Supported bit depth and sampling frequency

Hi,
Is it possible to determine at runtime what sampling frequencies and bit depths are available for audio capture -- other than by trial and error? It does not seem that way from the API, but maybe I missed something, so I decided to post a question here. The only thing I was able to find so far is that 8-bit PCM WAV is required to be supported by all devices that support audio at all.
Thanks in advance for any info!
Sergei

Thanks for the info. I guess I was looking for something more specific - the exact bitrates and sample rates that Creative claims to support. Would you know where official and comprehensi've data can be had? There must be a tech spec somewhere.
It is common these days in business to see a recording of, say, a conference call or seminar presentation at 32k bitrate/025Hz, or even 24k bitrate/8000Hz, posted to a company's website for download by those who could not be there, and MP3 players are increasingly used for their replay. Companies use low digitization rates because there is no need for hifi and the files are much smaller: less storage, faster download.
I'd be surprized to think that Creative don't have compatibility with the standard range of rates offered by ubiquitous programs like Audacity and dBpower, the latter being one they themselves recommend!

Similar Messages

Internal handling of Bit Depth and Sample Rate

I am wondering if iTunes allows accurate import of 48KHz/24bit files or whether it will convert to 44.1KHz/16bit as default. There is contradictory information in the Internet.
Thanks.

I just experimented on my machine, which is running iTunes 8.1. 48/24 WAV files can be added to iTunes. When adding to the library, it does not change them.
However, if you use iTunes as a conversion tool, it cannot create 48/24 files. The highest it can go is 48/16. That may be the source of the seemingly contradictory information that you read.
I did not try syncing the iPod. However, based on past problems with 48/16 files, I would not be surprised if the iPod has a problem with 48/24 files.

Bit Depth and Render Quality

When you finally export media to some sort of media format via the encoder does the projects preview Bit Depth and Render Quality settings affect the output file?
I know there is "Use Preview files" setting in the media exporter dialogue but I just want to be sure of what I am doing.

Jeff's response is my perspective, as well, which is both backed up by my own tests and the official Adobe word.
Exhibit A: My Tests
That is DV footage with a title superimposed over it in a DV sequence, with a Gaussian blur effect (the Premiere accelerated one) applied to the title; all samples are from that sequence exported back to DV. This was to show the relative differences of processing between software and hardware MPE, Premiere export and AME queueing, and the effect of the Maximum Bit Depth and Maximum Render Quality options on export (not the sequence settings; those have no bearing on export).
The "blooming" evident in the GPU exports is due to hardware MPE's linear color processing. I think it's ugly, but that's not the point here. Further down the line, you can see the effect of Maximum Bit Depth (and MRQ) on both software MPE and hardware MPE. I assume you can see the difference between the Maximum Bit Depth-enabled export and the one without. Bear in mind that this is 8-bit DV footage composited and "effected" and exported back to 8-bit DV. I don't understand what your "padding with zeroes" and larger file size argument is motivated by--my source files and destination files are the same size due to the DV codec--but it's plainly clear that Maximum Bit Depth has a significant impact on output quality. Similar results would likely be evident if I used any of the other 32-bit enabled effects; many of the color correction filters are 32-bit, and should exhibit less banding, even on something 8-bit like DV.
Exhibit B: The Adobe Word
This is extracted from Karl Soule's blog post, Understanding Color Processing: 8-bit, 10-bit, 32-bit, and more. This section comes from Adobe engineer Steve Hoeg:
1. A DV file with a blur and a color corrector exported to DV without the max bit depth flag. We
will import the 8-bit DV file, apply the blur to get an 8-bit frame,
apply the color corrector to the 8-bit frame to get another 8-bit frame,
then write DV at 8-bit.
2. A DV file with a blur and a color corrector exported to DV with the max bit depth flag. We
will import the 8-bit DV file, apply the blur to get an 32-bit frame,
apply the color corrector to the 32-bit frame to get another 32-bit
frame, then write DV at 8-bit. The color corrector working on the 32-bit
blurred frame will be higher quality then the previous example.
3. A DV file with a blur and a color corrector exported to DPX with the max bit depth flag. We
will import the 8-bit DV file, apply the blur to get an 32-bit frame,
apply the color corrector to the 32-bit frame to get another 32-bit
frame, then write DPX at 10-bit. This will be still higher quality
because the final output format supports greater precision.
4. A DPX file with a blur and a color corrector exported to DPX without the max bit depth flag.
We will clamp 10-bit DPX file to 8-bits, apply the blur to get an 8-bit
frame, apply the color corrector to the 8-bit frame to get another
8-bit frame, then write 10-bit DPX from 8-bit data.
5. A DPX file with a blur and a color corrector exported to DPX with the max bit depth flag.
We will import the 10-bit DPX file, apply the blur to get an 32-bit
frame, apply the color corrector to the 32-bit frame to get another
32-bit frame, then write DPX at 10-bit. This will retain full precision through the whole pipeline.
6. A title with a gradient and a blur on a 8-bit monitor. This will display in 8-bit, may show banding.
7. A title with a gradient and a blur on a 10-bit monitor
(with hardware acceleration enabled.) This will render the blur in
32-bit, then display at 10-bit. The gradient should be smooth.
Bullet #2 is pretty much what my tests reveal.
I think the Premiere Pro Help Docs get this wrong, however:
High-bit-depth effects
Premiere Pro includes some video effects and transitions
that support high-bit-depth processing. When applied to high-bit-depth
assets, such as v210-format video and 16-bit-per-channel (bpc) Photoshop
files, these effects can be rendered with 32bpc pixels. The result
is better color resolution and smoother color gradients with these
assets than would be possible with the earlier standard 8 bit per
channel pixels. A 32-bpc badge appears
to the right of the effect name in the Effects panel for each high-bit-depth
effect.
I added the emphasis; it should be obvious after my tests and the quote from Steve Hoeg that this is clearly not the case. These 32-bit effects can be added to 8-bit assets, and if the Maximum Bit Depth flag is checked on export, those 32-bit effects are processed as 32-bit, regardless of the destination format of the export. Rendering and export/compression are two different processes altogether, and that's why using the Maximum Bit Depth option has far more impact than "padding with zeroes." You've made this claim repeatedly, and I believe it to be false.
Your witness...

Bit Depth and Bit Rate

I have a pre recorded mp3 VO. I placed it into a track bed in GB. Clients wants a compressed audio file with bit depth: 16 bit and bitrate: 128kps max, but recommends 96kbps. If I need to adjust the bit depth and bite rate, can I do it in GB? and if so, where? Thanks for any help.

Please be aware that Bit Depth and Bit Rate are two completely different things!
They belong to a group of buzz words that belong to Digital Audio and that is the field we are dealing with when using GarageBand or any other DAW. Some of those terms pop up even in iTunes.
Digital Audio
To better understand what they are and what they mean, here is a little background information.
Whenever dealing with Digital Audio, you have to be aware of two steps, that convert an analog audio signal into a digital audio signal. These magic black boxes are called ADC (Analog Digital Converter) and “on the way back”, DAC (Digital Analog Converter).
Step One: Sampling
The analog audio (in the form of an electric signal like from an electric guitar) is represented by a waveform. The electric signal (voltage) changes up and down in a specific form that represents the “sound” of the audio signal. While the audio signal is “playing”, the converter measure the voltage every now and then. These are like “snapshots” or samples, taken at a specific time. These specific time intervals are determined by a “Rate”, it tells you how often per seconds something happens. The unit is Hertz [Hz] defined as “how often per seconds” or “1/s”. A Sample Rate of 48kHz means that the converter takes 48,000 Samples per second.
Step Two: Quantize (or digitize)
All these Samples are still analog, for example, 1.6Volt, -0.3Volt, etc. But this analog value now has to be converted into a digital form of 1s and 0s.This is done similar to quantizing a note in GarageBand. The value (i.e. the note) cannot have any position, it has to be placed on a grid with specific values (i.e. 1/16 notes). The converter does a similar thing. It provides a grid of available numbers that the original measured Sample has to be rounded to (like when a note get shifted in GarageBand by the quantize command). This grid, the amount of available numbers, is called the Bit Depth. Other terms like Resolution or Sample Size are also used. A Bit Depth of 16bit allows for 65,535 possible values.
So the two parameters that describe the quality of an Digital Audio Signal are the Sample Rate (“how often”) and the Bit Depth (“how fine of a resolution”). The very simplified rule of thumb is, the higher the Sample Rate, the higher the possible frequency, and the higher the Bit Depth, the higher the possible dynamic.
Uncompressed Digital Audio vs. Compressed Digital Audio
So far I haven’t mentioned the “Bit Rate” yet. There is a simple formula that describes the Bit Rate as the product of Sampel Rate and Bit Depth: Sample Rate * Bit Depth = Bit Rate. However, Bit Depth and how it is used (and often misused and misunderstood) has to do with Compressed Digital Audio.
Compressed Digital Audio
First of all, this has nothing to do with a compressor plugin that you use in GarageBand. When talking about compressed digital audio, we talk about data compression. This is a special form how to encode data to make the size of the data set smaller. This is the fascinating field of “perceptual coding” that uses psychoacoustic models to achieve that data compression. Some smart scientists found out that you can throw away some data in a digital audio signal and you wouldn’t even notice it, the audio would still sound the same (or almost the same). This is similar to a movie set. If you shoot a scene on a street, then you only need the facade of the buildings and not necessary the whole building.
Although the Sample Rate is also a parameter of uncompressed digital audio, the Bit Depth is not. Instead, here is the Bit Rate used. The Bit Rate tells the encoder the maximum amount of bits it can produce per second. This determines how much data it has to throw away in order to stay inside that limit. An mp3 file (which is a compressed audio format) with a Bit Rate of 128kbit/s delivers a decent audio quality. Raising the Bit Rate to 256bit/s would increase the sound quality. AAC (which is technically an mp4 format) uses a better encoding algorithm. If this encoder is set to 128kbit/s, it produces a better audio quality because it is smarter to know which bits to throw away and which one to keep.
Conclusion
Whenever you are dealing with uncompressed audio (aiff, wav), the two quality parameters are Sample Rate [kHz] and Bit Depth [bit] (aka Resolution, aka Bit Size)
Whenever you are dealing with compressed audio (mp3, AAC), the two quality parameters are Sample Rate [kHz] and Bit Rate [kbit/s]
If you look at the Export Dialog Window in GarageBand, you can see that the Quality popup menu is different for mp3/AAC and AIFF. Hopefully you will now understand why.
Hope that helps
Edgar Rothermich
http://DingDingMusic.com/Manuals/
'I may receive some form of compensation, financial or otherwise, from my recommendation or link.'

"current encoder settings for bit rate and sample rate are invalid" message

I have some files I am working with. Details and what happened:
Had some music files where I was doing some trimming/splitting. Files worked fine in itunes.
Open the files in Quicktime Pro 7 to trim them. Exported to aif.
Opened the files with itunes. They open and play just fine.
In Itunes I click"create AAC version" to convert them.
I receive the message "An error occurred while trying to import the file. The current encoder settings for bit rate and sample rate are not valid for this file."
I don't know what to do from here. Suggestions?
Thanks.

Make sure that the AIFFs that you export are set for 16 bit PCM.
Also, assuming that you are using 44.1 kHZ sampling in your AIFFs, make sure that you are not trying to use 48 kHz in the AAC.

In Finder how do I get iTunes to show bit rate and sample rate of an entry?

In Finder how do I get iTunes to show bit rate and sample rate of an entry so I can set my DAC?

TonyEyes wrote:
In Finder how do I get iTunes to show bit rate and sample rate of an entry so I can set my DAC?
Tony, Make sure you are in Songs view. Then View > Show View Options, and check the boxes for "Bit Rate" and "Sample Rate." See picture:

Turning on Render at Maximum Bit Depth and Maximum Render Quality crashes render every time

I've tried a few times to render an H264 version of my Red media project with Maximum Bit Depth and Maximum Render Quality. Premiere crashes every time. I have GPUs enabled. Are people using these settings with Red media and successfully rendering?

To answer your specific question did you see the tooltip?
I beleive it allows for 32-bit processing (16-bit if unchecked). Per the project settings help file at http://helpx.adobe.com/premiere-elements/using/project-settings-presets.html
Maximum Bit Depth
Allows Premiere Elements to use up to 32‑bit processing, even if the project uses a lower bit depth. Selecting this option increases precision but decreases performance.
The help file for export is somewhat less informative about what it actually does but does point out that it is the color bit depth - http://helpx.adobe.com/media-encoder/using/encode-export-video-audio.html
(Optional) Select Use Maximum Render Quality or Render At Maximum Bit Depth. Note: Rendering at a higher color bit depth requires more RAM and slows rendering substantially.
In practice the simplest suggestion is to export twice - once with / once without the setting and compare the time taken and perceived quality.
Cheers,
Neale
Insanity is hereditary, you get it from your children
If this post or another user's post resolves the original issue, please mark the posts as correct and/or helpful accordingly. This helps other users with similar trouble get answers to their questions quicker. Thanks.

TS1717 when trying to import cd the following message appears "the current encoder settings for bit rate and sample rate are not valid for this file"?

Trying to import cd when this message appears "the current encoder settings for bit rate and sample rate are not valid for this file". Any suggestions?

Thanks so very much. I chose MP3 and Download and it is working beautifully. Have a wonderful day and it is so nice of you to get right back to me. I'm trying to cut a DVD for my grandson's grad party and I got most of the photos and didn't have any music since I lost everything in a clean install. Could hug you!

The selected combination of bit rate and sample rate is not allowed.

I've always burned my purchased CDs into iTunes using a AAC 320 kbps and 48.000kHz. Now when I select that option in iTunes 9 I'm getting a "The selected combination of bit rate and sample rate is not allowed." pop up screen. So is it impossible to burn my albums in a 320kbps/48.000 AAC format?. I had no problems in previous versions of iTunes using this setting. I guess I might have to switch over using a MP3 import setting instead.

Audio CDs are encoded with a sample rate of 44.1 kHz. Ripping them at 48 kHz requires resampling and does not improve the quality. Unless you have a special need for 48, use 44.1.

Changing audio recording depth and sample rate in QuickTime Pro?

Last week, I set up QuickTime Pro 7.4.5 to record audio in 24-bit, 48 KHz format -- only now, I can't remember how I did that. I think that I may have used some other software to change the settings of my Mac Pro's built-in sound card. Does anyone know how to do this?
I made several such recordings, just last week, as a test, simply using a microphone plugged directly into my Mac Pro, and using QuickTime Pro as the recording application. During playback of such QuickTime audio files, I can hit <command-i> to see that the recordings are indeed 24-bit, 48 KHz. But now, when I repeat the same process, my QuickTime audio recordings are made with 16-bit resolution, at 44.1 KHz. I remember putting the settings back to those values after my recording session of last week. But I can't remember how I adjusted those settings. Does anyone know how to do that?
Thanks.

Thanks for your response.
I will use Apple MIDI Setup, as you suggest.
I want to change the sample rate to 48 KHz, as the final destination is DVD and I don't want to convert from 44.1 to 48 and thus suffer a possible loss in quality.
As for bit depth, I want to sample at 24-bit, since I'll be processing the sound afterwards and don't want audible rounding errors during the number "crunching". Afterwards, I'll just drop the eight least significant bits, or perhaps round them up or down (to zero or 256).

Bessel low pass filter and sampling frequency

I have a 20 kHz sine wave signal, use the Bessel LowPass filter.
The input parameters for the filter are as followings: sampling freq: 204800, Order: 8, low cutoff freq: 30000.
The filtered X has a much lower amplitudue compared to to the input signal.
If I change the sampling frequency to 80000, it seems the filtered X is better.
Can you please explain the behavior of the filter?

Thank you for the reply!
I attached a test vi, which generates a sine wave and calculate its RMS (basic DC/RMS). The sine wave is then passed to a low pass filter, a new RMS is calculated from the filtered signal.
Test case 1:
Frequency = 10 kHz, Amplitude = 4, Sampling Info: Fs = 80 k, #s = 1000; Low pass cutoff frequency = 30 k; Filter sampling frequency = 80k.
The signal RMS = 2.8284; Filtered signal RMS = 2.71.
Test case 2:
Same setup as case 1, change the Filter sampling frequency to 204800.
The signal RMS = 2.8284; Filtered signal RMS = 2.4688.
Test case 3:
Frequency = 20 kHz, Amplitude = 4, Sampling Info: Fs = 80 k, #s = 8000; Low pass cutoff frequency = 30 k; Filter sampling frequency = 80k.
The signal RMS = 2.8284; Filtered signal RMS = 2.21.
Test case 4:
same setup as case 3, change the sampling frequency to 204800.
The signal RMS = 2.8284; Filtered signal RMS = 1.5369
Can you please why increasing the sampling frequency will get lower RMS value?
Thanks.
Attachments:
LowPass Test-1.vi ‏34 KB

How can i know if DAQ hw support "buffered task" and "Sample Timing Hw"?

Hi,
i'm choosing the corrent DAQ hw for my application. I read in the NI-DAQmx documenation that if would use the DAQmxRegisterEveryNSamplesEvent(..) function, the hw should support the "bufferd tasks". How Can i know what DAQ device support this feature? Then, I know not all the DAQ device support the "Sample timing hardware". How can i know what DAQ hw use this timing type? Does the NI PCI-6220/6221 support these features?
Thanks in advace.
PS: sorry .. but i'm new of "NI World".

All of the Multifunction I/O (MIO) products, including the PCI-622x products, support hardware timing and buffered analog input tasks.
In general, if a product's specifications include the analog input sample rate (e.g. "up to 200KS/s") then that means the product supports hardware timing and buffered analog input.
--Joe

Volume, bit depth, and quality - for the boffins

hi y'all
my understanding is that with digital sound that if levesl are too low you lose quality as the signal is not carried using all of the bits.
THis is explained clearly in the book 'mxing with your mind' by michael paul stavrou using a 'photo of a skyscraper' analogy to represent audio clarity (focus) changing with volume (height of skyscraper)
With digital the highest part of the skyscraper is most in focus , whereas in analogue the middle part of the skyscraper is most in focus
3 questions:
1) Is this understanding correct?
2) If you have recorded levels that are low does adding gain or normalizing the files make any difference to the sound quality? - surely detail can't be added
3) When you add gain to a quiet mix is there a difference in the quality of the outcome between the following approaches.
1) normalizing the audio files for each track
2) adding gain to each track in the mixer either by adjusting faders or adding a insert that increases gain
3) adding gain at the mix output bus or master output fader?
Would value anyone's views on this.
best
tommy banana

my understanding is that with digital sound that if levesl are too low you lose quality as the signal is not carried using all of the bits.
Not really.
THis is explained clearly in the book 'mxing with your mind' by michael paul stavrou using a 'photo of a skyscraper' analogy to represent audio clarity (focus) changing with volume (height of skyscraper)\
Unfortunately, this kind of analogy has misrepresented digital audio for a long time, and has caused all kinds of myths, like "digital audio is steppy or discontinuous" and "a higher sample rate gives you more resolution because the steps are smaller" etc etc.
If you have recorded levels that are low does adding gain or normalizing the files make any difference to the sound quality?
Yes, it makes it very very slightly worse.
3) When you add gain to a quiet mix is there a difference in the quality of the outcome
between the following approaches.
1) normalizing the audio files for each track
2) adding gain to each track in the mixer either by adjusting faders or adding a
insert that increases gain
3) adding gain at the mix output bus or master output fader?
Normalising should really never be used. If all these processes are implemented correctly, there should be no practical difference between adding gain to all individual tracks, versus adding gain at the master fader.
This wasn't the case some years ago with poorly implemented digital mixers, but in this day and age, pretty much everyone does it right.
Bottom line, if you really want to understand this stuff it can be useful, but really, improving your song writing will have a much better positive effect on your music than worrying about which dither curve works the best and whether adding +1dB of gain is going to destroy your audio.

Creative Audigy 2 NX Bit Depth / Sample Rate Prob

This is my first post to this form
Down to business: I recently purchased a Creative Audigy 2 NX sound card. I am using it on my laptop (an HP Pavilion zd 7000, which has plenty of power to support the card.) I installed it according to the instructions on the manual, but I have been having some problems with it. I can't seem to set the bit depth and sample rate settings to their proper values.
The maximum bit depth available from the drop down menu in "Device Control" -> "PCI/USB" tab is 6 bits and the maximum sample rate is 48kHz. I have tried repairing and reinstalling the drivers several times, but it still wont work. The card is connected to my laptop via USB 2.0.
I looked around in the forms and found out that at least one other person has had the same problem but no solution was posted. If anyone knows of a way to resolve this issue I would appreciate the input!
Here are my system specs:
HP Pavilion zd 7000
Intel Pentium 4 3.06 GHz
GB Ram
Windows XP Prof. SP 2
Thnx.
-cmsleimanMessage Edited by cmsleiman on -27-2004 09:38 PM

Well, I am new to high-end sound cards, and I may be misinterpreting the terminology, but the sound card is supposed to be a 24bit/96kHz card.
I am under the impression that one should be able to set the output quality of the card to 24bits of depth and a 96kHz sample rate, despite the speaker setting that one may be using, to decode good quality audio streams (say an audio cd or the dolby digital audio of a dvd movie.) I can currently achieve this only on 2. speaker systems (or when i set the speaker setting of the card to 2.) Otherwise the maximum bit depth/sample rate I can set the card output to is a sample rate of 48kHz and a bit depth of 6bits.
Am I mistaken in thinking that if I am playing a good quality audio stream I should be able to raise the output quality of the card to that which it is advertised and claims to have?
Thnx

Importing audio - sample rate/bit depth

Hi forum,
I am working on project at 44.1K, 24 bit. Audio elements are being sent to me to be added. Some have come in incorrectly; at 48K, 16 bit. I can convert easily but didn't think I needed to.
I thought by selecting the "Convert Audio Sample Rate When Importing" option when creating the project that would all be worked out.
That is what I've done - and the file seems to be the correct pitch -- yet shows up in the audio window with it's original specs (48/16). Also ... will logic keep it at 16 bit and play all other files at 24 bit?
I want to be sure about this ... something seems fishy.
Cheers
Dee, Ottawa

Hi,
I am re posting.
Regarding the same project: must everything in the arrange window of a project be the same sample rate (and bit depth). My understanding is that there is real- time conversion during playback. That all file types supported by logic - and all virtual instrument samples are converted in real time to conform to the selected bit depth and sample rate of project.
I ask only as I recieved reference sound files to temporarily place in a mix to see how mix will sit when going to Post. Two audio files are almost a semi tone higher than they are supposed to be (which is odd - so I am pretty sure it was just quickly sung in the wrong key at there end). And one file which was supposed to be timed out is not lining up.
I can work around this on this project. And I can simply convert in the sample editor and re import to compare.
But again ... I just want to check my understanding for future reference. The manual indicates differing rates etc should not be a problem. (ie: That Logic allows one to have differing rates and bit depths. Inversely, the M Sitter video implies just the opposite.
I just want to be sure of this for future reference. Any adive?
Thanks in advance.
Cheers
Dee

Supported bit depth and sampling frequency

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