Mystery audio performance changes and distortion
I listen to a lot of rock, hard rock, and heavy metal on my 80GB Video iPod. In the past two months, I’ve been having problems with distortion and equalization. I do have the latest iTunes and iPod software updates installed and I have re-instaled iTunes and re-initialized the iPod.
Originally, I was using the iPod earbuds and upgraded to vModa Vibes. Most of my music sounded great with an EQ setting of “Dance” with the vModas.
After a while, I started to experience distortion and loss of detail, especially on the left side (crackling). I thought I had damaged the vModa Vibes, so I purchased a pair of Ultimate Ears Super.fi 5 Pro Earphones.
The Ultimate Ears Super.fi 5 Pro earphones sounded terrible with no highs or midrange detail and way too much low range. They seemed to distort easily and overload in the-mid range if there was a lot guitars playing (drowning the details out). I tried all sorts of EQ settings to no avail. The problems remained, even on treble boost, but I didn't have a problem specific to the left side.
I returned the Ultimate Ears Super.fi 5 Pro earphones and switched back to the vModa Vibes for a while and finally decided, on the recommendation of a friend, to try the Bose TriPort IE earphones.
The Bose TriPort IE earphones were much better than the Ultimate Ears Super.fi 5 Pro earphones, but still lacked high frequency and mid-range detail as compared to the vModa Vibes.
While switching back and forth between the vModas and the Bose earphones, the sound performance (or equalization) seemed to change at random, even though I kept the EQ set to off.
At one point, the vModas suddenly had incredible bass, while maintaining the other performance characteristics, with the EQ still set to off. Later, I played the same song again, and the bass was normal (all I did was charge the battery with the Apple USB AC adapter).
I’ve been using the vModas again for about a week now, and every time I listen to the same song, it plays differently, sometimes great, sometimes with extra bass, sometimes muddled and distorted. The same thing happens with Bose TriPort IEs, except they aren’t as clear as the vModas.
The performance issues seem to somewhat track battery charging (i.e. – after I charge the iPod up, audio performance changes and it sometimes improves as the battery drains).
I figure that either there is a hardware problem with the iPod or there is some type of EQ virus going around.
Anyone having similar problems or have a solution??
I went out and bought a 2 GB iPod Nano and loaded some songs onto it.
Everything sounds great with the vModa Vibe earphones so I guess my 80 GB video iPod might have a few twisted transistors.
I'll have to try to clear out the 80 GB video iPod and reinitialize/restore it again to see if it corrects the problem.
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This information has come from Durin, Adobe staffer:
Hi everyone,
A common question that comes up in these forums over and over has to do with recording latency, audio drivers, and device formats. I'm going to provide a brief overview of the different types of devices, how they interface with the computer and Audition, and steps to maximize performance and minimize the latency inherent in computer audio.
First, a few definitions:
Monitoring: listening to existing audio while simultaneously recording new audio.
Sample: The value of each individual bit of audio digitized by the audio device. Typically, the audio device measures the incoming signal 44,100 or 48,000 times every second.
Buffer Size: The "bucket" where samples are placed before being passed to the destination. An audio application will collect a buffers-worth of samples before feeding it to the audio device for playback. An audio device will collect a buffers-worth of samples before feeding it to the audio device when recording. Buffers are typically measured in Samples (command values being 64, 128, 512, 1024, 2048...) or milliseconds which is simply a calculation based on the device sample rate and buffer size.
Latency: The time span that occurs between providing an input signal into an audio device (through a microphone, keyboard, guitar input, etc) and when each buffers-worth of that signal is provided to the audio application. It also refers to the other direction, where the output audio signal is sent from the audio application to the audio device for playback. When recording while monitoring, the overall perceived latency can often be double the device buffer size.
ASIO, MME, CoreAudio: These are audio driver models, which simply specify the manner in which an audio application and audio device communicate. Apple Mac systems use CoreAudio almost exclusively which provides for low buffer sizes and the ability to mix and match different devices (called an Aggregate Device.) MME and ASIO are mostly Windows-exclusive driver models, and provide different methods of communicating between application and device. MME drivers allow the operating system itself to act as a go-between and are generally slower as they rely upon higher buffer sizes and have to pass through multiple processes on the computer before being sent to the audio device. ASIO drivers provide an audio application direct communication with the hardware, bypassing the operating system. This allows for much lower latency while being limited in an applications ability to access multiple devices simultaneously, or share a device channel with another application.
Dropouts: Missing audio data as a result of being unable to process an audio stream fast enough to keep up with the buffer size. Generally, dropouts occur when an audio application cannot process effects and mix tracks together quickly enough to fill the device buffer, or when the audio device is trying to send audio data to the application more quickly than it can handle it. (Remember when Lucy and Ethel were working at the chocolate factory and the machine sped up to the point where they were dropping chocolates all over the place? Pretend the chocolates were samples, Lucy and Ethel were the audio application, and the chocolate machine is the audio device/driver, and you'll have a pretty good visualization of how this works.)
Typically, latency is not a problem if you're simply playing back existing audio (you might experience a very slight delay between pressing PLAY and when audio is heard through your speakers) or recording to disk without monitoring existing audio tracks since precise timing is not crucial in these conditions. However, when trying to play along with a drum track, or sing a harmony to an existing track, or overdub narration to a video, latency becomes a factor since our ears are far more sensitive to timing issues than our other senses. If a bass guitar track is not precisely aligned with the drums, it quickly sounds sloppy. Therefore, we need to attempt to reduce latency as much as possible for these situations. If we simply set our Buffer Size parameter as low as it will go, we're likely to experience dropouts - especially if we have some tracks configured with audio effects which require additional processing and contribute their own latency to the chain. Dropouts are annoying but not destructive during playback, but if dropouts occur on the recording stream, it means you're losing data and your recording will never sound right - the data is simply lost. Obviously, this is not good.
Latency under 40ms is generally considered within the range of reasonable for recording. Some folks can hear even this and it affects their ability to play, but most people find this unnoticeable or tolerable. We can calculate our approximate desired buffer size with this formula:
(Sample per second / 1000) * Desired Latency
So, if we are recording at 44,100 Hz and we are aiming for 20ms latency: 44100 / 1000 * 20 = 882 samples. Most audio devices do not allow arbitrary buffer sizes but offer an array of choices, so we would select the closest option. The device I'm using right now offers 512 and 1024 samples as the closest available buffer sizes, so I would select 512 first and see how this performs. If my session has a lot of tracks and/or several effects, I might need to bump this up to 1024 if I experience dropouts.
Now that we hopefully have a pretty firm understanding of what constitutes latency and under what circumstances it is undesirable, let's take a look at how we can reduce it for our needs. You may find that you continue to experience dropouts at a buffer size of 1024 but that raising it to larger options introduces too much latency for your needs. So we need to determine what we can do to reduce our overhead in order to have quality playback and recording at this buffer size.
Effects: A common cause of playback latency is the use of effects. As your audio stream passes through an effect, it takes time for the computer to perform the calculations to modify that signal. Each effect in a chain introduces its own amount of latency before the chunk of audio even reaches the point where the audio application passes it to the audio device and starts to fill up the buffer. Audition and other DAWs attempt to address this through "latency compensation" routines which introduce a bit more latency when you first press play as they process several seconds of audio ahead of time before beginning to stream those chunks to the audio driver. In some cases, however, the effects may be so intensive that the CPU simply isn't processing the math fast enough. With Audition, you can "freeze" or pre-render these tracks by clicking the small lightning bolt button visible in the Effects Rack with that track selected. This performs a background render of that track, which automatically updates if you make any changes to the track or effect parameters, so that instead of calculating all those changes on-the-fly, it simply needs to stream back a plain old audio file which requires much fewer system resources. You may also choose to disable certain effects, or temporarily replace them with alternatives which may not sound exactly like what you want for your final mix, but which adequately simulate the desired effect for the purpose of recording. (You might replace the CPU-intensive Full Reverb effect with the lightweight Studio Reverb effect, for example. Full Reverb effect is mathematically far more accurate and realistic, but Studio Reverb can provide that quick "body" you might want when monitoring vocals, for example.) You can also just disable the effects for a track or clip while recording, and turn them on later.
Device and Driver Options: Different devices may have wildly different performance at the same buffer size and with the same session. Audio devices designed primarily for gaming are less likely to perform well at low buffer sizes as those designed for music production, for example. Even if the hardware performs the same, the driver mode may be a source of latency. ASIO is almost always faster than MME, though many device manufacturers do not supply an ASIO driver. The use of third-party, device-agnostic drivers, such as ASIO4ALL (www.asio4all.com) allow you to wrap an MME-only device inside a faux-ASIO shell. The audio application believes it's speaking to an ASIO driver, and ASIO4ALL has been streamlined to work more quickly with the MME device, or even to allow you to use different inputs and outputs on separate devices which ASIO would otherwise prevent.
We also now see more USB microphone devices which are input-only audio devices that generally use a generic Windows driver and, with a few exceptions, rarely offer native ASIO support. USB microphones generally require a higher buffer size as they are primarily designed for recording in cases where monitoring is unimportant. When attempting to record via a USB microphone and monitor via a separate audio device, you're more likely to run into issues where the two devices are not synchronized or drift apart after some time. (The ugly secret of many device manufacturers is that they rarely operate at EXACTLY the sample rate specified. The difference between 44,100 and 44,118 Hz is negligible when listening to audio, but when trying to precisely synchronize to a track recorded AT 44,100, the difference adds up over time and what sounded in sync for the first minute will be wildly off-beat several minutes later.) You are almost always going to have better sync and performance with a standard microphone connected to the same device you're using for playback, and for serious recording, this is the best practice. If USB microphones are your only option, then I would recommend making certain you purchase a high-quality one and have an equally high-quality playback device. Attempt to match the buffer sizes and sample rates as closely as possible, and consider using a higher buffer size and correcting the latency post-recording. (One method of doing this is to have a click or clap at the beginning of your session and make sure this is recorded by your USB microphone. After you finish your recording, you can visually line up the click in the recorded track with the click in the original track by moving your clip backwards in the timeline. This is not the most efficient method, but this alignment is the reason you see the clapboards in behind-the-scenes filmmaking footage.)
Other Hardware: Other hardware in your computer plays a role in the ability to feed or store audio data quickly. CPUs are so fast, and with multiple cores, capable of spreading the load so often the bottleneck for good performance - especially at high sample rates - tends to be your hard drive or storage media. It is highly recommended that you configure your temporary files location, and session/recording location, to a physical drive that is NOT the same as you have your operating system installed. Audition and other DAWs have absolutely no control over what Windows or OS X may decide to do at any given time and if your antivirus software or system file indexer decides it's time to start churning away at your hard drive at the same time that you're recording your magnum opus, you raise the likelihood of losing some of that performance. (In fact, it's a good idea to disable all non-essential applications and internet connections while recording to reduce the likelihood of external interference.) If you're going to be recording multiple tracks at once, it's a good idea to purchase the fastest hard drive your budget allows. Most cheap drives spin around 5400 rpm, which is fine for general use cases but does not allow for the fast read, write, and seek operations the drive needs to do when recording and playing back from multiple files simultaneously. 7200 RPM drives perform much better, and even faster options are available. While fragmentation is less of a problem on OS X systems, you'll want to frequently defragment your drive on Windows frequently - this process realigns all the blocks of your files so they're grouped together. As you write and delete files, pieces of each tend to get placed in the first location that has room. This ends up creating lots of gaps or splitting files up all over the disk. The act of reading or writing to these spread out areas cause the operation to take significantly longer than it needs to and can contribute to glitches in playback or loss of data when recording.There is one point in the above that needed a little clarification, relating to USB mics:
_durin_ wrote:
If USB microphones are your only option, then I would recommend making certain you purchase a high-quality one and have an equally high-quality playback device.
If you are going to spend that much, then you'd be better off putting a little more money into an external device with a proper mic pre, and a little less money by not bothering with a USB mic at all, and just getting a 'normal' condensor mic. It's true to say that over the years, the USB mic class of recording device has caused more trouble than any other, regardless.
You should also be aware that if you find a USB mic offering ASIO support, then unless it's got a headphone socket on it as well then you aren't going to be able to monitor what you record if you use it in its native ASIO mode. This is because your computer can only cope with one ASIO device in the system - that's all the spec allows. What you can do with most ASIO hardware though is share multiple streams (if the device has multiple inputs and outputs) between different software.
Seriously, USB mics are more trouble than they're worth. -
Replacing audio track with and synching with separate recording
Hello,
I've read everything I can find, including numerous websites, the help file, and a book, but can't find this answer with step by step instructions using Premiere Elements 9.
I have the trial version, and will purchase once I figure this out. I only need to make Youtube videos with a separately recorded audio file replacing the audio that is already on the video. in other words, I have a separate Mp3 audio file of the same audio performance using an audio recorder and I want to replace the audio on the original video with the other Mp3.
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I did not take your comment as dismissive in any way, and highly doubt that Steve did either. I do not have his newer version book, but only some older ones, or I would have grabbed it and checked to see if he DID cover this material in it, just maybe buried in another topic. My comments were pure speculation, as what you are doing is not "mainstream," though is becoming more so monthly.
For my PrPro (slightly older version), I have the printed manual (yes, they used to do those), the Adobe Classroom in a Book, plus several other volumes dedicated to it, and even the print-out of the Help File PDF. Still, I, and other users are always coming up with some use, that is not covered anywhere!
One of the recent changes to the Help File in many Adobe programs, is that it is now a live On-line document, and there are engineers, who do little but update and add to that. This was almost impossible with printed material (though I still have my old ring-binder for the original WordPerfect, and monthly, they used to send me new pages to either add, or replace - long time ago).
When I am working in the forums, I usually have the particular program open and in front of me, as I am always coming up against questions, thinking, "wow, I would never have thought of doing that," and having to duplicate the actions that a poster outlines, or experimenting to find if what they want is possible. Often, when I do one of the experiments successfully, I'll add an article on doing it, to the Tips & Tricks sub-forum here. One example of that was "Fading a Title Crawl," which turned out to be both fun, and relatively easy, when I got into the experiments.
I have a stong feeling that Steve will have a bit more "music video" instruction in his next PrE book, as it is becoming more popular, as evidenced by the number of posts that we are seeing lately. More folk are doing that. Same thing for shooting with what were considered non-standard cameras, just a few years ago, like various smartphones.
Good luck, and happy editing,
Hunt -
Sound clicking and distortion with X
Anyone else having sound clicking and distortion with their X-Fi Card? Its rare in most games, but I have a couple games where its very common Eve-Online for example. I have a X-Fi Fatalty FPS.
Using Asio4All will introduce a higher latency than using Creative X-Fi ASIO / Bit-Matched -mode, so the later is recommended. If you get distortion make sure that your volume is not turned up to max, it should be around 0db in the mixer, which corresponds to about 80%. If you get cracks/pops then it's due to the bad performance of the ASIO drivers of the X-Fi. The latest beta also says to remove cracks/pops and some distortion problems in audio creation mode, maybe try them.
Again, here is a list of real latencies as measured on my own system, using the latest stable driver (not the beta one):
Rate = sample-rate
Comp = latency compensation in ms (by how much the real latency differs from what the application/driver reports
Low = lowest no of buffers (in samples) that I was able to use
Real = real measured latency (in contrast to what the application/drivers reports)
ASIO4ALL
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48k hb: +6.18 80 12.2
44k kb: +15.5 80 19.1
48k kb: +6.20 80 9.53
48k kb: +5.66 80 8.99
96k: +6.71 80 8.38
192k +5.78 144 8.03
Creative X-Fi not Bit-Matched
Rate Comp. Low Real
44k: +9.05 44 13.3 !most times glitches!
48k: +8.29 48 12.4
88k: +9.73 88 12.9
96k: +8.94 96 12.0
Creative X-Fi Bit-Matched
Rate Comp. Low Real
44k: -0.65 44 3.61 !most times glitches!
48k: -0.61 48 3.47
88K: -0.27 88 2.86
96k: -0.25 96 2.79
Delta Audiophile 2496
Rate Comp. Low Real
32k: +0.15 64 6.09
44k: +0.11 64 4.42
48k: +0.08 64 4.04
88k: +0.05 64 2.20
96k: +0.10 64 2.08
Message Edited by Timur on 06-03-2006 09:18 AM
Message Edited by Timur on 06-03-2006 09:22 AM -
Audio playback frequencies are distorted...
All of a sudden, audio playback through my speakers/headphone jack is muddy and distorted. Sometimes, the vocal track disappears completely. This is not just in iTunes (where I've unsuccessfully tried to adjust the equalizer), it's also on Youtube videos. Sonos, which does NOT go thru system audio, sounds fine. I'm including etrecheck diagnostics. Thanks for any ideas. Oh, and the only thing that's changed, systemwise, is the latest Yosemite upgrade. The problem started after that, but not immediately after.
Problem description:
Audio playback is distorted
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Report generated April 16, 2015 at 11:20:10 PM CDT
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User Launch Agents: ℹ️
[loaded] com.adobe.ARM.[...].plist [Click for support]
[running] com.hp.printerAgent.plist [Click for support]
User Login Items: ℹ️
iTunesHelper Application (/Applications/iTunes.app/Contents/MacOS/iTunesHelper.app)
Dropbox Application (/Applications/Dropbox.app)
Safari Application (/Applications/Safari.app)
Mail Application (/Applications/Mail.app)
AdobeResourceSynchronizer Application Hidden (/Applications/Adobe Reader.app/Contents/Support/AdobeResourceSynchronizer.app)
Fitbit Connect Menubar Helper Application (/Applications/Fitbit Connect.app/Contents/MacOS/Fitbit Connect Menubar Helper.app)
Airfoil Speakers Application Hidden (/Applications/Airfoil Speakers.app)
Printer Pro Desktop Application (/Applications/Printer Pro Desktop.app)
Internet Plug-ins: ℹ️
FlashPlayer-10.6: Version: 17.0.0.169 - SDK 10.6 [Click for support]
QuickTime Plugin: Version: 7.7.3
AdobePDFViewerNPAPI: Version: 11.0.10 - SDK 10.6 [Click for support]
AdobePDFViewer: Version: 11.0.10 - SDK 10.6 [Click for support]
Flash Player: Version: 17.0.0.169 - SDK 10.6 [Click for support]
Default Browser: Version: 600 - SDK 10.10
Silverlight: Version: 5.1.30514.0 - SDK 10.6 [Click for support]
JavaAppletPlugin: Version: Java 8 Update 31 Check version
User internet Plug-ins: ℹ️
Stamps.com: Version: 1.1.13 [Click for support]
Safari Extensions: ℹ️
1-ClickWeather
Adblock Plus
SafariRestore
3rd Party Preference Panes: ℹ️
Flash Player [Click for support]
GPGPreferences [Click for support]
Java [Click for support]
Time Machine: ℹ️
Skip System Files: NO
Mobile backups: ON
Auto backup: YES
Volumes being backed up:
Macintosh HD: Disk size: 150.14 GB Disk used: 46.65 GB
Destinations:
Data [Network]
Total size: 497.62 GB
Total number of backups: 55
Oldest backup: 2014-03-29 22:27:52 +0000
Last backup: 2015-04-17 03:29:42 +0000
Size of backup disk: Excellent
Backup size 497.62 GB > (Disk size 150.14 GB X 3)
Top Processes by CPU: ℹ️
4% WindowServer
1% fontd
0% Fitbit Connect Menubar Helper
0% Dropbox
0% AppleSpell
Top Processes by Memory: ℹ️
361 MB Safari
100 MB com.apple.WebKit.WebContent
86 MB mds_stores
86 MB Mail
86 MB WindowServer
Virtual Memory Information: ℹ️
10.03 GB Free RAM
5.01 GB Active RAM
571 MB Inactive RAM
1.57 GB Wired RAM
1.63 GB Page-ins
0 B Page-outs
Diagnostics Information: ℹ️
Apr 16, 2015, 09:53:12 PM Self test - passed
Apr 16, 2015, 11:11:30 AM /Library/Logs/DiagnosticReports/CalendarAgent_2015-04-16-111130_[redacted].cras h
Apr 15, 2015, 12:23:01 PM /Library/Logs/DiagnosticReports/com.apple.WebKit.WebContent_2015-04-15-122301_[ redacted].cpu_resource.diag [Click for details]
Apr 14, 2015, 06:55:26 PM /Users/[redacted]/Library/Logs/DiagnosticReports/Mail_2015-04-14-185526_[redact ed].crash
Apr 14, 2015, 02:47:54 AM /Library/Logs/DiagnosticReports/CalendarAgent_2015-04-14-024754_[redacted].cras hWill try. There are no cables/exterior connections involved here.
Oddly enough, this worked. I was unable to hear anything in Safe Mode, so just exited and rebooted (yes, I'd already tried rebooting, with no effect). Gave iTunes a shot, and am now back to normal.
Whatever works!
Thank you.
bkb
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