How do I determine that my audio sample rate matches my tape?

How do I determine that my audio sample rate of my capture preset matches the sample rate of my tape?

You've really got two issues.
DV records audio and video differently and if the camera is substandard, you will have drift over long captures. Solutions: new, better camera or capture in small (max 10 minutes) chunks.
The other issue is the 32k sample rate, which will exacerbate issue 1. You can duplicate the 48k preset and modify the copy for 32k and work in that until you are ready to deliver. Then copy it into a 48k sequence, render and output.
It's funny, sometimes I get that pop up but everything is fine.

Similar Messages

How can I find out the audio sampling rate of BetacamSP tape?

Hi guys
I'm trying to digitize BetacamSP tape. But I'm afraid if I might choose wrong setting...
This tape is from very long time ago so we don't know which audio sampling rate we recorded with..
How can I find out the audio sampling rate of this BetacamSP tape?
Thanks:)

The sampling rate is set by the Sony DVMCDA2 you are using, when the conversion is made from the analog input to the digital (DV) output. You should be outputting standard DV which is 16bit 48khz audio.
Assuming you are in the US, your Easy Setup for FCP should be DV-NTSC, and then open the Log and Capture Pane and set the Capture Settings Device Control to Non-Controlable Device and you should be good to go.
You will have to roll the deck manually and start and end your capture manually.
You can download a user manual for the DVMCDA2 by clicking here.
MtD

How can I ensure that the audio sample rate of my capture preset matches?

Hi
1. I have found that none of my ten projects has sound, although audio meters & settings moves up & down.
Simultaniously, canvas displays that In: Not Set- & Out Not Set. Below those reads: Video V1: 00:03:42;13 Audio A1: 00:03:42;13 A2: 00:42;13. These happen on the images of all my ten projects when the playhead stops in timeline
2. In Viewer the In & Out sets show a thin red line connecting them.
I never had this problem before. A/V of all projects were doing just fine till last night.
I wonder if these issues are interrelated, or if I may have clicked something wrongly that has triggered this, which I don't remember now.
I would appreciate it if you could kindly address this problem and help me to resolve the isssue. Thank you. Faruk.

The red line indicates the audio must be rendered. This may be caused by several things but your playback settings may be set too high for your system to handle. You may have too many audio tracks for your computer to play in realtime.
Select one of the audio files in the timeline, get properties, not the audio format and sampling rate.
Now got to Sequence>Settings and determine if the settings match your audio settings.
Your online help system can guide you through adjusting your system settings, user settings and sequence settings.
bogiesan

How can I ensure that the audio sample rare of my capture preset matches?

Hi everyone
When capturing tapes I get warning that the audio sample rate of one or more of captured files does not match the sample rate on my source tape. This may cause the vidio and audeo of the media files to be out of sync. How can I make sure that the my capture preset matches the sample rate of my tape? Can anyone be able to show me how? Thank you. Faruk.

Hi
Fuerther, I have double checked and found that none of my ten projects has sound, although audio meters settings moves up & down. Simultaniously, canvas displays that in- & out of clips are not set , and in browser I see time codes on the images when the playhead stops in timeline. I did not have this problem before. I wonder appreciate if if these issues are interrelated, and if I may have clicked something that has triggered this.
I would appreciate it if you or other friends could kindly address this problem and help me resolve the isssue. Thank you. Faruk.

How do I capture at an audio sample rate of 44.1 KHz?

Hey,
I'm wanting to capture some footage that's been recorded at a 44.1 KHz sample rate. Presets on FCP 5 only have 48 KHz.
How do I change this?

I just checked the import settings on two other editing systems I use... Canopus DV Storm II and the Matrox RTX 100 Extreme. Only the Canopus has a preset for 44.1k import. Even Adobe Premiere, using the native IEEE 1394 connection, doesn't have it. I am not sure if FCP has a "CUSTOM" import dialog that might allow you to manualy select?
If you have found this information helpful, please select the icon for this. If it helps you resolve this issue, please select "answered".

Audio Sample Rate Query

Hi FCE users,
I am about to start capturing 4 hrs of wedding video. I have a fair bit of experience on FCP but am new to FCE and its variations in functionality.
When I ran a quick capture test, it warned me on completion that the audio sample rates were different. The video was shot in Hungary and I have no way of contacting the videographer, however I assume that it was recorded in 32khz as when I changed the capture preset to this, no warning message occurred. The DV cam I'm using most helpfully doesn't display a bit rate but given that no warning occurs after capture, I'm happy to go with the 32khz preset.
Most of the articles posted on the net however state that the 'SEQUENCE PRESETS' as well as the 'Capture Presets' need to be changed. Almost all of it relates to FCP, as opposed to FC Express.
The equivalent on FCE - 'Sequence Settings' doesn't seem to offer a 32khz option, and the Audio Rate shown in the Browser seems to be fixed to 48khz.
My question is:
Does the sequence setting variation matter? - Do I assume that given the scaled down functionality of FCE that it takes care of this automatically OR that 48khz is a fixed audio rate for sequences in FCE and all non 48khz audio clips needs to be rendered prior to output.
No doubt I will be bringing music from CD's in to the project and sequence also, so your help would be greatly appreciated.
I don't want to bring in 4 hours worth of footage and then have to recapture!
Thanks for your help!
Rich

Thanks for your swift reply Tom.
I've just re-reviewed the menus. I can display the 12/16 info on record, but it doesn't offer it as a playback display option. However... I've just recorded more footage into FCE and received no error message with the capture preset on 32 so can I be fairly confident that its 32?
I've just figured out how to change the sequence format to 32 (it seems to be created by default with capture preset set to 32, wheras I was trying to alter an existing sequence).
Rich

Problem with audio sample rate when exporting

I am having an issue with my audio sample rate. For some reason it is set at 8 khz by defualt when I am exporting. This was not an issue until the latest update (2 weeks ago). I'll try and give all the info I can up front. Also I tried customer support...that was a nightmare.
Ok the issue I am addressing is that the audio sample rate when exporting is at 8khz by default. Which to my knowledge has NO relevant use what so ever. Even when I set the file format to h.264 it insists on resetting to 8khz. Even if I manually set everything look at the render queue and click on output module again it changes the sample rate back to 8 khz automatically which I must adjust again. Why can it not just be set at 48000 like every other adobe product is and like it used to be two weeks ago. Does anybody have an answer?

This problem is addressed by the After Effects CC (12.2.1) bug-fix update, which is now available:
http://adobe.ly/AE_CC_1221
Note the part at the end of that page about a crucial update for the Creative Cloud desktop application, which addresses some severe problems with AME, Premiere Pro, and After Effects.

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?

Determining DV tape audio sample rate

I have DV tapes of "archived" final edits recorded on to mini DV tape using two "early" (late 90's) Sony DV cameras with some segments of the same tape recorded with 48k audio and other sections with 32k audio (on the same mini DV tape). I tried to capture the footage into FCP 5.1.2 using a Sony DSR-11 deck and a G4 1.42 GHz dual processor w/ 2 gigs RAM, Radeon 9800 Pro video card, Mac OS 10.4.8
Everything seemed fine until I got a message after capturing a 3 minute clip using a stock FCP 5 NTSC preset with 48k audio; the message stated the "audio source rate and capture sample rates did not match & footage may be out of synch" - the captured clip played back perfect. The DSR-11 did not display the DV tape's audio sample rate; I put the DV tape into the camera it was originally recorded on and "16 bit" displayed on the camera's LCD screen.
I tried capturing the same footage using FCP 4.5 installed on another drive in the same computer, and got no warning message re: audio sample rate. Just to "be sure", I created a custom capture preset that included 32 bit audio and captured the same clip using FCP 4.5 with a preset for 32k audio - no warning message, and everything was perfectly in synch. Everything was in synch for ALL the captured clips & the audio sample rate in FCP for captured footage was either 48k or 32K for each respective preset.
The dilemma: how to be certain what the true/original audio sample rate is.
G4 1.42 GHz Mac OS X (10.4.8)

"audio source rate and capture sample rates did not match & footage may be out of synch"<<</div>
William,
The message itself seems to be a bug that started with FCP 5 - most users that receive this message have reported that there is NO actual problem.
-DH

Async tcp client and server. How can I determine that the client or the server is no longer available?

Hello. I would like to write async tcp client and server. I wrote this code but a have a problem, when I call the disconnect method on client or stop method on server. I can't identify that the client or the server is no longer connected.
I thought I will get an exception if the client or the server is not available but this is not happening.
private async void Process()
try
while (true)
var data = await this.Receive();
this.NewMessage.SafeInvoke(Encoding.ASCII.GetString(data));
catch (Exception exception)
How can I determine that the client or the server is no longer available?
Server
public class Server
private readonly Dictionary<IPEndPoint, TcpClient> clients = new Dictionary<IPEndPoint, TcpClient>();
private readonly List<CancellationTokenSource> cancellationTokens = new List<CancellationTokenSource>();
private TcpListener tcpListener;
private bool isStarted;
public event Action<string> NewMessage;
public async Task Start(int port)
this.tcpListener = TcpListener.Create(port);
this.tcpListener.Start();
this.isStarted = true;
while (this.isStarted)
var tcpClient = await this.tcpListener.AcceptTcpClientAsync();
var cts = new CancellationTokenSource();
this.cancellationTokens.Add(cts);
await Task.Factory.StartNew(() => this.Process(cts.Token, tcpClient), cts.Token, TaskCreationOptions.LongRunning, TaskScheduler.Default);
public void Stop()
this.isStarted = false;
foreach (var cancellationTokenSource in this.cancellationTokens)
cancellationTokenSource.Cancel();
foreach (var tcpClient in this.clients.Values)
tcpClient.GetStream().Close();
tcpClient.Close();
this.clients.Clear();
public async Task SendMessage(string message, IPEndPoint endPoint)
try
var tcpClient = this.clients[endPoint];
await this.Send(tcpClient.GetStream(), Encoding.ASCII.GetBytes(message));
catch (Exception exception)
private async Task Process(CancellationToken cancellationToken, TcpClient tcpClient)
try
var stream = tcpClient.GetStream();
this.clients.Add((IPEndPoint)tcpClient.Client.RemoteEndPoint, tcpClient);
while (!cancellationToken.IsCancellationRequested)
var data = await this.Receive(stream);
this.NewMessage.SafeInvoke(Encoding.ASCII.GetString(data));
catch (Exception exception)
private async Task Send(NetworkStream stream, byte[] buf)
await stream.WriteAsync(BitConverter.GetBytes(buf.Length), 0, 4);
await stream.WriteAsync(buf, 0, buf.Length);
private async Task<byte[]> Receive(NetworkStream stream)
var lengthBytes = new byte[4];
await stream.ReadAsync(lengthBytes, 0, 4);
var length = BitConverter.ToInt32(lengthBytes, 0);
var buf = new byte[length];
await stream.ReadAsync(buf, 0, buf.Length);
return buf;
Client
public class Client
private TcpClient tcpClient;
private NetworkStream stream;
public event Action<string> NewMessage;
public async void Connect(string host, int port)
try
this.tcpClient = new TcpClient();
await this.tcpClient.ConnectAsync(host, port);
this.stream = this.tcpClient.GetStream();
this.Process();
catch (Exception exception)
public void Disconnect()
try
this.stream.Close();
this.tcpClient.Close();
catch (Exception exception)
public async void SendMessage(string message)
try
await this.Send(Encoding.ASCII.GetBytes(message));
catch (Exception exception)
private async void Process()
try
while (true)
var data = await this.Receive();
this.NewMessage.SafeInvoke(Encoding.ASCII.GetString(data));
catch (Exception exception)
private async Task Send(byte[] buf)
await this.stream.WriteAsync(BitConverter.GetBytes(buf.Length), 0, 4);
await this.stream.WriteAsync(buf, 0, buf.Length);
private async Task<byte[]> Receive()
var lengthBytes = new byte[4];
await this.stream.ReadAsync(lengthBytes, 0, 4);
var length = BitConverter.ToInt32(lengthBytes, 0);
var buf = new byte[length];
await this.stream.ReadAsync(buf, 0, buf.Length);
return buf;

Hi,
Have you debug these two applications? Does it go into the catch exception block when you close the client or the server?
According to my test, it will throw an exception when the client or the server is closed, just log the exception message in the catch block and then you'll get it:
private async void Process()
try
while (true)
var data = await this.Receive();
this.NewMessage.Invoke(Encoding.ASCII.GetString(data));
catch (Exception exception)
Console.WriteLine(exception.Message);
Unable to read data from the transport connection: An existing connection was forcibly closed by the remote host.
By the way, I don't know what the SafeInvoke method is, it may be an extension method, right? I used Invoke instead to test it.
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How do we determine that there is a need of RFK (OR) CKF???

Hi gurus,
can anyone explain me how do we determine that there is a need of RFK (OR) CKF???
Thanks in advance..
S Ram

Are you asking
- How to determine whether to use RKF or CKF? Ie RKF Vs CKF?
or
- How to determine when there is a situation when you have to create one (CKF or RKF, anything)
Answer to both is, the requirement should make the choice obvious. RKF and CKF are for entirely different reason and can't replace each other. So, you will know based on your requirement whether it can be fulfilled by one, or the other.
The same for the second question - if you do not have a basic KF which fulfils what you want to report, you will try to see if an RKF or CKF can fit in.

How is Core Audio sample rate set?

When I play a particular movie in QuickTime, the audio and video are out of sync. The movie plays fine on other computers.
This Mac Pro (OS 10.6.8) is used exclusively for Pro Tools and Final Cut Pro. The audio hardware is a Pro Tools HD Native card:
Native card --> Pro Tools Digital I/O boxes --> Lavry D/A converters --> monitors
The sample rate of the Digital I/O boxes is set by an external master clock, a Lavry Gold A/D converter.
I opened the DigiDesign Core Audio Manager and it says:
Connected @ 44.1K, 32 In/32 Out, Buffer Size 512
Yet the movie is at 48K (I know because I created it in Final Cut Pro), and the external clock is set at 48K. So, I don't know where the 41K in the Digidesign Core Audio Manager came from. This is why I am guessing the problem is sample rate.
Please help me understand what sets the sample rate.
Does the application using Core Audio set it?
If so, how is this made consistent with the sample rate set for the hardware by the external master clock?
Do I have to be sure I always change the external clock setting to be consistent with the movie being played?
BTW, I have not had a sync problem when I play video in Final Cut Pro..........it is always in sync. So, I have never worried about how Core Audio works. The sync problem is only with QuickTime.

You might ask on the GarageBand, Logic or Final Cut forums. There is not much audio traffic here.

How do you set sequence audio sample rate?

I tried posting this to another, but it was already answered, and noone will see it.
I am getting the capture error "audio sample rate doesn't match" and yes, I can see in my browser that the clip is 48khz/16bit, but the sequence is 48khz/32 bit. Howver, wherever I look to change the sequence setting, it is 48/16 already. I've gone to FCP on the menu dropdown to audio/video settings - it's correct all through there. I've gone to the menu dropdown Sequence settings, and it's correct there. I've closed down, opened a new sequence, restarted, everything I can think of. Is there a secret to getting them to match? And, can I fix a project already edited with this discrepancy? Its export to QT is WAY out of synch.

Annoying - I can't see your post when I am in reply mode.
Yes, I get this error when i am capturing. From reading a previous post about the error, I thought the solution was to check the audio rate of the captured clip in the browser, and then make sure it matched the audio rate of the sequence. Like I said, everywhere you get to change the sequence setting, it SAYS it is 48/16, but yet, whe I scroll over in the browser, it says the audio rate is 48 KHz and the audio format is 32-bit floating point. Am I looking at the right places?
I can't check the settings in the camera until this evening...don't ask.
I'm not so sure this is not a QT issue instead of anything to do with capturing, etc. It plays back fine in the timeline.

Capturing issues-audio sample rate & locating timecode break

I am a first time FCP user.
I am working on an educational video with hours of DV source tapes.
The source tapes are from a 2R50MC Cannon mini DV camcorder also used as the deck I am capturing from via firewire. My scratch disk is Maxtor III 500GB external drive hooked up via a firewire (I'm realizing the firewire is years old could that be a problem?)
In my first capture test I captured 1 minute of tape by setting in and out points and after completeing the capture I received an error message—"The audio sample rate of one or more of your captured media files does not match the sample rate on your source tape. This may cause the video and audio of these media files to be out of sync. Make sure the audio sample rate of your capture preset matches the sample rate of your tape."
For the past day I have been scouring the user manual, this discussion group and training tapes from Lynda.com and can not resolve this issue. I have determined that the source tape is 12 bit or 32 kHz and yet can not find a way to set up capture preferences to 32 kHz. Is this the problem/resolution? Page 320 of the user manual shows the QuickTime Audio Settings dialog box but I can't find it in FCP or Quicktime. Is this where I make the change to 32kHz?
After doing the first test, I tried again and ran into a new problem—it stopped capturing with the message: Locating timecode break [press esc to abort]. It never seemed to resume capturing and I pressed escape to abort. If I'm reading FCP time code window correctly I received this time code break message within a section I had already captured previously (with the audio sync warning).
I have done several capture tests at different points in the tape and on different tapes all with the same results. I've used the capture clip and capture now buttons. I've tested with drop frame turned on and off. Confirmed my setting of: At timecode break "Make New Clip." Confirmed my Easy Setup as DV-NTSC. My capture preset is DV NTSC 48 kHz. I've turned off and on FCP. Restarted my computer. Restarted my computer with the shift key down and ran permissions.
Any ideas are GREATLY appreciated, T.

I am concerned about available memory when capturing to my hard drive and the babysitting and extra steps involved considering the amount of tape I want to capture...but it does work that's GREAT!
This is always a concern, but in your case I think having the camera and external on the same bus was causing your problem. You may have to capture a little and then transfer, rinse and repeat. Just don't try to do too much at once and let your System Drive get too full, you'll run into other problems there. Slow and steady is the pace!
I shouldn't have stacked my questions since you answered one and Chris answered the other. Don't know how to apply the answered question and who gets the points.
No problem, just mark it answered and divide up the solved and helpful points as you wish. All in all we really don't care too much about the points, but they do make us feel good! Thanks for your desire to use the forums properly.
K

Highest audio sampling rate in CS4?

Hello,
I apologize if this has already been asked, but I have been searching everywhere and I simply cannot find the answer to this.
What is the highest audio sampling rate that can be utilized in Premiere Pro CS4? Can it import and export 192kHz 24-bit audio?
Thanks in advance

Hey Hacienda,
I might not have the experience in audio work you have since I've only been doing this for the past 6 years or so. But I've been a musician for far longer than that, and I've learned A LOT mostly from really smart people in the industry. So, I'm not gonna lie to you and say that I've done extensive testing in this area because I simply do not have the equipment, nor the money to buy it (WAY too expensive). But we do share the neophyte status when it comes to video editing :-P
Anyways, the Nyquist Theorem is not a theory, which is what people are led to believe. It is a theorem, meaning it's already mathematically proven. It is proven that, as long as you follow the premise of capturing twice the highest frequency of the sound source, you'll get a perfect reproduction of it. To capture more than that is a waste of bandwith specially because most people won't even hear above 18KHz, nor do they have the equipment to reproduce such frequencies. Most consumer systems and audio gear, including those found in professional studios, go up to about 22KHz. You need to spend BIG dolars for anything that goes beyond that. So, who are we really making music for here? The super rich? Dolphins?
Now, I know you're not just talking about higher frequencies, but the amount of samples needed to recronstruct a perfect copy of the original waveform. OK, well, this is the kind of snake oil marketing BS I was talking about. The biggest one being that 1bit DSD crap that Sony/Phillips is pushing. Adding more samples to the recording will not make any difference on how faithfully you can reproduce a sound. It'll just make the files bigger for no reason. Again, the Nyquist Theorem already proves this. This is FACT! Here's a link I found interesting regarding these audio industry lies, maybe you will too: http://theaudiocritic.com/back_issues/The_Audio_Critic_26_r.pdf It starts on page 5, but the one pertaining this discussion is lie #3 on page 6. :-D
Don't forget that modern converters already sample at much higher frequencies than the target sampling rate. I believe my RME Fireface 400 samples at 5.6MHz, which is twice the amount of samples compared to DSD technology, before going back down to the target rate. But, like I said, it does so for other reasons and NOT because it needs that many samples in order to faithfully reproduce a waveform. Of more importance are the quality of the FIR (Finite Impulse Response) filter and the clock inside the converters. These components are what make a converter high grade, among others. The converter chips themselves are very inexpensive (in the tens of dolars) which why you hear some companies advertizing having the same converter chip as a ProTools HD rig (not the best example I know).
By the way, I didn't say humans can only hear up to 20KHz. I'm sure there are people who can hear above that. My point was that the 20Hz - 20KHz range is what's generally accepted as an average for humans (which implies that there are people who can hear avobe/below that). Also, the reason why modern-day pop records causes headaches and sound horrible is because of a totally different issue known as "The Loudness War" (I'm sure you know about it so I won't go into details). However, I do agree with you as far as compressed audio goes. Unfortunately, there's a reason for that and there's nothing we can do about it until the day Internet bandwith becomes more accessible and cheaper. Eventually it'll get to the point where uncompressed audio can be streamed reliably through the net. But, until then, we're stuck with MP3, AAC, DTS and other audio compression formats. As far as digital media distribution goes, it's the future and companies are seeing that. More and more people download music rather than buying CDs, so I do believe those numbers are accurate. Just look at sales from iTunes and even games like Guitar Hero and Rock Band. It's just a matter of time.
Take care!

How do I determine that my audio sample rate matches my tape?

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