get a proper video file.  Takes some time with a lot of clips, but it works every time. ... View more

Hey, bro, you preach to the choir.  I used the adobe "frame blending" technique back in the days of CS6.  It took care of the frame rates and sync issues.  Today, it actually just reads the change in frame rate and does nothing!!!  Well... ...unless you change the format, the data rate or some other major factor; alter these and it will reformat the video entirely, rebuilding the frames. The Quicktime or Compressor version of this operation is much easier, and they build better frames overall (motion adaptation algorithm is great but use as light a touch as possible; 1-10%). Right on. Of course this is akin to INGESTING your data, which is just good practice IMHO. ... View more

That's not a fix.  That only works if the MP4 file is .MOV compliant.  Not all cameras actually produce a fully compliant file.  That's why the more accurate and permanent fix is running an INGEST type conversion. Ingestion of data has been a part of the workflow since the beginning of digital formats, all the way back to the beginning.  You could capture with anything, but you had to run it through a machine that ingested the signal into your editor program, then run the final product back out.  Compression wasn't the issue, playback was.  The linear edit styles had editors running the video hundreds of times, creating lists of edits, then running the output to a master media that was later used to create new copies. Today, there are many formats that differ only slightly in their algorithms that compress the video and the full frames that are used to compress the others.  The algorithms used, the variable settings and the chosen values for those settings are all factored into the algorithms.  Those settings play a huge role in how compatible the actual data is with different wrappers.  One camera might create files you can just change the extensions on to make them work, where other cameras typically don't--in which case, changing the extension could completely destroy the usability of the file, or just make it unreadable until you change the extension back.  Ingestion of digital files for nonlinear editting is an essential part of the process.  Different cameras will have different color palettes.  Pre-grading angles is the first step, but can be done by simply converting all your shots to the format with the smallest or most narrow palette.  The next step is balancing them so they match, and finally you convert those to the format you'll use for edits, and the format you'll use for output.  Edit formats are smaller compression and faster playback with lower quality, but the final product versions need to be as full-format as possible, so they don't lose quality in recompression of the output.  This also removes the step of decompressing the edit files to compress them back again, and saves hours upon hours of render time.  This all turns a weeklong job into one that takes about 3 days.  That's just for the video of course, but you get the point. ... View more

Ok... ...So I've confirmed my issue isn't premiere itself.  It's in the way the audio is moved to Media encoder.  When I export from premiere, all's well; with queue to media encoder, failure to capture audio. Tests: I ran audio only out of premiere and AME.  The files were exactly the same size.  I played both in the same players with visualizers etc.  The one out of PRPro worked flawless.  The one from AME is completely Broken. I ran a full file from premiere, and viola, it plays. To Note: AVCHD was designed as an alternative to Blu-ray formatting.  Depending on your camera, you can end up with some horrid data problems.  Image every card before you start.  Then INGEST THE FILES FROM THE CARD ITSELF (try it with the camera if all else fails).  I also recommend finding apps like iMedia HUD, and checking the formatting of your AVCHD file first.  Then Ingest and match it.  DNxHD works well for large file size and super quality.  For lesser work, h.264 will work also, in your native wrapper format (macs use mp4, pc's AVI).  If your footage is interlaced and you want it progressive, use Compressor by Apple.  So far, their motion weighted algorithm is still the best. My project includes an AVCHD transcode where I forced another audio to merge with it.  I probably bandjaxed the whole thing with that. But Prelude wouldn't see my files correctly either (my AVCHD).   Somewhere along the way, I must've screwed up.  This is an old project from 2014, and a client wants clips for a mashup\montage.  I still have it all marked and cut.  I seem to recall having a similar problem outputting to a video file for burning.  In the end, I output the video with AME and audio with premiere, tied together and put through encore. ... View more

Congratulations man.  You chose what I and the rest of the world call the Right way. MTS files are Multiplexed Transport Streams.  If you remove them from the AVCHD context folder structure, especially with some older AVCHD variants, you will destroy the linkage.  For some reason, this makes audio output impossible from premiere.  I had to export the two files separately then reattach on my output stage.  This is a preposterous waste of resources. Prelude will only view AVCHD from some cams if the camera is connected, forget a disk image.  It won't work with older AVCHD.  It gets blocks through the camera, which translates them to another sizing before moving the files (i mean data blocks).  It's a bit like the old macrovision schema in that respect; a standard block size is filled with the data, but random areas are left clear when being blown up to a larger block size and a computer or Java device doesn't care, it only reads the good blocks, as the copy is still good. Quicktime now doesn't care where a file comes from as long as the data has a format list (file header) that matches something it can play.  Because quicktime sees AVCHD similar to Final Cut, as a footage file with markers for clip breaks, it will read the file just fine, catching the MTS data and properly reading the overlaps, as well as the clip points.  MTS files usually don't break 4gb, due to the disk formats used in cameras, and the datachunk memory.  However, smaller MTS files may actually be separated clips.  Unfortunately it's not always the case.  Depending on your camera, you may get 2 and 4gb data chunks, only 2, or only 4.  I've seen it happen.  Quicktime reads the data properly from the MTS linkage.  You can output your clips from there, and them ingest into premiere compatibles.  There is one downside.  If your footage is interlaced, it will force a top field first conversion, dropping quality by a massive amount. AVCHD formatting: Multiplexed means they are combined audio and video, and they are similar in design to the old VOB datachunks of DVD's.  The only real difference is that they hold a different compression size and overlap structure, along with a different clipping mark style.  In VOB, you could have clips that are separate videos in the same chunk, and there was a marker there to show the break.  In MTS, you simply don't add the overlap mark at the end of the file, close the file chunk right at the clip point, and start a new file. To fix interlace problems: Yes one more step to get the interlace back in order.  No you cannot simply reencode it.  It won't help.  You'll get a bigger mess.  I've tried it.  The best method is the Field Splitter method.  You put the video into two video tracks, right click the top, set to upper, right click the bottom and set to Lower in a LOWER FIELD FIRST timeline.  For most 1080i, you're at 29.97 frames per second, so set that, but set interlaced lower field first.  Output that video to something and it will look a little better. The only real one step fix is a dedicated AVCHD transcoder.  THey don't care about the camera blocks, they just grab MTS files and output to a single long file.  You could use final cut, and send a transcode to compressor, with the correct settings.  It is still the only way to grab these files correctly. Where do I get all this? Study the files, study the formats, study on the internet...  And in libraries. ... View more

This is an ongoing problem.  I've exported the same AV output over and over.  Somehow, when you try to work with AVCHD, if you don't import it into the program normally, through the Media panel, you break the audio functions.  IF you bring in the MTS files on their own, it breaks.  The AVCHD file supplies information on how they are synchronized and how much they overlap, along with several smaller datachunks that hold that overlap.  Otherwise, you get blank audio in some areas that cannot sync samples properly.  Rather than simply run the audio through an index, creating an output that is useable, it just outputs data with no linkage (It outputs the audio data, but it isn't readable by anything). The only remedy I've found is to export both separately, and cut accordingly.  You'll have to create your own WAV or other standard audio file.  For long video, I've found this troublesome, as WAV can get large enough to not function properly in Premiere.  You may have to go with MOV.  If you have trouble with audio, you should replace it with a completely different format (relink to a different file extension with sound data).  Once done, you can link back to a WAV, but one that has been repaired.  Unfortunately, you'll still have to recombine audio and video. Alternatively, Import your AVCHD files, Clip them up in sequences or subclips that match your MTS sizes and timings, and replace the MTS files, along with their audio.  This takes a bit of time and hastle.  You could also cut them into sequences, export those sequences, and relink your MTS files that are broken to files that are not.  Again some extra time and hastle.  But a possible fix. ... View more

Another update for you... Older versions of AE use the basic 32bit version of Quicktime server.  This is currently not compatible with the newer codecs.  Newer adobe versions do function correctly with new prores codecs, provided they are installed correctly, and that the presets for media encoder are installed.  These presets are linked lists that forward sequence and comp data through the codecs.  There may still be a copy of the 32bit server for older versions of the video files, and it might work with older encoded files, but I haven't seen any kind of reliability with it.  Using only the new codecs has proven to be the best method.  They will work with older files as well.  On mac, if you purchase the advanced codecs, you'll need the presets that match in Adobe Media Encoder for reliable operation.  On windows, the apple codecs are unreliable on their own, but you can fashion your own from FFMPEG.  Installed first, FFMPEG will allow you to fashion some settings for prores.  Here you'll have to look it up yourself.  Once you've done that, install your video software (premiere, After effects, etc).  There's also a preset that lets you link to FFMPEG in Adobe media encoder, running the linkage through FFMPEG and placing the files in the Broadcast folder under Apple Prores, for the basic output styles.  For 444 camera data, you'll have to recode to 422 for use in your apps. Note: Some have reported great success with windows and the HQ codecs bought from apple. Here are the basic requirements-- 12gb ram, non-nvidia graphics card of 4gb, intel i7, very powerful cooling, RAID work drives on sata3 or faster, SSD internal OS and APP drive.  Adequate cooling--3 fans at least. Anything less will cause problems.  If you want to make sure of performance, 16-24 gb of ram and at least one xfire graphics card at 6gb will almost guarantee performance.  More cooling will help, as well as maintenance. Nvidia cards have difficulty if you install cuda, and it doesn't appear to be limited to macs.  If you use the advanced apple codecs, the programming in cuda can cause problems.  OPENCL does not.  You can use nvidia cards if you don't utilize cuda.  XFIRE on the other hand is more of an OPENCL threading and worksplitting architecture.  With multiple cards, you increase performance. ... View more

Would you like to know why? MOV processes missing frames automatically when in h.264 mode.  I've seen this happening with MP4's from iPhones.  WHen I run them through QT, they play fine, then I can transcode them almost instantly because they're already in that format, just wrapped in the MP4 extension. For your camera, Perhaps the MP4 file metadata places the reframe marks in the correct spot for MOV, but not for crossplatform MP4.  Both are apple-originated formats, but MOV is a QT proprietary, and some cameras confuse the metadata order when writing the file headers.  If they are out of order, or the values are written in a different number format, you will get problems with the file when sequencing it.  Be careful with this advice you gave.  It worked for you because the format was written as an MOV variant, but wrapped in an MP4 extension.  It won't work every time, for every instance of this issue with different cameras. The BEST advice is to process in the manner in which YOUR SOFTWARE SUITE EXPECTS.  Premiere expects video without data issues, whether elements or PRO.  Start with a TRANSCODE.  If you want to keep the same format, go ahead.  IT should be extremely fast if there are no issues.  In some cases, some motion is actually processed better do to a recode from the I and B frames that are used to calculate what to drop from the compression.  Set Frame Blending to on, to reframe where necessary.  You should also set the data rate higher than the original by at least 0.2mbps so it will apply the quality adjustment necessary to calculate missing frames and proper blends. Apple Compressor uses motion adaptation for this and they've kept it proprietary.  It often does a cleaner job, but takes longer.  I only use it for deinterlacing and blowups I don't want to do in photoshop.  If you're shooting SPORTS or HIGH MOTION, go with compressor for clarity.  Everything else looks pretty damned good with PRELUDE and ADOBE MEDIA ENCODER.  Short clips or SD's I want in HD, I usually use PHOTOSHOP for because I can use plugins that retain detail using VECTOR MAPS (My whole lab's still CS6).  Using vectorized mapping of detail, you can blow something up to 300% without a great drop.  However, going from SD wide to HD is just 167%.  Some plugs are even action scriptable, so you can apply the same settings to every photo of a folder.  In After Effects CC, the Detail retaining upscale actually does something very similar, applying some unsharp masking, and microcontrast adjusts to make the whole process more natural looking at the end.  It's pretty good for a one step uprez at 133% (Or 1080 to 4k, 720 to 1080). ... View more

A year old, and still an interesting thread.  Mind if I chime in? I too am thinking of trying this with several motion cameras.  If I can output the footage at the same rate and push the record buttons simultaneously (wireless remote anyone?), they should line up. Process: First, output your two folders of images.  You can do this from after effects, premiere, media encoder, prelude; it depends on what you shoot with for what path you'll have to take.  If you have OIS on (any image stabilization really), you won't be able to do this with every set.  It will fail once for every reframe.  If you think you can stitch these yourself, try.  You might also try reframing them yourself, then stitching.  It will vary a lot more with the quality of your tripod, and the quality of your camera. My way: Prelude to images, find reframes by playing at a slow frame rate and stopping in after effects or premiere.  Noting which frame you're on, you can merge two frames to create a new one, or even overlay data from a good frame from the other camera.  Once you have your reframes marked and redone, you can begin stitching sets of images.  This is unbearably long, but if you know any extendscript, and can read the phoshop script for stitching, you can probably have it repeat the stitching as you need much the same as any other action.  It is already a batch action by nature, but if you were to alter it slightly with nested looping, you could have it pull from two folders and stitch two images before moving on to the next.  You'd have to save the images with the same numerical name to another folder of images that has twice the space. They also have to overlap about 1\6 to 1\3 of the image.  I'd love to do this with some shots.  I could turn medium distance angles into wider angles without much work, and I could still use the closer angle at native resolution. ... View more

     Okay sports fans.  Here's the deal.  DVD players have a 9.4mbps limit on combined audio-video for DVD.  Blu-ray on DVD can get 15mbps.  Balance that out between audio quality and video quality.  If your audio is 256kbps (decent audio for most purposes), thats  0.25mbps and will leave you with 9 as the max setting for DVD (you might say 9.2, but its more like 9.14, and I just round it to keep some distance to prevent the errors that will undoubtedly appear).  When you encode for DVD: Your TARGET (or average value over the whole video) should be .75 to 1mbps lower than the max value you set, while your lows can be anywhere, but are best at .75 lower than your target (if you have this minimum option).  This is an average of your non-motion frames, which will keep clear quality within 2mbps of your max, for slower action (some stage performances and such will work great with this; for sports, keep your minimum within 4, and it will actually keep base clarity with your faster motion, keep your target closer to your max, and all motion will keep general look and feel). The whole point here is that you can set whatever bitrate you want and get great quality files, but no DVD player will push past 9.4mbps.  It's a hardware limitation of EVERY MODEL.      The best way to get great results with encore is a 2 step encoding.  First, output an extremely high quality file in any format that has a high quality, like 30-40mbps max with a split of 5 for the other sliders.  If you use mpeg2, you'll get a HUGE file.  However, if you divide the data rate, you'll divide the file size.  Check your average data rate.  This will tell you if your file can be compressed well.     IF you set a 40mbps max and a 30 min, and your average rate is between 30 and 35, you can compress this file well down to 8-9mbps at 1\5th quality of data.  Now you can do the same rip, but use the highest quality file (if that's your source, which is unlikely, as it's compressed to a lot lower data rate), go from your sequence; if your new file has a higher data rate than your source, it will look the same quality, but compress with BETTER quality, as the compression will be more accurate to each frame, and should take less time.     There are 2 definitions to contend with for the term quality.  The first is DATA Quality, the second is VISUAL Quality.      Why? With compressed data coming into your sequences (most consumer cameras compress the capture to proprietary formats), you're seeing a 94-98% accurate file (just checked an online listing for AVCHD qualities).  When you blow that data back up, you get closer to 94-98% per frame (There is some distinction you'll see in a minute).  Since each frame in your source (camera data) probably has less data built, the rest needs to be guessed, and reformatted, so it will very likely be a less accurate calculation.  When re-encoding between formats, the decompression step will USUALLY do bare minimum, and will cut decimal data down severely as well.  This results in lower precision and accuracy of data to format to the new file.     IF you expand that data first you cut down on the inaccuracy.  Also, moving to a compression within the same file format reduces the inaccuracy in the compressing stage, mainly because you're not rebuilding the frame, or reformatting--you're just dumping more of the less relevant data (pixels that generally don't change between frames need only be in one frame and left out of the next).  Because the math of the compression math and other formatting are kept the same, they are inherently more PRECISE.  Thus the file shouldn't APPEAR to lose as much quality as it would if you went from your source.      Just note, I said APPEAR.  It will generally Appear to have better quality VISUALLY, but be poorer "quality" in the accuracy of what data was actually captured by the sensor in the camera (there will be differences, but you should have more precise values for the data between frames, rather than a more "Perfect" match to capture, and the visual will be smoother in small details, like skin, but sharper at edges where things change more).      It will lose a little more data quality, mathematically; but 98% of something at 98% is above 97%, Do that again, but compress down to maybe 90%, an you get more like a 90-91% overall either way (source or decompression).  The excess loss may seem unnecessary, but if you have compressed source, you're almost always better off decompressing coming out of edits or going into them; it's not about whether or not you lose anything, it will happen no matter what. It's about where you lose it.  If you lose it in the rebuild of each frame, you have a low grade frame going to lower grade frame.  If you have a more complete set of data, you get a higher grade frame being compressed to lower grade. Lastly, only because I've pontificated about DVD playback, and you may want to encode Blu-ray or high-def video: To encode Blu-ray data to DVD, you can build a 15mbps blu-ray quality in h.264 (combined audiovideo data rate, by the way), create the Disk image in encore, don't burn, and then use software like TOAST to burn it to a DVD.  That disc will play in many blu-ray players.  Why only 15mbps?  The read spead of a DVD in a blu-ray player is limited to 15mbps (about half normal blu-ray rate due to reflective properties and track distances); what this means is that it can play any dvd at normal rate, while blowing it up to size, but play blu-ray data from a dvd slower than normal blu-ray due to laser movement time being about double. PS      Set top playback does the REVERSE of any PC youtube screenplay.  A tv has one input going in, meaning you can remap the output without affecting the playback of other inputs.  Not so on a PC.  Look up PPI, pixels per inch.  Generally, video conforms to its pixel height x pixel width \ Pixels\inch where pixels\inch is a value of 72 on playback.  If you adjust the 72 down to say 24, you've blown everything else up 3 steps.  That turns DVD resolutions into fullscreen height size for a TV, without really losing much.  Some things will be smoothed out too much but the TV usually adds some noise to clarify the image.  Visually, it's difficult to tell the difference; you only see that it's bigger, and maybe has a bit more noise added to sharpen things a bit.       On a pc, you have many other apps to contend with, though some create a virtual screen in gfx memory to overlay onto the main screen, shutting out all other inputs, which is how they can play a dvd at a similar quality to TV sets and it functions much the same. ... 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Okay sports fans.  Here's the deal.  DVD players have a 9.4mbps limit on combined audio-video for DVD.  Blu-ray on DVD can get 15mbps.  Balance that out.  If your audio is 256kbps, thats  0.25mbps and will leave you with 9 as the max setting for DVD (you might say 9.2, but its more like 9.14, and I just round it to keep some distance to prevent the errors that will undoubtedly appear).  Your target should be .75 to 1mbps lower, while your lowes can be anywhere, but is best at .75 lower than your target (if you have this minimum option).  The whole point here is that you can set whatever bitrate you want, but no player will push past 9.4mbps.  The best way to get great results with encore is a 2 step encoding.  First, output an extremely high quality file in any format that has a high quality, like 30-40mbps max with a split of 5 for the other sliders.  If you use mpeg2, you'll get a HUGE file.  However, if you divide the data rate, you'll divide the file size.  Check your average data rate.  This will tell you if your file can be compressed well.  IF you set a 40mbps max and a 30 min, and your average rate is between 30 and 35, you can compress this file well down to 8-9mbps at 1\5th quality of data.  Now you can do the same rip, but use the highest quality file (if that's your source, which is unlikely, as it's compressed to a lot lower data rate), go from your sequence; if your new file has a higher data rate than your source, it will look the same quality, but compress with BETTER quality, as the compression will be more accurate to each frame, and should take less time.  With compressed data coming into your sequences, you're seeing a 97-98% accurate file.  When you blow that data back up, you get closer to 97-98% per frame, rather than trying to decompress and recompress on the fly.  Since each frame in your source probably has less data built, the rest needs to be guessed, and reformatted, and it will be a less accurate calculation.  Moving within the same file format reduces the inaccuracy, and using a file that's less compressed (more data built into each frame) takes less time to calculate the rest of each frame for recalculating the corresponding new compression; in fact, it only need examine existing data, and reduce it for each frame by the factor necessary.  Thus the file shouldn't Appear to lose as much quality. To encode Blu-ray data to DVD, you can build a 15mbps blu-ray quality in h.264 (combined audiovideo), create the Disk image in encore, and then use software like TOAST to burn it to a DVD that will play in many blu-ray players.  Why only 15mbps?  The read spead of a DVD in a blu-ray player is limited to 15mbps (about half normal blu-ray rate due to reflective properties and track distances); what this means is that it can play any dvd at normal rate, while blowing it up to size, or play blu-ray data from a dvd slower than normal due to laser movement time being about double. ... View more

Audition really doesn't work in BPM terms.  It works in samples and milliseconds.  If you have a track with BPM information, you're getting it from a DMCW not a general DAW.  There's a big difference here.  DMCW is for Digital Music Creation, and is in fact a type of DAW all it's own.  A general DAW only deals with recording live audio.  However, I believe you can use a metronome signal and synchronize the count of your metronome to the music notation.  You'll need at least 2 full lines of music (4-6bars each) and you'll have to time your metronome for that beat, then match the sound to the metronome marker by speeding it up or slowing it down to match.  Generally speaking, this is particularly difficult to do if you don't have another metronome signal.  It's best to always record one.  The easiest way is to have a drum major or the vocalist, tap the beats on a keyboard plugged into a software instrument.  IF you can see the drum major, you can keep your count from the control room by tapping it out.  You can use several different notes to indicate different counts.  If your board is midi, you can save a midi file that can be used for a metronome, and the data is much smaller.  The difference is that MIDI files aren't useable by all DAW's.  You'll need an "instrument" to play the midi, and create the sound data from the keyed notes before it can be used by Audition, and even then, if you haven't got one for both files, it's difficult to match.  And always remember, if you speed up or slow a sound, some of the imperfections can get so much worse...  Some say compression and attenuation before the speed and slow opps.  Before speed is okay.  Before slowing? No.  You'll just make the artifacts and slight noise longer.  Slow it, then apply the factor to your compression timing.  If you drop to 50% speed, double your compression attack\release, then drop slightly by 1 or 2 ms.  I wonder if anybody could create a midi that senses conducting movement? I've had to match the counts for some videos being played at performances with subtitles using nothing but dvd player\projection.  I did it by recording a rehearsal, matching the notes to timing, then adjusting for a broad count measure, like a quarter note in 4\4 time, and finally, counted through the music, to get the starting point of each phrase, offset by 10 frames before (preemption) and placed the title.  When I was done, It felt like the titles were spot on, even though they preempted the sound in my video mix.  I added a lead in to get the conductor in line with it, and any slight variation, natural or otherwise was undetectable.  It went over extremely well.  Just an example of trying to get the bpm to attach.  I've also done it with video that was already created.  I had to get the video to play at the speed of the song.  Similar prep.  Start by getting a count from following the sheet while listening to audio, set up a burn in counter on a video track with only the audio playing.  At the last note, or last count, stop replaying the counter, bring the last one up onto another video track, place the video on video track 1, and see how much longer it is by count mark using the upper area.  Divide upper markers by number of lower.  Add to 1.  This gives you a percentage based value to work with.  Your video is This Percentage of your audio.  Now you can adjust your video timing to fit using percentage or time\frame.  Divide the time by this number to get an exact time of the video, plus or minus some framing...  I've gotten similar results.  You can do the same with just audio.  Find the starting point, place a marker, find a decent end, use that for the size of the marker, divide the time in milliseconds by the full beats (in 4\4 there are 4 to a bar, get a general idea).  Do the same for the next file, compare the times of each songs single beat and full bar (if in the same time), or the same number of beats if they are in different time.  Speed or slow one song to the other based on that analysis If song A needs to Adjust to song B, Divide the time for the same number of beats in song A by those in Song B.  Remove the decimal.  Set the speed of Song A to this percentage.  If you have a 0NN value, think 0N.N as the value.  Say song A has 1.3s for a full bar, song B 1s.  A is 130% of B for time (1.3 divide by 1 = 1.30 and remove decimal--> 130, if it came to 1.033 id go 103.3) , so I need to speed it up that much to match the timing.  Now the BPM should match fairly close. You can vary the speed by 1 or 2% to get it closer. ... View more

Thought I'd throw my half a cent of whatever in here. AV edits are a'changin' .  There's a lot more going into the project files, while not actually changing the starting files.  I find that its great to mix some old school with the new. Old school ways, you copied the original several times, using the copies to cut different portions together.  Today, you just list your editing and then run a program to copy the data, adjust, output.  Simple.  When you did multicamera mixes and multisound source cutting, you had to be even more careful.  If you made a cut and had to change it a little, you had to get copies of each clip in the section and recut them again.  EDL came along, and it was a masterful idea; different operations in different programs could be written into a file with common jargon\language, and be read by other programs to show the preview output.  When a program lacked a function, you could read the decision list, see what program created it, know when to apply that effect later, and send it down the line.  At the end, each cut could be mastered by one or several programs as it went through its final stage, and come out on the other side with a finished product. Today we want to move everything we do in one program to everything else.  Though they use a similar language to build their project files (the list of operations to perform on the data), they aren't perfectly interchangeable.  The Tags used to describe each operation are very different, as well as how the operation is described.  HP and TI calculators have differed for many years in how they process mathematical operations.  One processes them the way we do normally, the other in reverse (RPN).  In recent years, however, they've become very much alike, only differing in what they call particular operations, and how they display their functions on screens.  I could program a mathematical text on one, but it wouldn't be recognizeable to the other.  Video editing has gone a similar route. Now we also want the older programs to open what the newer ones made.  Its common to think that new versions simply build on old ones.  In many ways they do.  However, the languages used to describe similar functions may have changed, or the way the information is layed out, or their may be a new set of functions that uses the same name as the old while the old has been renamed.  All of this creates a problem when old programs try to open files created by the new ones.  With standard text in unicode, you cannot open it in an ASCII only editor and get a perfect rendition of the same text.  You may have to pass it through an operation to convert the text over to another format.  That's the old school way. Why the hell is this all so important?  Let's take a look at some repetitive tasks you want to move. For moving Multicam sequences, I find it easier and often more rewarding to simply not try to move them.  You'd have to flatten them and all kinds of other stuff just to get them to move.  If you have a fast enough system, and are good with your older program, you can run the multicam cutting in the old program.  There's always a few seconds of shooting before a sync with a clapper or other loud noise.  Cut all your video the same length by cutting off some at the start so their start points are all at the 0.  If a camera has extra video beyond the others you want to keep at the end, put in black video and fil audio on the other tracks until they reach the end of the long one.  Select all the video\audio on each track one track at a time and copy\paste it into its own sequence.  Now you've handled the sync.  You can export the sequences as Self contained movies, and you won't have to sync them again; or you can export the sequences as xml\edl and try it that way.  In this fashion you can get your sync rebuilt in your old program, and run the multicam in it very quickly.  Old school ideal: USe the SYNC point to create a common 0 start point (that's what they use the CLAPPER for) and then separate the tracks into their own reels again until you get to the old program.  Put them back together again and run the multicam function. If you prefer a more new age approach, and you've got some time to procure some repeater programs (bots), you can get a program that watches your keypresses, then run the multicam in the new program and have your repeater do the same in the old program.  It should run your keypresses at about the same time intervals, which should render the same multicam sequencing.  This also works for markers, and other effects so long as you use your keyboard to actually get there.  If the window layouts are the same, you might get away with using the mouse for some things as well.  New School: You do it once, the machine can do it a thousand times after. XML or EDL are really the only way to do this stuff quickly for most operations.  But if you've already made a multicam sequence, get yourself a copy of Premiere Pro CS5 or CS5.5 and try the quick tricks of changing the version number, offlining, then opening up and relinking all your media.  You won't have to worry about the multicam. For automating it on the mac, you can use Automator to WATCH WHAT I DO, then have it repeat the actions at the same speed when you run it.  Then you just have to prep for the recording by having the app open, the correct panel open, the record button ready to fire and set to a keypress; when recording use the dock to bring the app front, then start your work and keypresses as you go through the motions.  When you finish, stop recording your multicam by keypress, and stop recording your actions by clicking the stop button on the small widget it brings up.  Run your action again, you'll get the multicam rebuilt in your old app (so long as the shortcuts are the same).  I'm sure theres a similar method on the PC for copying your exact actions at the exact same time intervals.  In code, you could start a timer, and when you push a key, log the time and the key.  In java or JSCRIPT this is simply a forever loop with an event listener.  When you press a key, an event fires, logging a loopcount into a file with the key after, jumping out of the first loop if the event is a special escape key or mouse click.  Not exactly rocket science is it?  You can get very high loop counts if you don't reset the counter.  Reset the counter to zero each keypress and you'll never go to high.  If you have a timer function that counts milliseconds, you can have it count very accurately without getting numbers that are too high to represent with a small memory footprint.  Count the values in the file, build an array of those values, and loop through the array (faster than file reads), and you'll get a very accurate repeat performance of keystrokes.  The last values of the array may be your STOP COPYING ME routine, so you can omit them from your looping, play a sound and exit the program. I don't know the exact code, but this method works. You can do the same for making your blade edits and such, essentially rebuilding the file with markers etc, then add any effects based on the markers.  Using a keyboard to scrub and apply common edits or operations can save you a lot of time if you record the actions and plan your work a little. Old School ideal: plan your work, work your plan. If you do your work in passes, you can record yourself doing it, and have the computer reproduce it exactly in an older version of the same program or in a different program, while you work on something else.  By that method you can transfer most of your work easily where others might think it impossible.  The pros know you gotta be OCD about your notation (in this case, recording everything you do in a machine reproduceable form) if you want to do less re-formatting of the work yourself. New Age Ideal: Programmers want their programs to be different\better than the other guy's.  They use formatting of binary data to store the instructions for what has to get done to a video, and then have to have that language read back into something your renderer can use to apply the changes.  This may sound like simply translating one language to another.  It happens twice.  Once from the Proffessional jargon to the programming data layout, then from the instructions in the data layout to the renderer's operations.  Programmers are people too, and can write their languages anyway they chose, even if there are some common accepted rules and schemes.  They also have artists and visualists help them design a way to present the information on screen so it can be read by the professionals the program is designed for.  That's another language.  Finally, there's the prospect of cross-compatibility.  Since they perform some operations in a different logical manner, they can't always translate them right over.  Often times they have to settle for a function that comes close enough.  Close enough is where they usually draw the line, as they'd never be able to work on new projects or make any money if they couldn't release an old one.  Having a common format to target makes it easier, and having a common limitation means it will usually be acceptable.  I know this is a lot to take in, but there are ways of working around compatibility issues before you ever have them.  One of those ways is keeping older computers running with older software that is occasionally less strict.  Another way is recording your keypresses and timing, and matching your keyboard shortcuts in all your apps.  This saves a simple datafile you can use to reproduce your work.  WHere timing isn't an issue, you can have your timer ignored, and process the work even faster. ... 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Glad you got yours working. Others have had different issues. You had CFR but you were setting a 30p frame rate in NTSC standard film count. When you say 30fps or 30p, you're really saying 29.97fps average. This is a simple explanation, going back to film history in the US. In most places where PAL standards are the normal color and count list, you'll have exact values, for much the same reason, just a few adjustments to the mechanism. In film, or in celluloid days, you recorded audio and video separately and synchronized as best you could. In the metal oxide days, or VHS\8mm\hi8, you could record both in the same manner. However, there was a noticeable delay in sampling and imaging already in many television cameras (using electrical signal to broadcast), and bandwidth was precious, so rather than up the transfer bandwidth and frequency (more power needed), many cameras employed a slight delay in audio. When VHS came around it went the opposite way. Of course the main speeds were 24i, or 23.93fields (halved picture sets of tv lines) per second. This delay came as a result of separate heads recording, and sending at different rates or frequencies. The heads were independent, you could mute sound input and the head wouldn't touch the tape. All this mechanism introduced slight sync problems. By dropping the rate down to an average, the problem became unnoticeable. That way the US way. When other nations wanted to be more exact, the set the rates, and every so often time stretched and compressed to fit. 25p, 30pPAL, 50p, 60pPAL; these were used mainly in Europe and eastern nations. The difference isn't much, some bandwidth, and data, but their mechanism corrects itself. Just a note, 60p in the us is 59.94fps. It's a hold over from sync problems in early film. Even today, the exact frame rate isn't perfect in digital, and sync problems do occur at higher rates. Drop frame is a great way to adjust, stretching the problem over so long a string, it's infinitesimally small, and completely unnoticeable. But the NTSC color schemas... ...yuck! "Never the same color" = NTSC Glad you realized it was your set rate. For those with a similar issue, most HD is drop frame, but a lot of telecoms speed hike or create the frame. If you've set your sequence to a rate of exactly 30fps, and are out of sync, try droping the sequence rate to 29.97. Some game capture devices don't properly encode this value, hoping you'll use imovie or some other home software that will re-time your audio for you, sensing they are out of sync by checking track data and timecode. If you want to stay closer in sync, try a 24p rate. This is standard video rate, but it matches up with 48k audio sampling in 24 exact, and 44.1k in drop frame. For future reference, a pro workflow (for pro software like this) looks similar to: 1. receive video 2. ingest video 3. process video 4. output video 2 and 3 have a lot more to them. Ingesting is supposed to get rid of any sync problems, and you can set your video rate as you please. Going from 29.97 to 30 isn't that big of a leap, and AME can handle it if you turn on Frame blending in the options. I usually output 2 files at once from ingestion (2 for each video clip), one for editing and one for final pass through. There is a difference in the time it takes to render. More data going in always takes longer, but the output usually looks a little better. Ingesting video takes the guesswork out of the whole thing. Ripping to a single codec saves render time, as it doesn't need to open a new decoder for each clip. Color grading is better when you use a heavily compressed and a lightly compressed version of the same codec, and your end quality will be more accurate. Processing your video requires precision in some instances. Knowing the timecode's exact place, and that each and every video complies with that speed means that your machine isn't constantly re-timing playback during your edits. It would be great if you could get perfect output from every video codec at whatever speed and audio sample rate was preset all the time, just on a whim. Professionals rely on fact, figure, number, and naming. They rely on themselves, not everybody else, to do their work right. That's what this software targets. It's important to remember that. I remember watching an engineer with a PHD in Physics trying to get data from his TI calculator to his pc to run in a program he built. He kept crashing out of the program, and having to re-enter all the data. He insisted there was something wrong with the calculator or the pc. There was... ...He was using data measured in radians in a 4 dimensional plot in his math, but his calculator was set to degrees, and kept sending error data back to the pc, and his program would end up with bad output or a crash. He had no Idea how to set the thing to radians. He read that it computed radians, assumed it was the norm and went on pulling his hair out. It never dawned on him to learn the tool before using it. After I showed him how to set it, he sprang for a campus cup of coffee. Colleges do worse than starbucks... For those using Elements software, remember that it is derived from the same core as the professional set. It is called PROSUMER grade, not perfectly pro, but far above consumer grade. Saying "but isn't that different?" means you should've bought the $20 POS your kid said was trash, and been happy with it. It assumes there are problems, runs slower, and you have to pay out to add functions to it, but it assumes no knowledge on your part. It will baby you through it. Personally, I like learning pro software and using it at the same time. It's a walk back through history, and older workflows that were spot on for "Perfectionism" and professional quality. It takes more prep, but once through, the prep-stage makes you ready for everything else to happen at tachyon speeds (particles traveling faster than light are called tachyons, by the way). Processing requires tagging and knowlege of each clip or subclip, so you can quickly move them around. Nobody wants to worry about whether or not the framerates line up, and you don't want your system to give you the spinner (a computer's middle finger) that says it has to reprocess something over and over again. It's harder on hardware, and on your time, and your budget... ...not to mention your reputation. I urge you all to adopt an ingest step, where you tag or name your clips as you export them to an edit format. It will eliminate this problem completely. No more guessing. Pop goes the weasel... ...that ran away with my sanity. ... View more

MOV files are Quicktime file formats.  THey usually come from high end cameras with fast imaging sensors, and OIS that actually recalculates drop-frames or marks them for preprocessing.  In other words, it is handled BEFORE playback, not DURING.  When using MOV on windows, there really isn't a lot of support for playback, but adobe does have its own codec for them, and will process them with all adjustments on import. There are two ways reframes are handled in the short workflow: 1. Drop audio samples 2. Retime Audio These can result in acceptable results, but with worse cases, they don't help much. The middleware method, which is my common operation (I do it every time, no matter what), is to run the video through Adobe Media Encoder with frame blending.  It helps with some short run drops, and allows me to see them, then I just do a freezeFrame if necessary. Old Hat PRO's who are perfectionist or set in their ways will probably use the playback window to identify the dropped frames, then look before and after for areas of audio they might cut out.  This allows you to compensate for some problems and be exact if you do it by frame count. I've also found that the Frame Blending works on the timeline, but you have to highlight the area + up to 3s on either side, and render previews.  This will create the blends, and play them back, putting the audio in sync.  I havent done an output but I'd wager turning on frame blending in the export will carry out any timeline based blending. ... View more

that indicates that your cam uses strict eis not eois. causes sync problems adjusting both audio and video tracks when frames are dropped. i turn off all stabilization on those models and tripod or steadycam them. if an iphone or similar, then make sure no other apps running while you shoot, audio has gone out of sync for me even when video has 0 reframes and phone was on a tripod, all because several other apps were open. some claim ois but simply retiming audio fixes it, meaning its audio drift from electronic stabilizer trying to speed or slow to meet frame rate adjustments, but the values are not accurate enough in the sampler chip. dropping more than one or two frames in this instance causes a longer drift out of sync for each frame dropped as each drop increases the inaccuracy of audio sampling. ie you expect 48k, but you get a mix of anywhere from 22k to 52k audio samples depending on drop and catch-up. since sample count varies when sampled but is played back standard, small drop can be handled by clipping or retiming audio (old fashioned way is retime it with speed, but less accurate and not considered professional, which is why premiere allows you to see where frames are dropped in program/source monitor so you can localize the adjustment which is accurate and pro). you dropped audio samples as well, otherwise youd speed up audio locally like so: place cut marks around offending area if 1 frame dropped, select video area of frames 1 before drop, 1after since this area should have 3 frames, there's enough audio samples to fill 150% of number of actual frames (3/2=1.5 or 3 is 150% of 2 mathematically). speed up sound to 150% and it will only fill 100% of the same amount of time. there will appear a small gap, drag end side (right side of gap) audio clip over to fill. audio should be in perfect sync. adjust this as necessary. with dropped audio samples you drop speed or extend time of audio slightly. not exactly the same problem. but if you see audio behind video (happening after), dropped video frames; if happening before, dropped audio frames. always check media metadata for reframes count first, then be prepared for work in any pro app. live adds send through frameservers that play audio and video separately out multiplexed broadcast signal. they dont care what data is in each stream chunk; video frames are rebuilt with single pass blends and audio is played by samples in each second; when played back audio sampling varies as necessary as each sample chunk is timecoded to milliseconds of time rather than frames. the two are not sent in the same signal, though similar band. they are separated signals that are captured separately but played back simultaneously. even some pros dont realize that they are separated in multiplexing streams (chunks of separated data, 1set audio, 1set video) that are replaced by the next chunk as they are played back. thats new broadcast. i used to be able to get audio from tv shows on radios because of the separate signal in analogue air broadcasts. now the data is digital so its in chunks on the same band, and is processed separately, but played simultaneously. a 120hz tv has 88k or 96k audio (hidef 60p is =120i data rate, but only plays up to 60p due to nyquist theory of accurate representation-->usually tries to overframe and enhance motion with smoothing to keep it from looking sqaureish or aliased). i hope this all makes sense. you dropped audio samples if you slowed it to match video; or your mic was set for 44.1k when you recorded while you were using 48k in your sequence. you dropped audio somewhere. Sent from my iPhone ... View more

the file is now compressed folder with xml file in it. if you can figure out how to get the numbers to match up in the tags, youd be able to edit it and open the file. adobe will only go back 3 years for total compatability(or close to it). otherwise... export in a friendly format of xml file, then reimport into older apps. Cc lets you download cs6 for backward compatible ops Sent from my iPhone ... View more

30p, or 30i have a frame rate of 29.97fps (frames) and 29.983fps(fields per second). It is called drop frame or NTSC Drop Frame shooting. It's a carry over from old film days when you had to drop some frames and sampled audio to sync. Try this: look in VLC for reframes--same place you found frame rate, but REFRAMES are actual MISSING frames that are handled by playback engines. If it doesn't show any, maybe your sync in camera is off. Now import video to prem. project panel, right click it there, and select to turn on Frame Blending. This will guess missing frames for you and should fix sync problems. Also make sure your comp is fast enough to playback in realtime. When you output, check the box at the bottom with Frame Blending option to keep it on, your output will be fixed too. Old days you had to drop audio portions to fix butn now we just guess new frames. On Tue, May 3, 2016 at 5:22 PM, konsolekingz <forums_noreply@adobe.com> ... View more

did you try turning on frame blending on the clip before going from project panel to timeline? ... View more

audio overdriven is an easy one.   0db VU is actually +12 above optimal sound level. (ie more than 1000 times greater signal strength than 0db output level).  -12db is considered optimal level in digital.  up to -6 for headroom.  also, turn on frame blending in your premiere to compensate for dropped frames. ... View more

And for those using the labs we were born with? IE CS6?  There's a way to create the markers and capture them in a picture file.  If you can leave the interface a light grey, and screenshot your markers, you might, and I emphasize might, be able to OCR them.  Create a photoshop action to turn the image BW, burning the blues and blacks into a dark black, and raising the contrast.  Then OCR your image.  It may work, it may not.  I don't know why, but sometimes I get it to work, others it just wont.  Whatever the case, there's no XML access to the audio files' markers in audition.  You'll have to find it by dumping a file's metadata, and there are few programs with the capability to read each other's tags from audio files.  Something to do with offsets. ... View more

What version of Premiere do you have?  IF CS6, you need to add your card to the short list in the system files, but AFTER you make sure it is CUDA capable, and can run higher than version 3 of CUDA ( Version 3.4 or above).  Best to have version 4 or higher, since more ops in third party plugs are compatible, and will save processing time.  Make sure your card meets those requirements and then read the posts above.  They show you what file you have to change in order to get your card to work.  If you are on OSX (a mac) don't bother with CUDA.  It doesn't work.  Apple have removed the capability by overriding the library of functions with their own, lesser design. ... View more

Would you like to know why it plays fine in players and not an editor like Premiere? VFR= Variable Frame Rate. Premiere expects you to deal with frames that were dropped or weren't recorded by either turning ON the auto function, or retiming the audio yourself to compensate for the frames that aren't there. "How can they not be there!?" It's really very simple.  Most cameras do image stabilization at least one of two ways--the first is inoccuous, using floating optics to maintain stability--and the second method, which has become extremely popular, is to detect motion of the camera, or the outer edge of the frame of video and then drop some of those frames to avoid excessive frame changes that will degrade the fidelity of the following frames when uncompressed.  This dropping of frames PLAYS BACK fine, as most apps support "REFRAMING".  Look in vlc at the full info on the stream of video, and find the section with the name "REFRAMES".  If this shows up, you're missing frames, and the app replaces them by blending those around them intelligently into a new "Intermediate" frame that keeps visual continuity.  Premiere is more of an ALL AROUND kind of app.  It offers the amenities of automation, but you have to turn them on.  In PREMIERE, find your source clip in your project panel, right click, find FRAME BLENDING and turn it on.  When you output your video, check the box at the bottom that says USE FRAME BLENDING, to make sure it continues the action in the output (HINT: This option is also in Adobe Media Encoder, which ships with Premiere Pro, so you can queue your export).  By turning on frame blending, you are telling Premiere to blend any missing frames the same way VLC would.  The difference?  It won't do it automatically, which some people prefer.  In some cases, with high-definition video, you get horrible pixellation or there are so many missing frames that the blend horribly blurs some motion areas.  In these cases, It's usually best to either blend it yourself, and make slight adjustments in After Effects or Photoshop, or to retime your audio so it is back in sync.  See below for most common methods in film and professional video. RECAP: to turn on auto fix, right click the file in your PROJECT PANEL, Select Frame BLENDING to turn it on, then drop in any sequence.  It will try to play it back, but it may need to render a preview first.  Good luck. For the Curious and Intrepid... Pro Tricks: The most common method in the pros is to retime audio using a few key settings. If you can record at 24 Frames per second (a full 24 frames, not 23.93, you can use an old audio trick to retime what you need to with excessive camera shake or frame drop.  Set your audio sampling to 48k or 96k, nothing below 48k.  Bit depth 16-24bit is great fidelity, and 16bit is cd bit depth.  48k is FILM audio standard, just as 24P (24 progressive frames per second) is the video standard.  With 24 frames in a second, you have 48000 audio samples in that same second, and for each frame, you should have 2000 audio samples.  In audition, or premiere, you can clip audio by Frames.  This means you can find an area where things drift, find the spot where they drift (use Program monitor with FRAME MONITOR on--it lets you know where there are dropped frames from gfx low power or from missing frames in video, though both look the same).  You'll have to use the mouse to move across the section a bit slowly.  Then, where you see the missing frame, cut the audio in the frame after it, leaving the video alone.  With several missing frames in a row, this takes a while, and your audio might get a little hinky.  So you can fade it out quickly on the frame before, and back in on the frame after the drop.  You'll also have to clip the video into subclips and replace it in your timeline.  When you export the video, turn off Frame Blending, and you should have sync'd output the professional way. If you use 23.93FPS, this will not work.  Use a NON-DROP-FRAME setting on your camera, or record your screen with full 24 frames Progressive video.  It will remove exactly 2000 samples of audio every frame you clip.  Otherwise, it will vary by which frame you're in, and won't line up the same.  You might clip too many or not enough samples by only a small ammount and have to either undo the clip, or clip even more.  If this does happen, you may have to go down to the SAMPLE level and view the entirety in audio samples, using very short, very slow selections.  This isn't just tedious, it's like using a jackhammer to drive Rail Nails through your head.  But it is the PRO way.  This is the methodology that Premiere was built upon. ... View more

The reason for dropping backward compatibility is NOT nefarious.  It seems like it, and even smells like it at times, but it really isn't.  If you wish to make adjustments ON YOUR OWN that make it work, That's FINE with adobe, just don't expect them to support it. The reason is simple: Many of the plugs and added effects are not readable by cs6 and lower, and they are built differently in some cases even if they are included in cs6.  They also compressed the data in the file (not really heavy compression) and applied a light encryption.  WHy?  It secures the data while working and saves a bit of space on drives, but also keeps some viruses from actually working on the file and destroying the data.  IF a file is compressed in this way, there is a hash to hint about contents, making it easier to recover some of the file, and even import that into a new project for use. To get around these features, copy paste the file in place with "copy" appended to the name, then rename it and give it a GZ extension.  Now unzip that to it's own folder, and open the folder.  There's a file in there that is the same XML data you would use in CS6, but you'll need to change the XML version to 25 at the highest.  Now save that file, copy\move it out of the folder, then give it the prproj extension.  You should be able to open it.  You may get warnings, and if it doesn't work at all, you'll have to change one more value.  My file has an Info version 3 at the top, try 1 or 2.  I'm not going to guarantee this method, but open it and try it.  If it fails, try importing the project into a clean one.  That might do the trick.  Good luck. Personally, I do a lot of homework and some nonprofit in CS6, then touch up in CC for better color and a few other refinements.  I'm hoping to have the funding stream to upgrade to CC in the near future.  Not a sure thing, but if you never risk, you never win it all. ... View more