Maximum Bitrate/Sample Rate?

What is the maximum bitrate and sample rate(44.1/48k) that can be read by my iPod? Does it depend on whether I use mp3 or aac encoding? Can I use variable bit rate (VBR)? If I can use a 48k sample rate, would that mean using a smaller bitrate?

Thank you, that article was helpful, but it did not say what bitrates could be used with AAC. MP3 is from 32 Kbps to 320 Kbps, can I use 320 Kbps for AAC too? What about sample sizes for WAV and AIFF?

Similar Messages

Maximum audio sample rate and bit depth question

Anyone worked out what the maximum sample rates and bit depths AppleTV can output are?
I'm digitising some old LPs and while I suspect I can get away with 48kHz sample rate and 16 bit depth, I'm not sure about 96kHz sample rate or 24bit resolution.
If I import recordings as AIFFs or WAVs to iTunes it shows the recording parameters in iTunes, but my old Yamaha processor which accepts PCM doesn't show the source data values, though I know it can handle 96kHz 24bit from DVD audio.
It takes no more time recording at any available sample rates or bit depths, so I might as well maximise an album's recording quality for archiving to DVD/posterity as I only want to do each LP once!
If AppleTV downsamples however there wouldn't be much point streaming higher rates.
I wonder how many people out there stream uncompressed audio to AppleTV? With external drives which will hold several hundred uncompressed CD albums is there any good reason not to these days when you are playing back via your hi-fi? (I confess most of my music is in MP3 format just because i haven't got round to ripping again uncompressed for AppleTV).
No doubt there'll be a deluge of comments saying that recording LPs at high quality settings is a waste of time, but some of us still prefer the sound of vinyl over CD...
AC

I guess the answer to this question relies on someone having an external digital amp/decoder/processor that can display the source sample rate and bit depth during playback, together with some suitable 'demo' files.
AC

Verify my maximum achievable sample rate .

I am using a PXI 6031E and an SCXI 1001 Chassis populated with six 1102c modules. I am setup to multiplex 192 channels to one DAQ channel(0). I am using every channel on every module. The 1102c is cable of 333KS/sec but using all 32 channels my max should be a little more than 10Khz. My PXI 6031E is capable of 100KS/sec but now I am limited by the module speed so the best I can get is around 10Khz. Is this correct?

I have a problem that's kinda similar.
We are using a PXI 6031E in a PXI 1036 housing connected to a computer with a PXI 8360 MXI card.
I was trying to measure 17 Channels at 10k, but it just didn't work. According to the Datasheet, the max. sample rate of the 6031E is 100kHz.
Then I figured, that maybe the whole sum of all used channels may be 100kHz.
The maximum samplerate that works is 7262 Hz.
But 7262 * 17 is about 123 000, which is more then 100 000.
Could someone could explain, whats behind that?
Attachments:
highsamplerate.PNG ‏24 KB

Finder - audio file info - bitrate, sample rate, etc

Is there a way to find out what bit rate and sample rate an audio file is without opening it in QT and then CMD+I from there?
I don't see detailed info when I get information from Finder. Is a script or anything that will allow me to find this info quickly in Finder?

juhani h. wrote:
Works fine here, too. But make sure you´ve ticked the box "Convert audiofile...etc", like M-Dan pointed out.
I tested it and if unchecked the imported file plays at the wrong speed. Amusing at first, but potentionally disastrous.
/J
Does NOT work in repeated tests on my system, and yes, the 'convert audiofile' box is ticked. Wouldn't report it as a bug if it wasn't....
This almost ruined a session of mine and cost me a deadline, but luckily I caught the SNAFU in time and converted the files individually.
Very odd though to hear that it works on some users' systems. How can this be a system specific bug? Are you sure you're dragging files in from the Finder to your arrange page? And Logic converts them? Are we talking 48k files being converted to 44? Or vice versa (44 to 48)?
In my case, it was a 48k session and Logic failed to convert the 44k files I had mistakenly been sent.
Since I've noticed other weirdness (like malfunctioning Apple Loops) at 48k, I still think Logic is less than bullet proof at different sampling rates. Just a hunch though.
T

Does anyone know how to find maximum sample rate on an mac 10.7.5

Trying to see if I can downlaod a audio plug-in that requires a maximum sample rate of 199kHz. Where do I find this info for 2009 Mac 10.7.5?

According to http://support.apple.com/kb/ht3913:
"The internal microphone supports recording at bit depths of 16, 20, or 24 bits per sample and at sample rates of 44.1 kHz, 48 kHz, or 96 kHz."

Changing sampling rate and bitrate

If I drop in an .aiff audio file into Garage Band 2 (or iTunes), is there a function that allows me to adjust the audio sampling rate and bitrate?
When I created this podcast in Adobe Auditions, the were about 10MB for a 22 min show.
When I do it through GB, it's about 26MB for the same time;
I just want to tune it down slightly.
Thanks!

GB exports uncompressed 44.1K 16-Bit AIFF files. You need to convert them to Mp3s or AAC files
http://thehangtime.com/gb/gbfaq2.html#converttomp3

Sample Rate And Bitrate Not Allowed ... ?

I have almost 70GB of my music in my iTunes, ALL in AAC 320kbps 48Khz. Now that I just updated to iTunes 9, I just tried to add more of my new CD's to my library and they are being imported at 44Khz...?? I went to change it back to 320/48 and I get ...
"The selected combination of bitrate and sample rate is not allowed"
Bug? Anyone else have this problem? HELP!
btw ..I tired to use my laptop and my work PC to see if it's just this machine that is causing the problem....and those too (now updated to iTunes 9) are doing the same thing.

I don't know why a previous version might have allowed a 48khz sample rate, but audio CDs are 44.1kHz (that sample rate is locked into the Red Book CD standard) so there would be no good reason I can think of to attempt to import at a 48KHz sample rate. It won't increase quality and could cause artifacting.
Regards.

Sample rate too high for swf or PowerPoint export

Hi,
I have a few presentations that need to be exported to swf or PowerPoint. Each of the slides use audio Voice Over. The problem is that when I go to export I'm told that because the audio uses a sample rate of 48khz it can't be exported.
Anyone know if there is a way around this?
Thanks

Yes I meant to say AC3 (Dolby Digital) instead of
AAC.
Sorry for the confusion.
OK, thanks for clearing that up.
What bitrate (avg./max.) are you using for the video? With three angles, the maximum combined bitrate for each track (including audio) cannot exceed 8Mbps.

Premiere Pro CC and Encore CC are ignoring the Target and Maximum Bitrates

Adobe Premiere Pro CC and Adobe Encore CC are ignoring the Target and Maximum Bitrates settings in the export window, The files that are exported are much smaller than what the estimated size of the file should be shown in the export window and are highly compressed. What can cause such an issue?
I haven't drastically changed my presets.

There is a very definite bug in the Adobe Media Encoder engine which is used in both Premiere Pro and After Effects. H.264 exports seem to randomly choose their own export quality, using 2-pass VBR. I have found with some DV-size exports on 'match quality' in Premiere Pro, that setting the export above 10Mbps pushes the quality down to well under 1Mbps, e.g. 167kbps. I've found that using the default min 6, max 8 seems to work the best. I've worked with this on two different machines and fiddled with it extensively and this is definitely NOT an intended 'feature'. In fact, I have found that exporting from the same sequence, if I select the first 20 seconds, I get very low bitrate, but if I select 30 seconds or more, I get the higher bitrate that I want! And I tend to get different results when I enable and disable 'use previews'.
I am also having some trouble with missing frames when exporting progressive animation files to h.264 mp4 files that never occured before. Not sure yet if this is related to the same issue, but interestingly, I don't get the 'blinking' missing frames issue when the bitrate drops itself down to a very low 150-200kbps. When I do get the export rate up higher, then the missing frames on just this animation clip return.
So, after my tests, here is a screen capture of sample export settings (yes, 15Mbps is a very high bitrate for DV, but the same happens at 6Mbps or 7 or whatever I try in some cases):
This gave me a 1.5MB file with a reported 167kbps bitrate and a quality level as shown below:
A currently working 'workaround' is to use 1-pass VBR which gives me a crisp, clean 5.5MB file. Would a few others please try to reproduce this so we can get it escalated to bug status? I'll report it if you can confirm it there, this is happening on two computers here.
Thanks,
Sean

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

IPhone: AudioQueue - is it possible to change the sample rate?

I've been playing around with the AudioQueue stuff for a few days and it's all working fine.
I was trying to build a low-latency playback system by making the streaming buffers the same size as the audio file and pre-loading the buffers (which works fine) but I've hit a snag.
I've been trying to get the streaming to work at different sample rates so that I can play back the same sample at different pitches. I managed to do it by modifying the sample rate in the AudioStreamBasicDescription structure but in order to actually make the stream playback at the new rate it seems you have to create a new output, reload the audio file into the buffers and re-enqueue the output queue before starting playback again, otherwise the sample rate change has no effect.
There is a method to set queue properties; AudoQueueSetProperty() but unfortunately the sample rate Property (kAudioQueueDeviceProperty_SampleRate) is read-only
Can anyone suggest a way to achieve this with AudioQueue or do I need to move over to OpenAL?
Thanks,
Neil

Dan,
there is one point in your understanding, which i am not sure what you think about when talking about it: I understand E-series devices do not support this property change while the VI is running.
infact, you cannot change the sample clock rate during acquisition. but
this does not mean that you cannot change it while the VI is running.
you have only to interrupt the acquisition. since you want to acquire
continuous, this would have the same effect as stopping the vi, i asume.
so the best way to accomplish this task is to use an external clock.
this is e.g. often used for acquistion on rotating shafts. the
acquistionrate is always e.g. 24 points per revolution regardless of
the rotational speed of the shaft, except for a maximum frequency of
course.
Norbert B.
NI - Germany
Message Edited by Norbert B on 09-14-2005 04:16 AM
CEO: What exactly is stopping us from doing this?
Expert: Geometry
Marketing Manager: Just ignore it.

Bit Depth & Sample Rate: 24 bit 96kHz? 192kHz?

I am using the Apogee Duet for Mac and iOS on my Mac and I love it - I'm thinking about getting an iPad for mobile recording (voice overs, mostly) and I wonder if Garage Band can manage 24 bit audio at 96 kHz or 192 kHz? I know that the Auria app can, so if nothing else I can just buy that, but since all I would use the iPad for is Voice Overs to edit later in a computer, a $50 app feels like overkill. Comments? Thoughts? Specs?

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

Recording LP records as source material- Sample Rate

Using recorded tracks from LP records to make DVDs, Blu-Ray DVDs or simple CD's. Am not sure what maximum sample rate to use. I understand the end product limits of the various digital media, but LPs are analog. Do I gain any sound quality by recording the original LP at a sample rate higher than 48000/32bit, say 96000 sample rate) and then resampling (downsizing) the audio file if the end product cannot produce the higher sample rate?

Conversion de LP -Archivos Digitales
Se recomienda Grabalos con estas velocidad de Muestreo
Blue-Ray 96000Hrz. / 32 bits
DVD & 48000 Hrz. /32 Bits
CD DE AUDIO A 44100Hrz /32 Bits ó 24Bits
Te recomendaria Cambiarte a Adobe Audition
Saludos
http://soundcloud.com/creativoxpro/restaurando-audio-de-un-vinil
Para audio
8000 muestras/s
Teléfonos, adecuado para la voz humana pero no para la reproducción musical. En la práctica permite reproducir señales con componentes de hasta 3,5 kHz.
22050 muestras/s
Radio En la práctica permite reproducir señales con componentes de hasta 10 kHz.
32000 muestras/s
Vídeo digital en formato miniDV.
44100 muestras/s
CD, En la práctica permite reproducir señales con componentes de hasta 20 kHz. También común en audio en formatos MPEG-1 (VCD,SVCD, MP3).
47250 muestras/s
Formato PCM de Nippon Columbia (Denon). En la práctica permite reproducir señales con componentes de hasta 22 kHz.
48000 muestras/s
Sonido digital utilizado en la televisión digital, DVD, formato de películas, audio profesional y sistemas DAT.
50000 muestras/s
Primeros sistemas de grabación de audio digital de finales de los 70de las empresas 3M y Soundstream.
96000 ó 192400 muestras/s
HD DVD, audio de alta definición para DVD y BD-ROM (Blu-ray Disc).
2 822 400 muestras/s
SACD, Direct Stream Digital, desarrollado por Sony y Philips.
Para vídeo
50 Hz
Vídeo PAL.
60 Hz
Vídeo NTSC.
*informacion extraida para apoyo de la pregunta en el foro // http://es.wikipedia.org/wiki/Frecuencia_de_muestreo
> http://en.wikipedia.org/wiki/Sampling_rate

Sample Rate & Timing

I'm using a DAQmx task to continuously acquire analogue data.
I have used Task.Timing.ConfigureSampleClock to specify a 'rate' of 1000 and a 'samplesPerChannel' of 1000. So, I expect to receive 1000 points of data every second and this is exactly what I seem to get in my AsyncCallback.
However, I want to also generate an array containing the elapsed time since start of capture. My plan was to start at zero and then increment values in my time array by Waveform.Timing.SampleInterval however I find that Waveform.Timing.SampleInterval is 0.00062 seconds - why is it not 0.001 seconds?
I then double-checked the task sample rate by using Task.Timing.SampleClockRate and this seems to be 1612.9... - why is it not the 1000 I set during my call to ConfigureSampleClock?
Any pointers would be appreciated.
TIA

Thanks, you're right - I was specifying a sample rate lower than was available for the module I was using (the NI9239). I actually was expecting an exception to be thrown if I attempted that.
So, to make sure I don't get into this situation again I can make sure I select from the supported sample rates of 50/1→31 kS/s for this particular module.
Would you know if there is a programmatic way of finding the available sample rates for all module types? Perhaps I can assume it will always be Maximum Sample Rate/1→31?
TIA

How do I change the audio sample rate from 48kHz to 44.1kHz for Mpeg2

Hey all,
I've been searching for a while but haven't found any direct answers in the forums or the user manual so here goes.
I have to dispatch a file to Bloomberg TV and the file specs they have given me are as follows:
Video Standard: MPEG-2, MP:ML, 4:2:0
Frame Rate: PAL 25fps
Video Size 720 x 405
Aspect Ratio 16:9
Audio Standard MPEG-1 Layer 2
MPEG-2 Program Stream Mux rate 6mbs per second
Bit Rate Type: CBR
Video Bit Rate 5.7mbs
Audio Bit Rate 192kbs
Audio Sample Rate 44.1Khz
Interlacing: Upper Field first (why they want interlaced for web streaming is beyond me)
GOP Structure: IBBP
I-Frame distance 12
Now everything above is fine except the audio encoding because even though I have set up a new setting from scratch I cant find anywhere to adjust the audio sample rate. The Inspector tells me the Audio encoder is set to:
Format: MPEG
Sample Rate: 48.000kHz
Channels: 2
Bits Per Sample: 16
Anyone Know how or even if I can change these audio settings? The only adjustments I can find are the filters or the transport/program stream option. I have it set to program as specified by Bloomberg.
Thanks in advance
J

The only setting that I could find in compressor that lets your change the bitrate to 44.1 is when you create a new dolby digital setting and then under the inspectors audio tab/Target System button, change the button to Generic AC-3. When done, you can change the Sample Rate to 44.1.
Hope this helps?

Maximum Bitrate/Sample Rate?

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