Audio Data Rate digitizing DVCPRO50 over firewire

I'm having this weird issue when I digitize DVCPRO50 footage into one of my Dual 2Ghz G5s. For some reason, once the clips are in (I'm bringing in entire tapes) -- the audio data rate is at a strange number (like 47722 khz) rather than the standard 48khz. And that number is not consistent -- it ranges from 37782 to 47991. Has anyone ever seen this? It seems to be something with this machine or version of FCP because I can digitize on another G4 or G5 running 4.5 and it comes in fine. But then I have to transfer it via FW drive which is a big hassle. The specs are: Dual 2ghz G5, Panasonic DVCPRO50 SD93 Firewire deck, internal 1TB raid, 2.5GB RAM.
Please help if you have any ideas.
Thanks.
Aaron

It it maybe a situation where the source uses Drop Frame timecode and the FCP capture setting doesn't? Or vice versa?
I haven't done the math yet to check, but your sample rate numbers could reflect errors resulting from FCP trying to compute NDF sample rates while looking at incoming DF timecode. Or vice versa.
The different sample rates would then be the result of different length clips, which would have more or less DF "adjustments" in them, and so would produce different apparent sample rates if view as NDF clips.
Like I said it's just a guess, but check out to see that your DF or NDF timecode settings are the same for the source as for the capture.
Maybe I'm right. Or maybe I'm talking through my hat. It's happened before...

Similar Messages

Video (or Video + Audio) Data Rate

Wondered if anyone here might know of a software utility that has the ability to scan QuickTime compatible files and either display the video (or video + audio) data rate excursion or, as an alternative, graph the instantaneous variation in the data rate over time.

Apple +I (show info) in QT player will show you the framerate and bitrate while playing.
Thanks for your suggestion, Rick. However, I must point out that the data rate so displayed is simply the total average for the entire clip and not the instantaneous (i.e., constantly changing) variance with time. Preliminary observations tend to indicate the single pass H.264 algorithm begings roughly 10% under the requested data rate and quickly settles very near the target. The multipass algorithm, on the other hand, appears to begin in the vicinity of 300% above the targeed data rate and monotonically decreases throughout the remainder of the movie clip. So much for a quick qualitative analysis based on observations during the actualy encoding process.
What I now wish to do is actually perform a bit of quantitative analysis and, based on those results, correlate actual data rates with the ability of a given H.264 multipass file to sync to an iPod. Basically, I am trying to determine whether or not a user data rate input is used as a comparator during the initial phase of multipass H.264 coding and, if so, how it is implemented. (I.e., how are the delta values handled -- linearly, exponentially, etc.) In addition, I wish to compare QT v7.0.3 with v7.0.4 clips to determine why the latter are less iPod compatible. Too, there remains a question as to whether or not any data rate information is now embedded in the clips themselves since various work arounds tried (i.e., clipping file lengths) have proven unsuccessful.

Audio data rate

Upon completing a capture, a warning window indicates that the audio data rate does not match that of the source tape. A loss of sync could occur. How do I make the correct adjustments? I'm capturing from a Sharp VL-Z3 via firewire. Also, how do I monitor audio IN during capture. I can only hear audio via my camera's speaker. Meters indicate audio coming in on Final Cut and clips play back. Could this be related to the afore mentioned issue?

Welcome to the forum,
I'm not familer with your camera but many cameras allow the recording of 32khz audio as well as the more standard 48khz version. This audio shift happens when you put the camera into LP recording mode.
There is a way to turn on audio monitoring during capture (check in the manual) but it will not change your problem. You can not change the nature of the audio while capturing and, monitoring while capturing does nothing to the audio stream.
If your audio is not being recorded at 48khz but you need it to be in that format to use the Easy Setup Settings for Editing DV, you can export just the audio portion (using quicktime export), open it in Quicktime Pro, conform (transform) it to 48khz - then reimport the resulting file back into FCP.
If this is not your problem, please offer up more information and we'll do what we can.
cheers,
x

224 audio data rate

Hi Folks,
is there a much different between 192 & 224 bit rate in the audio? I'm doing Stereo DVD & had always use 192, but want to try 224 this time.
Any suggestion?
thankx
JP

Hey Jpng
as far as size there is not much of a difference.
i would say try 224 audio, you might like it a bit more.
i spoke with a dolby tech a few months ago, he suggested this bit-rate actaully. when i asked why he just responded that 192 was the lowest, acceptable setting and 224 was the optimal for stereo. kind of like using 384 for 5.1 surround and using 448.
its just that most people know 192 since all dolby documntation says so, but the same doscumentation states for 5.1 surround, 384 is acceptable, which is, but i rarely use.
i say use 224 but make sure all preprocessing for the audio file is OFF!
this is one of those questions that is really hard to answer since no one that i know has sat down with the Dolby hardware analyzers and actually done tests (me included. to add to the difficulty, some dolby techs say 192 is a must, others say 224 is optimal, where most dolby documtation says 192 is fine.
see what i mean! by the way, one thing that all dolby techs agree on is that ANY bit-rate setting greater than 224 is a total WASTE. nothing is gained!
Mikey m.

Analyzing DV Audio - Validating Audio Data

Sony UVW-1600 through ADS Pyro A/VLINK using S-Video in/Firewire 400 out
Audio through Alesis MultiMix 8 USB mixer - 44.1 kHz output only
Just did clean install of Snow Leopard, reinstall/updates on FCP
Otherwise same configuration as before but lost original Custom AV settings and Easy Setups.
Now when I capture BetaSP material, video through the A/VLINK and audio through the Alesis, I'm getting messages I never used to get:
If I set my Capture Preset to 48 kHz audio, at close of capture I get "Analyzing DV Audio - Validating Audio Data"
If I set my Capture Preset to 44.1 kHz (matching the Alesis output), I get a long message saying the audio data rate of my captured media file doesn't match the sample rate on my source file.
From checking old captured clips, I see that I've always been capturing at 48 kHz but never got the "Analyzing DV Audio" message before. Not seeing any sync problems.
Anyone know why I'm suddenly getting this message? I don't want to capture audio through the DV stream as my source tape levels are all over the place. Thanks

Okay, I've answered my own question (I think) but could be helpful to others: in setting up my replacement (after the reinstall) Capture Preset I made a custom preset that selected "USB Audio Codec" for "Quicktime Audio Settings Device." So FCP was seeing video come in through the Firewire stream and audio through the USB stream. It evidently wanted to "validate" the rejoining of the two. I've just changed that Quicktime Audio Settings Device back to DV Audio and now I don't get the validating message. In OS X System Preferences I have "USB Codec" chosen for default audio input so FCP must be using that setting but "thinks" it's getting audio from the Firewire DV stream. I'd tried that earlier but lost audio completely during FCP capture. Now it's working. Odd...

Send audio data over UDP

Hello again,
Sorry for asking a lot of questions, but I'm developing an application in jmf and I don't know so much how it works...
I'm trying to receive rtp data from the network, then uncompress to RAW format (but it has to be coded in some codec like g711, g729 or GSM) and save it into a buffer. For this I'm following the example DataSourceReader. I have attached some things like keep off the rtp header, and save the audio data into a bytebuffer (instead of using the printInfo() method I'm using saveIntoByteBuffer() method).
Another class is sending the data (saved in byteBuffer) over the network (UDP, not RTP-UDP).
And anotherone is receiving this data, and creating a DataSource to play the audio data received.
To do this, I don't know exactly how to do it. I need to create a DataSource like the examples (LiveStream and DataSource from [http://java.sun.com/javase/technologies/desktop/media/jmf/2.1.1/solutions/LiveData.html |http://java.sun.com/javase/technologies/desktop/media/jmf/2.1.1/solutions/LiveData.html] ) or can I do it like JpegImagesToMovie from [http://java.sun.com/javase/technologies/desktop/media/jmf/2.1.1/solutions/JpegImagesToMovie.html|http://java.sun.com/javase/technologies/desktop/media/jmf/2.1.1/solutions/JpegImagesToMovie.html] ?
Sorry if it's an easy question, but I don't understand so much why implementing the 'same' thing (custom DataSource) it's implemented different.
Thanks

Sorry if it's an easy question, but I don't understand so much why implementing the 'same' thing (custom DataSource) it's implemented different.They aren't the same thing, actually. DataSource is the parent class, but there are 2 different kinds of DataSources.
A PushBufferDataSource will "push" the data out when it's available to be read. Whenever it decides it has data ready to be read, it informs whatever is reading from it that data is available.
A PullBufferDataSource will not do that. Whenever it has data ready to be read, it doesn't do anything to inform what's reading from it.
The next obvious question is, why does it matter?
PullBufferDataSource's are good for situations where the data is always present. For instance, if you're playing a file from your hard drive, it's better to just let your Player object fetch data when it's needed. There's no need for a "data available" event, because, the data is always available...
PushBufferDataSources are good for situations where data is being generated / received from an outside source. You can't read from it until the data comes in, so rather than blocking and waiting for the read, it'll tell your reader class when to come back for the data.
Hope that helps!
P.S. For your needs, you'll want to be using a PushBufferDataSource, so the Live example code.

Data rate slows over time: Using Time Capsule (4th) and AirPort Express (1st)

Time Capsule using Airport Express to extend network.
Over 3-4 days my data rate gradually gets slower and slower until I have to reboot both TC and AE.
Fresh reboot gets me 10Mbs or better for a while and then slows over time to 2Mbs or less.
I can cable directly to the ISP modem, taking TC and AE out of the equation, and its smokin fast
No hair left to pull out, so any help or suggestions would be appreciated.
Time Capsule (4th gen) - Model A1409
Version - 7.6.4
Wireless Mode: Create a wireless network
Wireless Security: WPA2
Channel: 149 (Automatic), 11 (Automatic)
Allow this network to be extended is checked
Radio Mode: Automatic
Multicast Rate: Low
Transmit Power: 100%
Use wide channels is checked
Guest Network is NOT enabled
Mac Address Access Control: Not Enabled
Connection Sharing: Share a public address
Using DHCP
One printer attached to Time Capsule
Airport Express (1st gen) - Model A1143
Version 7.6.4
Wireless Mode: Extend a wireless network
Wireless Security: WPA2
Channel: 11 (Automatic)
Mac Address Access Control: Not Enabled
Using DHCP

You have your wireless channels on auto.. it is usually better to control it.
You will notice I have different names for 2.4ghz and 5ghz.. then I can force the link to either band.
After you set Radio Channel selection to Manual,, click edit. You can then set the channels.
Lock the channels.. ie you take control and you set what the wireless channels will be.. when you extend wireless this works far better. The auto channel system has to run on the Extender.. but since the main base is fixed on the 2.4ghz so it will be also.. the only channel that can flap around is 5ghz (extender) since that is not under my control. Of course it is up to you to determine which channel works best.. I strongly recommend the upper channels for 2.4ghz and the lower channels for 5ghz.
So.. I have set 5ghz to 48 on the main base.. and the auto channel sets it to 149 in the extender.
One strange thing results from my setup .. (amongst many!!)
You cannot name the wireless for 5ghz in the extender.. so in my case the upstairs wireless extender has both 2.4ghz and 5ghz named TC24ghz.. nought that I can do about it. But it is not a big deal.
You can see the result in the wireless diagnostics. The last wireless I have put in the box.. it is named TC24ghz but is in fact running on 5ghz. That is just how the over automated system of the apple routers work.. If I could I would fix it.. and maybe shall via serial console at a later date.

Audio sample rate does not match (HDcam to dvcam)

I'm trying to import clips and keep getting this 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."
Footage was originally shot on HDCAM and transferred to DVCAM elsewhere. Using FCP 5, am importing via firewire from a Sony DSR-11 deck. Using DV NTSC 48kHz Anamorphic as capture settings (though I've tried everything that I thought might possibly work with no success). The audio does not seem to drift over the course of several 5 minute or so clips. Clip settings show audio at 48 kHz (don't know if that's from capture settings or from actual data). Seems to me all audio should be 48 kHz 16 bits, so can't figure out what's going on. I have to export an EDL for the project to be finished in HD. Read some similar threads that ended in December, seemingly without much resolution. My broader concern is why this is happening; my immediate concern is do I need to worry about this right now since the media files will need to be recaptured in HD anyway. Any thoughts?
Thanks

A little more info. I'm having this problem on 4 tapes (from different cameras) that were transferred to DVCAM in a squished format to appear full screen on a 4x3 monitor. Video that was letterboxed and I can bring in with the standard DV NTSC capture settings does not have this problem. Still have the problem if I try to import the clips from the squished video with standard settings. Any thoughts?

I have a question about Data Rates.

Hello All.
This is a bit of a noob question I'm sure. I don't think I really understand Data Rates and how it applies to Motion... therefore I'm not even sure what kind of questions to ask. I've been reading up online and thought I would ask some questions here. Thanks to all in advance.
I've never really worried about Data Rates until now. I am creating an Apple Motion piece with about 15 different video clips in it. And 1/2 of them have alpha channels.
What exactly is Data Rate? Is it the rate in which video clip data is read (in bits/second) from the Disc and placed into my screen? In Motion- is the Data Rate for video only? What if the clip has audio? If a HDD is simply a plastic disc with a dye read by "1" laser... how come my computer can pull "2" files off the disc at the same time? Is that what data transfer is all about? Is that were RAM comes into play?
I have crunched my clips as much as I can. They are short clips (10-15seconds each). I've compressed them with the Animation codec to preserve the Alpha channel and sized them proportionally smaller (320x240). This dropped their data rate significantly. I've also taken out any audio that was associated with them.
Is data rate what is slowing my system down?
The data rates are all under 2MBs. Some are as low as 230Kbs. They were MUCH higher. However, my animation still plays VERY slowly.
I'm running a 3GigRam Powerbook Pro 2.33GHz.
I store all my media on a 1TB GRaid Firewire 800 drive. However for portability I'm using a USB 2 smartdisk external drive. I think the speed is 5200rpm.
I'm guessing this all plays into the speed at which motion can function.
If I total my data rate transfer I get somewhere in the vicinity of 11MBs/second. Is that what motion needs for it to play smoothly a 11MBs/second data connection? USB 2.0 is like what 480Mbs/second. So there is no way it's going to play quickly. What if I played it from my hard drive? What is the data rate of my internal HDD?
I guess my overall question is.
#1. Is my thinking correct on all of these topics? Do my bits, bytes and megs make sense. Is my thought process correct?
#2. Barring getting a new machine or buying new hardware. What can I do to speed up this workflow? Working with 15 different video clips is bogging Motion down and becoming frustrating to work with. Even if only 3-4 of the clips are up at a time it bogs things down. Especially if I throw on a glow effect or something.
Any help is greatly appreciated.
-Fraky

Data rate DOES make a difference, but I'd say your real problem has more to do with the fact that you're working on a Powerbook. Motion's real time capabilities derive from the capability of the video card. Not the processor. Some cards do better than others, but laptops are not even recommended for running Motion.
To improve your workflow on a laptop will be limited, but there are a few things that you can try.
Make sure that thumbnails and previews are turned off.
Make sure that you are operating in Draft Mode.
Lower the display resolution to half, or quarter.
Don't expect to be getting real time playback. Treat it more like After Effects.
Compressing your clips into smaller Animations does help because it lowers the data rate, but you're still dealing with the animation codec which is a high data rate codec. Unfortunately, it sounds necessary in your case because you're dealing with alpha channels.
The data rate comes into play with your setup trying to play through your USB drive. USB drives are never recommended for editing or Motion work. Their throughput is not consistent enough for video work. a small FW drive would be better, though your real problem as I said is the Powerbook.
If you must work on the powerbook, then don't expect real-time playback. Instead, build your animation, step through it, and do RAM previews to view sections in real time.
I hope this helps.
Andy

PGC...data rate too high

Hallo,
message
nunew33, "Mpeg not valid error message" #4, 31 Jan 2006 3:29 pm describes a certain error message. The user had problems with an imported MPEG movie.
Now I receive the same message, but the MPEG that is causing the problem is created by Encore DVD itself!?
I am working with the german version, but here is a rough translation of the message:
"PGC 'Weitere Bilder' has an error at 00:36:42:07.
The data rate of this file is too high for DVD. You must replace the file with one of a lower data rate. - PGC Info: Name = Weitere Bilder, Ref = SApgc, Time = 00:36:42:07"
My test project has two menus and a slide show with approx. 25 slides and blending as transition. The menus are ok, I verified that before.
First I thought it was a problem with the audio I use in the slide show. Because I am still in the state of learning how to use the application, I use some test data. The audio tracks are MP3s. I learned already that it is better to convert the MP3s to WAV files with certain properties.
I did that, but still the DVD generation was not successful.
Then I deleted all slides from the slide show but the first. Now the generation worked!? As far as a single slide (an image file) can not have a bitrate per second, and there was no sound any more, and as far as the error message appears AFTER the slide shows are generated, while Encore DVD is importing video and audio just before the burning process, I think that the MPEG that is showing the slide show is the problem.
But this MPEG is created by Encore DVD itself. Can Encore DVD create Data that is not compliant to the DVD specs?
The last two days I had to find out the cause for a "general error". Eventually I found out that image names must not be too long. Now there is something else, and I still have to just waste time for finding solutions for apparent bugs in Encore DVD. Why doesn't the project check find and tell me such problems? Problem is that the errors appear at the end of the generation process, so I always have to wait for - in my case - approx. 30 minutes.
If the project check would have told me before that there are files with file names that are too long, I wouldn't have had to search or this for two days.
Now I get this PGC error (what is PGC by the way?), and still have no clue, cause again the project check didn't mention anything.
Any help would be greatly appreciated.
Regards,
Christian Kirchhoff

Hallo,
thanks, Ruud and Jeff, for your comments.
The images are all scans of ancient paintings. And they are all rather dark. They are not "optimized", meaning they are JPGs right now (RGB), and they are bigger then the resolution for PAL 3:4 would require. I just found out that if I choose "None" as scaling, there is no error, and the generation of the DVD is much, much faster.
A DVD with a slide show containing two slides and a 4 second transition takes about 3 minutes to generate when the scaling is set to something other than "None". Without scaling it takes approx. 14 seconds. The resulting movies size is the same (5,35 MB).
I wonder why the time differs so much. Obviously the images have to be scaled to the target size. But it seems that the images are not scaled only once, that those scaled versions of the source images are cached, and those cached versions are used to generate then blend effect, but for every frame the source images seem to be scaled again.
So I presume that the scaling - unfortunately - has an effect on the resulting movie, too, and thus influences the success of the process of DVD generation.
basic situation:
good image > 4 secs blend > bad image => error
variations:
other blend times don't cause an error:
good image > 2 secs blend > bad image => success
good image > 8 secs blend > bad image => success
other transitions cause an error, too:
good image > 4 secs fade to black > bad image => error
good image > 4 secs page turn > bad image => error
changing the image order prevents the error:
bad image > 4 secs blend > good image => success
changing the format of the bad image to TIFF doesn't prevent the error.
changing colors/brightness of the bad image: a drastic change prevents the error. I adjusted the histogram and made everything much lighter.
Just a gamma correction with values between 1.2 and 2.0 didn't help.
changing the image size prevents the error. I decreased the size. The resulting image was still bigger than the monitor area, thus it still had to be scaled a bit by Encore DVD, but with this smaller version the error didn't occur. The original image is approx. 2000 px x 1400 px. Decreasing the size by 50% helped. Less scaling (I tried 90%, 80%, 70% and 60%, too) didn't help.
using a slightly blurred version (gaussian blur, 2 px, in Photoshop CS) of the bad image prevents the error.
My guess is that the error depends on rather subtle image properties. The blur doesn't change the images average brightness, the balance of colors or the size of the image, but still the error was gone afterwards.
The problem is that I will work with slide shows that contain more images than two. It would be too time consuming to try to generate the DVD over and over again, look at which slide an error occurs, change that slide, and then generate again. Even the testing I am doing right now already "ate" a couple of days of my working time.
Only thing I can do is to use a two image slide show and test image couple after image couple. If n is the number of images, I will spend (n - 1) times 3 minutes (which is the average time to create a two slides slide how with a blend). But of course I will try to prepare the images and make them as big as the monitor resolution, so Encore DVD doesn't have to scale the images any more. That'll make the whole generation process much shorter.
If I use JPGs or TIFFs, the pixel aspect ratio is not preserved when the image is imported. I scaled one of the images in Photoshop, using a modified menu file that was installed with Encore DVD, because it already has the correct size for PAL, the pixel aspect ratio and the guides for the save areas. I saved the image as TIFF and as PSD and imported both into Encore DVD. The TIFF is rendered with a 1:1 pixel aspect ratio and NOT with the D1/DV PAL aspect ration that is stored in the TIFF. Thus the image gets narrowed and isn't displayed the way I wanted it any more. Only the PSD looks correct. But I think I saw this already in another thread...
I cannot really understand why the MPEG encoding engine would produce bit rates that are illegal and that are not accepted afterwards, when Encore DVD is putting together all the stuff. Why is the MPEG encoding engine itself not throwing an error during the encoding process? This would save the developer so much time. Instead they have to wait until the end, thinking everything went right, and find out then that there was a problem.
Still, if sometime somebody finds out more about the whole matter I would be glad about further explanations.
Best regards,
Christian

Audio data to be acquired at 192khz

hi ...
well i have made a VI to acquire audio data and then write it to a file to furthur process it. the problem is that when i send stram of audio data upto 48 khz the data is captured continuously, but if i try to send a stream of 64khz or 192 khz the data is acquired but for a little time. i want to acquire data of 192khz stream continuously. i m using labview 6.2 and dio32hs card.
i want to acquire data at 192khz for atleast 6 min continuously.
is it possible???
please help..
thanx and regards
vivek modgil

Look at the producer consumer template for multithreading.
Essentially what happens if you use a simple model like
Initialize()
DaqLoop
   Read()
   Process()
   Display()
   Save()
LOOP
is that if the saving and/or sace takes too long the next itteration of read will have caused a buffer over run in the daq task. This is more evident in the faster Acquisition rates. Making the buffer larger only delays the onset of this event.
A better model is (Threads are just parallel loops)
Thread 1 (Producer):{
Initialize DAQ()
DaqLoop
   Read DAQ data() //all data avaliable.
   send data to a Queue
}Loop(until done)
Kill queue();
Thread 2 (Consumer):
DataLoop
Wait for data at queue; // put in some timeout ie 100ms
If new data-> Process() Display(); Save();
}Loop(While queue is valid)
This is just a starting point. Look at the producer/consumer model.
This ensures that the DAQ loop will (should) never overrun and the consumer loop will attempt to keep up with the data. Saving data to file is slow so do this post acquisition if possible (IE build array in memory).
Paul
Paul Falkenstein
Coleman Technologies Inc.
CLA, CPI, AIA-Vision
Labview 4.0- 2013, RT, Vision, FPGA

Best Data rate and format for Big Screen Playback

I am a video producer supplying content for a tradeshow display consisting of 18 flat screens oriented vertically, 8 over 8. Very heavy duty Windows playback machine, with 3 graphics cards, 1 for each set of 6 monitors. The total pixel size is 3240x 1280 px. We have produced a number of full screen and smaller movies that the interactive programmer is trying to intergrate into a Flash program where some videos will play full screen, and others in smaller windows (those were produced in 1920x1080). All videos were mastered at the specified pixel count, square pixels, progressive, and output in Pro Res 422, 44.1K audio. Flash is being used to create the interface, and I am transcoding to .F4V at lower bit rates like 4, 6, and 9 Mbps. They are having trouble playing these movies back smoothly. We have done similar projects in the past with Director without issue. Of course they are blaming the video, I am offering to encode in any format at any data rate they suggest. Can anyone suggest a direction here?

Are you trying to burn a standard DVD or are you trying to put a QuickTime .mov file on DVD media?
A standard DVD doesn't worry about file size (only the duration).
A "data" DVD is limited to the type of media used. A single layer DVD is about 3.7 GB's.
In order to make a data DVD you need to keep the data rate low enough to not exceed the DVD media playback abilities. They can't handle the higher rates found in many of the preset options.
You can avoid all of these headaches by telling the viewer to "copy" the .mov file from the DVD to their Desktop. Then nearly any of the presets option will work.
H.264 is a great video codec and the "automatic" preset should work just fine. Use "multi-pass" for best quality (takes a very long time). Remove the check mark for "Audio" since your file has none. Leave it checked and you waste file size because a silent audio track would be added.

Encoded video does not have expected data rate

Hi
I've been encoding with Flash Media Encoder (CS3) recently
and come across a contradiction which I can't seem to make sense
of. Can it be possible that the longer the video, the harder it is
for the encoder to maintain data rate?
This was revealed when I started using FMS2 (space on a
shared server as opposed to my own server). I encoded a video at
what I thought was a data rate which was within my data rate limit
for my stream and it stuttered on playback over the stream. The FMS
supplier informed me that the FLV I thought I'd encoded at 400kbps
video 96 kbps audio actually had a data rate of around 700kbps in
total. But if I open the video in a FLV player, and look at info,
it still shows the setting I originally intended. However, my
supplier suggested checking data rate using this equation:
(filesize in MB x 8)/videoduration in seconds = video
datarate in kbits/sec
If you apply this calculation to my video it does indeed show
what they said:
Using Flash Media Encoder:
Encoded at 400kbps video - 96 kbps audio
Compressed filesize = 26,517 kbytes
Duration = 240 seconds
CALCULATION GIVES DATE RATE OF 880k/b a second
(note file contains an alpha channel)
Furthermore, applied to a shorter video, we get the following
more logical results:
Encoded at 400kbps video - 96 kbps audio
Compressed filesize = 6,076 kbytes
Duration = 90 seconds
CALCULATION GIVES DATA RATE OF 527 kb/sec
(no alpha channel)
The only differences between the two vids is the alpha
channel and the duration. Somethings doesn't add up here.
Any clues?
Thanks
Simon

I've just partly answered my own question. Funny how writing
a post can help you think your problem through sometimes.
The issue is the alpha channel, which adds to the file size
thereby making the data rate "incorrect". In other words, the same
file encoded without the alpha channel provides the expected
results. I guess this changes my question to: How do I create
predicatable data rates when encoding using an alpha channel. I
could just estimate of course and see what comes out, but is there
a better way?
SImon.

Acquire/manipulate audio data from internet stream

I want to acquire the audio data from an audio stream (i.e. pandora, youtube, etc.) from the internet. If I can get this data and be able to play it/graph it/etc in Labview, I will be pleased. However, I'm not sure if this is even possible not knowing the details behind how packets are built and transmitted over the internet.
The closest I've come is using the acquire_sound.vi and taking the data from the sound card; albeit, it's very noisy.
For fun though, I'd like to do something directly from the audio stream itself (perhaps building a dll?) to gain a better understanding and experience.
Any advice is greatly appreciated!

Dnor -
As far as I am aware there is a specific data time for audio streaming online, and it's something specific you would likely need to know on a low level to communicate/copy to. Although it wouldn't be real-time, you can run a Windows audio recording program like TotalRecorder which should save and convert the audio output to .wav. Then run a LabVIEW vi to read the wave file, which you could graph. There is a KnowledgeBase on setting up the read from .wav file here if you'd like to go that route: http://digital.ni.com/public.nsf/allkb/1BE60DBBFA22ACC186256DFB000B8464.
Regards,
Ben N.
Applications Engineering
ni.com/support

Multiclip Dropped Frames/Data Rates from from FW 800, 7200rpm Drive

I am using an external Lacie 7200 rpm "Little Big Disk" .. Editing DV Multiclips in Final Cut (3.6 MB/sec data rate per clip... 7 clips/streams) But it won't play back all at once, anything over 15 MB/sec data transfer rate (as I watch Activity Monitor) it starts choking and dropping frames til it just freezes. Lacie claims that this drive handles a steady 80MB/sec transfer rate... All I need to do all these streams is 26 MB/sec (That's the speed it does at work from an XSAN and no problem playing back 7 streams of DV) Is the bog down in my system the FW800, my RAM, processor, or the Cache of the drive? (16MB)

Do you have any other FireWire devices (incl. cameras) plugged into other ports on the computer or on the drive itself? There's only one FireWire bus on a G5, and it's shared between the FW800 port and the FW400 port on the back in addition to the FW400 port on the front.
If you have other devices plugged in, they can cause the entire bus to slow down to the speed of the slowest connected device.

Audio Data Rate digitizing DVCPRO50 over firewire

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