Nice to see that it installs pretty good with Vista drivers, however, when changing the audio bit sample buffers (@ 44.1K) from 256 samples to 300 samples, it caused a BSOD and rebooted the computer. Mackie - 400F Program=Ableton Live 7
I am working on a packet switched network. I want to capture the audio data from mic into a buffer at a desired sample rate.How can I capture the audio data from mic into a buffer instead of a file? Also how can I control the audio capture rate as per the desired sample rate?
I'm working on an Adobe Air application written in Flex 4 that plays .mp3 audio files on the user's computer. Note: these are are not audio files shipped with the application -- they are .mp3's on the user's computer that they select for playback through the application. The application works fine for .mp3s encoded at 44.1 kHz, but can give unpredictable results if other sample rates are used. I've done plenty of research to know the limitations of the Sound class and how .mp3 will basically be my only option in Flex. My question is: Is there a way to detect the sample rate of the .mp3 audio in Flex 4 ActionScript? Rather than worry about making the application work well with non-standard sample rates, at this point I'd like to just catch those cases and prevent files with non-44.1 kHz sample rates from loading. To be specific: if a user selects an .mp3 for playback that has been encoded at 48 kHz, for example, I'd like to be able to detect that case and take action preventing the file from loading and then announce to the user that this is not a supported audio file.
My audio analysis application needs a microphone sample rate of greater than 8000 Hz.At least 11025 Hz is required.It seems that some phones support a higher rate and some are limited to 8000 Hz.Does anyone know what percentage of Android phones will support the higher sample rate?And if I do develop this app, how can a potential customer determine whether or not his particular phone supports the higher sampling rate?
I know that there are some specifications as to the format the video must be in and the specifications. But as far as I am concerned my video meets those standards. below are the details of my video file. Why this mpeg4 video can't be uploaded to my ipad? Video: Length: 00:46:20 Frame width: 1280 Frame height: 720 Data rate: 1099kbps Total bitrate: 1228kbps Frame rate: 23 frames/second audio: Bit rate: 128kbps Channels 2(stereo) Audio sample rate: 48 k Hz Info: iPad (3rd gen) Wi-Fi, iOS 5.1.1
I have Nokia X6 (S60V5) phone and I am trying to play wmv videos on it with no luck.Its Device details says that it can play wmv9 Video Playback Formats 3GPP formats (H.263), Flash Video, H.264/AVC, MPEG-4, RealVideo 7,8, WMV 9 I did encode a sample video with following config. Resolution: 480x360 Video Bit rate : 1024K Video Codec: wmv3 (which is WMV9) Audio codec: wmav3 Sampling Rate: 44100 Audio bitrate: 128K This configuration is not supprted on X6 Could anyone tell me what should be the configuration of wmv file inorder to be played on Nokia X6?
i'm trying to convert mpg videos to flv videos. i'm using adobe cs3 flv converter as well as quicktime pro. both yield a flv with no audio. i open the video in quicktime pro. it plays with sound. then i go up to FILE > EXPORT > MOVIE TO FLASH VIDEO (FLV) click on OPTIONS then click on AUDIO. it's all grey. no audio options. i am aware of this problem/solution regarding flv's stopping short but i'm not even get audio options. "It has everything to do with sound sample rate! You need to export your movies that are currently using 22.050 kHz and to a new quicktime movie with a 44.100 kHz sample rate. THEN export as a .flv with whatever frame rate and .mp3 compression rate you need. This should fix the problem!"
I noticed that the macbook unibody does not have firewire port. So if I need to use firewire device, then I have to get the USB to firewire adapter for it. I just want to get this right, if I use the adapter then that means I will get tranfer rate of general USB 2.0 speed which is 480Mb/s. Is that right? If it is, then I'm gonna miss the higher transfer rate of firewire on old macbooks
I just recently bought myself a NewerTech miniStack v3 500 GB external hard drive.It has a quad interface, which can connect to USB 2.0, FireWire 400, FireWire 800 and eSATA. I currently have it connected to FireWire 800. I also have an audio interface that's connected to FireWire 400. I've been told that the FireWire 400 & 800 use the same internal bus, and therefore using the audio interface and external hard drive at the same time wouldn't be recommended. Even if it shares the same internal bus, would I be OK to use them at the same time? Anyhow, I'm thinking about changing the connection of my hard drive to eSATA. I'd like to know how I can go about setting this up. I mainly work with audio, but the occasional video job does arise every now and then. Information: Dual-core 2GHz PowerPC G5 Mac OS X (10.4.11) 5 GB 533 DDR2 SDRAM
I'm having a problem with a Firewire Audio card when recording from a MIDI input. It works okay for a few seconds, then i get the dreaded BSOD - I haven't had time to look through the dump file yet, but the thread ks.sys is mentioned. I've had a look about, and I think the problem is something to do with the ASIO4All driver I installed (and subsequently uninstalled) OR how the IRQs are set up in windows.
Hello all, I have been using popcorn to convert ts_video files to MP4's when finished the sound quality is horrible, I have tried numerous different settings and a lot of wasted time. The video quality is alright be the sound is horrible. Everything else on the phone sounds clear (music, purchased movies, ringtones) Any suggestions for settings or different software. Audio Format:AAC-LC Data Rate:320kbps Channels:Stereo Sample Rate:48kHz Sampling Quality:Best
I am working on a VOIP based application. I need to capture the audio from mic at 8000 hz sampling rate, get the 20msec packets, encode and send the packets over the network. Can somebody tell me how to configure the mic for the desired sampling rate?
Hello,I have got a file with this specification:Code:MPEG 2.5 Layer III128 kbp/s (CBR)8000 Hz, Mono4:27When using the following formula...Code:Frame Size = (144 * (BitRate * 1000) / Sampling Rate) + PaddingFrames = Audio Size / Frame SizeDuration = Frames / 27.8...the duration is 1:12 (1/4 of the real length).The real number of frames is 3973 and my program returns 1986. Therefore, something is wrong withCode:Frame Size = (144 * (BitRate * 1000) / Sampling Rate) + PaddingThanks in advance!
I have a PNY 570 gtx videocard hooked up with a HDMI to monitor. No matter if I connect it to a audio receiver or to the monitor I only get a maximum of 2 channels and Bit Depths 16-bit.Sample rates are only shown as 44.1kHz and 48.0kHz and the HDMI jack only shows L and R. It wont let me test a surround sound speaker system.The stats to this video cards sound will allow much higher samples and bits.All this information was gathered in the Windows 7 sound properties.I click on advanced in the sound properties and still cannot change to a higher quality that 48000 Hz.
I'm experimenting with Android's audio recording and playback. Is there a way to enumerate the available audio parameters on my device? Right now, when I pass a combination of parameters that the hardware (or emulator) doesn't like, I just get an error. So I am having to "guess":code... Surely there's a better way! This chart indicates that the only supported audio input sampling rate is 8 kHz? Is that correct?
Using a firewire device and watching some online video training via my firewire device. When I go to launch the AUDIO application that I am trying to gain expertise in, it says the audio driver is in use (channels 1 and 2) so am trying to figure out how to RELEASE the driver into the background so I can play both the training video (Lynda.com via quicktime) and the audio application. (REASON 4).
Still working on getting audio exported to an AAF, and having problem doing this. So, how FCPX is handling my audio. When it's on the timeline, is it being conformed to the frame rate of the video? When audio is exported, does FCPX force it to conform to the video frame rate in the project settings, audio properties, or both? Or does it allow the audio to export at whatever rate the original media is at? I export an XML from FCPX, then open that with X2Pro and it makes an AAF file, but a report tells me that "Conform rates are not supported. The Conform rate at 01:04:49.23 has been ignored." So when I go to that timecode, all I see is video at 24p and a detached audio piece at 23.98. Why are these different? Info: Final Cut Pro X, OS X Mavericks (10.9.4), 3.5Ghz intel Quad-core i7, 16 GB RM
I've started to see some issues with using firewire 400 devices on my 800 compatible imac. Issues being not seeing any signal from my dazzle analog to dv video bridge, and also not getting an output signal from my Alesis Multimix 8 track firewire audio interface. Is there some sort of problem with my adapter? or is this a prevelent dilhemma in the conversion from 400 to 800 firewire cables? Information: eMac 80 GB hard drive, 160 GB firewire hard drive, 512 MB RAM
I want to capture audio samples from the microphone in my adobe AIR application and then save them to an flv file. I have the following code: mic.setSilenceLevel(0, DELAY_LENGTH); mic.codec = SoundCodec.SPEEX; mic.encodeQuality = 6; [Code].... The problem is that I suspect that in my handler I am only getting raw samples and not compressed samples. The reason for my suspicion is that the number of bytes I get per message is equal to 20 ms (which my definition is 1 speex frame) of raw audio and not compressed audio. Also the number of bytes doesnt change if I change the encodeQuality. Reading the documentation suggests that adobe will only compress the audio before transmission to a flash media server or another peer. Is there a way to publish and read the stream locally in order to get compressed samples. ? Or any other way to get the compressed samples?
I'm running Pro Tools M-powered 7.3 on a Macbook Pro I got in January (which is apparently a "late 2008") with an M-Audio Firewire 1814 interface. Currently I have a card slot thingy with firewire 400 and USB 2 that I use to connect the Firewire 1814. I would like to get a fast external drive for audio recording and editing (and possibly video) and have heard ESATA is the fastest. I've seen the ESATA cards and thought I might get that and then get a firewire 800 to 400 adapter cord to use the Firewire 1814. BUT I've read Digidesign "doesn't support" ESATA drives, it recommends firewire drives. However, in my experience, Digidesign not officially supporting something doesn't mean it doesn't work. So with all that being said, can anyone: A.) Recommend the best/fastest external drive for my purposes with my setup? and/or B.) Can anyone out there verify they've used an ESATA drive with a similar or same Pro Tools setup with some success on a Macbook Pro? Information: Late 2008 Mac OS X (10.5.7)
I'm having a problem when I connect or disconnect my firewire cable I get a BSOD. I'm not sure what to do. I think I have the latest drivers for firewire.
I have a lot of video I'm in the process of indexing and I'd like to gather information about the videos for queries and stuff. When I right-click a video and select properties, I can see a list of values under the summary tab: Audio Bit Rate, Sample Rates, Title, Comments, Duration, etc. [Code]...
I'd like to start creating some 3GP video files for my X02HT/S630, but what format have you guys find works best?I am referring to: -Video codec -Video Size -Frame/sec -Video Bitrate -Audio Bitrate -Audio Codec (AMR NB or AAC) -Sample Rate -Mono or Stereo
I want to play youtube videos and programmatically direct firefox's audio out to a particular sound card. Playing raw data 'stdin' : Unsigned 8 bit, Rate 8000 Hz, Mono aplay: set_params:1059: Sample format non available Available formats:[code].... The command-line suggests that it worked, but audio came out of hw:0,0 (the default).Do I need to make my own plugin? Or do I need to force alsa to take 44100Hz?I refuse to use pulse since the memory leak bug makes it crash often.
Is there any way to get Audio Midi Setup to output the 88KHz sampling rate to analog outs? I'm sure the capability is there in Core Audio. Got a 2.8GHz Core 2 Duo iMac, BTW.
Hi fellow VBers...I need to be able to open an MP3 file and decipher the audio within it to create a waveform display based on the content.I have already established how to do it with raw PCM, but I know MP3s have other stuff in them, which means it cannot be interpreted in the same way.Basically, I need to filter through the headers and tags etc, to get at the audio frames and analyze their amplitude vs. sample rate etc.If anyone can help, I’d very much appreciate it.Thanks in advancePhil
Does anyone know how to convert videos for the BOLD ? I have an 8Gb card.I am using moviavi which converts from avi to .mp4.I am using these settings below ; MPEG4 Width 480 Height 320 25fps 698kbps Profile Simple Quality Medium Audio AAC 192kbps Profile LC Sample Rate 48,000 From here, I use Desktop Manager > Media > Copy Down and still my audio is way off the video..
import audio into Flash without something going awry?Ok, I've dealt with problems in Flash before concerning audio. I figured out that it prefers WAV over MP3 for one. I thought I finally had it down.But after taking a break from Flash I seem to have forgotten what kind of WAV file it prefers.I'm creating sounds using Soundbooth CS5. But I can't seem to get Flash to import the sounds. So, I've been playing about with the sample rate and such but to no effect.I'm using PC, Flash CS5, also Soundbooth CS5
I am running an optical cable to a Channel Islands outboard DAC that can handle 24/96. I downloaded some 24/96 albums from HDTracks and convertd them using XLD to 24/96 Apple Lossless in iTunes. When I go into my Audio/ MIDI program and try to change the output on the Mac (13" Macbook Pro from 2010) to 24/96, the computer won't accept it. It just bounces the setting back to 24/44. Why won't it accept a 96 kHz sampling rate?
Hello!I have attached a Visual Basic project, which is the pre-alpha version of VBSpot (http://vbspot.sourceforge.net). I was wondering if any of you might give me some tips on how to make some things better (making the code faster, make it use less variables...).Feedback would be greatly appreciated!Well, to get an idea of what the program should do:VBSpot is a tool which returns various information about audio file. This pre-alpha version supports only MPEG Audio (MP1, MP2 and MP3) and only returns basic values, such as bit rate, sampling rate, channel mode, flags... It also offers support for Xing/Info and VBRI headers, which are used to "mark" a VBR encoded file. Finally, this version also detects APE, ID3 and Lyrics3 tags. Beside that, it also reads out the ID3v1(.1) fields.The final version should support at least MPEG Audio, MusePack and Ogg Vorbis, and also display information about SCFSI, scale factor scale, bit reservoir, block type, granule type, bit rate distribution...For more information about VBSpot, please visit the project's site.Sebastian MaresPS: Source code licenced under GPL!
I want to record a sample from the microphone, then I want to play the recorded file maximizing the volume and apply some audio effect, like modifing sample rate or addind an echo.Waht is the best way? is there an example?
how audio is output from the iPad. This is also the case for the iPhone and iPods.When you play music on these devices they are digital for us to hear them they have to be converted to sound waves which are analogue. When we plug a set of headphones into the headphone socket you can here the music and adjust the volume because the iPad has a DAC ( Digital to Analogue Convertor) and an amplifier. Because of limits in space and price the in built DAC and amp are not the best, they are very good and for most of use they are more than adequate to listen to music through. The other way to get music out of an iPad is through the 30 pin dock connector. The advantage with this that the music steeam is digital, no DAC or amp involved. We the listeners can then decide how we want to process the music so we can hear it. We still need to use a DAC and amp but we can decide the quality and price we are willing to pay.So what is the best way to connect an iPad to an audio system/We can use the headphone socket on the iPad and with the appropriate cable connect it to our audio system. The sound will be very good and acceptable for most people. The quality of the music is limited by the iPad DAC and amp as well as the audio system. This is also the cheapest method We can use a dock connector which has an audio output (the apple dock connector does this) we can then connect this to our audio system. The quality of the music is now limited by the dock connector and the audio system. This is more expensive and is only limited by how deep your pockets are. [URL] One other thing to remember is the quality of the orginal music, what rate it was origionaly sampled at, the higher the better and has it been compressed (loss in quality) to make the music file smaller. High sample rates and no compression mean best quailty but bigger music files. Low sample rates and lots of compression, low quality and lower music files.
I have a database in ms access 03, the table name is ResourceMaster, where it contains ID(autonumber), Items name (Text datatype) and Rate (Number datatype), and these are the sample rates which i stored in Rate column (99.99, 45.09, 45.01, 10.03, 10.05) When i bind this table to my DGV the above mentioned numbers are getting changed like this (100.00, 45.00, 45.00, 10.00, 10.00) in DGV. I think the numbers getting rounding off but i dont want to round off, whatever in the table it should not changed in DGV.
Has anyone come across any Firewire memory keys? (Similar to the USB Keys we have now days). Firewire has a much higher transfer rate than USB 2.0, so I was wondering if any company made the firewire memory keys. I did a quick google search but didn't come up with anything recent. All the articles I read were from years ago.
how I will use both my Firepod (which is a firewire audio interface) and a Firewire external hard drive when my Mac Book Pro only has one Firewire input. Information: Mac Book Pro Mac OS X (10.5.7)
I have audio interfaces and Hard drives that have Firewire 400 ports that require power from the computer. My new imac only has firewire 800. Does a firewire 800 to 400 cable or adapter transfer power from the computer to power a device?? I can't find and answer anywhere, and my buddy just bought a mac mini and it does not.. Information: iPhone 4 iOS 4
connecting an Apogee Duet digital to analog audio interface device to the open and available firewire 800 port on the rear of my G-tech 2 TB external HD that is now connected to the iMac's Firewire 800 port. I believe that there are no issues with daisy chaining devices using Firewire 800 connections. Information: 27" iMac 16GB Mac OS X (10.6.4)
I've written a "C" program which transmits audio to a number of computers over a TCP LAN connection. I'm using ALSA, the preemptive kernel, and pthread. After running for 30 minutes or so the slight variation in sampling rates (~+-.01%) among the computers accumulates and manifests as a noticeable differential delay in the sound from the speakers. I know how to detect the variation and would like to dynamically compensate for it by individually varying the sampling rate (ever so slightly) of each playback device to oppose the variation. Does anybody out there in Linux Land know how to dynamically vary the playback sample rate? I've tried using snd_pcm_hw_params_set_rate() and snd_pcm_hw_params_set_rate() followed by snd_pcm_hw_params() to no avail. They don't seem to work when playback is running.
What is your guy's sampling rate on setcpu? It comes standard as 46875.. This is what it means if you don't know what it means.Sampling Rate is defined in microseconds. The lower this value, the more responsive the CPU will be in scaling its speed up or down. However, a lower sampling rate will also negatively impact overall performance.
I keep getting "An internal application error occurred" when I put in my video. (MPG) Running Windows XP The Encoder at work does not work but the Encoder at my house does. 2011-01-06 11:44:37 : ENCODING FAILED- Source file: M:UIVI_Video20110104145741.mpg- Output file: M:UIVI_Video20110104145741.flv- Video codec: On2 VP6- Alpha channel encoded: no- Deinterlace: no- Frame rate: 0 fps- Key frame interval: 0 frames- Video data rate: 400 kbps- Width: 0 pixels- Height: 0 pixels- Audio codec: MPEG Layer III (MP3)- Audio data rate: 96 kbps (stereo)- FLV duration: 00:00:00- Encoding time: 00:00:00
.flac file with 16k sample rate or .wav file with 16k sample rate and then convert it to .flac? I tried to use MCI to record it in .wav with 16k sample rate but i can't convert it properly to .flac with flac fronted command line because something is wrong.
I need to load an audio resource and play it, but I also need to be able to modify some parameters (like the volume, or the playback rate) while the audio is being played. For example, I might want to play a 10 seconds audio stream, and change the volume only after 3 seconds. Is there a way to do it? I've been experimenting a little with AudioTrack without results.
I have to deliver content to Comcast On Demand here in the Atlanta area and their requirements for video are as follows: Video settings: MPEG2 Transport Stream 4:2:0 /---/Bit Rate:3:18 / Resolution 528 x480 / Aspect Ratio 4:3 .Â Interlaced upper field Audio settings: Mpg1 Layer 1 audio at 192kbs/ Coding Mode Stereo (2/0) Layer 2 Dolby AC-3 / ES Bit Rate I haven't found a way to get a MPG2 Transport Stream file with AC3 audio. Is this even possible in Compressor?
I need to transcribe audio files and need an application that will play audio files at double their rate (and half their rate) so that I can quickly go through audio files and then concentrate on the pertinent portions. Any applications for Mac OS X? Free is always good!
How can I check the quality and bit-rate of audio & video files in C# asp.net?
I have just finished the audio on a feature I'm working on in Logic 7.2.3. After I bounce it and put the file back in the sequence, however, the waveform of the new file does not match/if off sync with the waveform of the original material on there. I am totally confused as to why this is happening. Opening up the Audio window, I see the sample rate of the original files are 44100. The bounced sample rate is the same as well, but it's still off. Information: Macintosh Dual 2 Gig G5 Mac OS X (10.4.7)
I have 2 RTM copies of Windows 7 I'm testing both are 64bit and both normally have ATI cards installed (one is a 4650 the other a 4870). The problem is I get a BSOD doing the following: 1. Change resolution 2. Change Refresh rate 3. Change Primary/secondary monitors 4. Load a game that changes the resolution The following do not: 1. Setting Over-scan 2. Setting 3d settings 3. Under clocking GPU [Anything else on the system including playing games that do not change the resolution]. I have about 20 hours into setting up this system and have already wasted an activation on it. I need a way to adjust the resolution without a BSOD occurring every time (note changes are lost). I'm looking for some kind of solution to this issue.
My application cannot handle the JNIE overhead, as I need to make 2 JNIE calls to the native C/C++ code for every 20msec. So I am trying to use the native media libraries and the realted C/C++ files provided by Android in the folder "//external/srec/audio/test" of the donut build. Ex. AudioHardwareRecord, AudioInRecord etc.. I am implementing all the calls in the native layer. I am using the sampling rate as 8000 and want to read 160 samples (320 bytes) at a time. I used the following values to create the AudioRecord.. CODE:.......... It says "success" after creating this interface but I am getting the error "Bad Value" if I am doing the InitCheck. And it is crashing once I start reading the samples or if I try to retrieve the configured sampling rate. Has anybody tried to access this native code?
Usually when you select a track in the arrange window, then click sample editor, that audio file would then appear in the sample editor. But say for eg I have edited the 'kick drum' in the sample editor but would then like to edit the 'snare', when I click the snare track, it doesn't appear in the sample editor? instead, the kick drum remains in the sample editor and I can't seem to change what file is in there. Information: mac book pro Mac OS X (10.5.8)
hi asad,i have some dss & vox audio files. Firstly, i tried to get the duration of dss and vox audio file to divide the file size by the standard average data rate of the audio, but there are different data rate for different files. So it never works. The second way i use to get the metadata i.e. custom average data rate of each file and then divide the file size, but in this case there are no method in vb to find out the file internal information like, bit rate, average data rate etc. of a file while you can get file creation, modified date etc. easily.Now i m in trouble, so plz help me.
I've been doing some searching and can't find any threads regarding capturing audio (cassette tape) to a mac. Wanted to rip some old school tunes from some tapes and convert to MP3. I connected an old tape deck to my G5 via ADS Pyro A/V Link (RCA audio input to Pyro, firewire connection out to G5), that doesn't seem to work. I've also tried connecting to my iMic USB, but that's not picking up any audio either. I've tried using QT audio recording, iMovie, FCP, and Analogue Ripper with no luck. They recognize the Pyro and iMic, but not picking up audio. Information: G5 Dual 1.8Ghz (Q37), 4.5GB ram, Pioneer 112D burner (internal) Mac OS X (10.4.10) 320GB + 500GB SATA HDDs, Radeon 9600 Pro Mac 256MB, LaCie firewire burner (ext)
I am herewith enclosing the sample data, which is one row, containing the branch name, branch code, itemcode, rate, qty, amount . Each branch has several items with different quantity of various rates, which has come in the horizontal line. Now I want that in one by one, for converting that data into oracle. BR.NAME BR.CODEITEM CODEITEM RATE QTYAMOUNTITEM CODEITEM RATE QTYAMOUNTITEM CODEITEM RATE QTYAMOUNTITEM CODEITEM RATE QTYAMOUNTITEM CODEITEM RATE QTYAMOUNTxx1104.5100450153250750263.25158513.5394.5180810506.55003250result should be like thisxx1104.5100450xx2153250750xx3263.25158513.5xx4394.5180810xx5506.55003250
[BSOD] Get BSOD sometimes when restoring from hibernation...
Anyway i'm having real issues with audio interfaces, they never show up in any of the usual places (audio/midi setup, sound, or preferences for audio programs) I've tried both firewire (Mackie Onyx Satellit, and Phonic Helix 18) and usb (Tascam US-122) interfaces and still nothing. Ive tried upgrading to snow leopard and updating everything. Search high and low for drivers, but they all say they are coreaudio compatible so don't provide any. Information: Macbook 2ghz Mac OS X (10.6.4)
Back in Windows XP, I was able to increase a PS/2 mouse's sampling rate from its default setting to 200 Hz at `Device Manager > Mouse > 'Name of Mouse' > Properties > Advanced Settings > Sample Rate` (like this [link]), but this setting is now gone in Windows 7. Is there another way to increase or change a PS/2 mouse's polling/refresh/sample rate in 64-bit Windows 7?The mouse uses generic installed-by-Windows default drivers back in XP, and the same with Windows 7.Edit: The screenshot is not from my PC. I just got it somewhere in the internet. It is just to illustrate how to edit a PS/2 mouse's sample rate back in Windows XP.
it seems I've gotten my first BSoD in years, in boot camp, Vista x64. I was watching a movie, and exited full-screen, and for some reason I wanted to resize VLC's window (the one that has the movie playing) - while the movie was playing. The audio froze and then bam! BSoD. It was kinda quick, I didn't get to see which .dll caused it. Could it be an audio issue since the audio froze? Or a video issue since that's where it all started from?
I exported an audio and video file from FCP, but in the export settings I changed the sample rate from 44.1 - which it was in FCP and exported it as a 48KHZ file. After exporting I tried the file in FCP and in quicktime and it sounds fine. As soon as I import into DVD studio pro it starts making this clicking sound through the audio, every few seconds. Its almost like it is clipping off a few frames of audio on playback. But I am still puzzled why it sounds different in DVD SP to anywhere else I have tested it. I have also tried exporting an AC3 file but it does exactly the same thing.... Information: Dual 2.0 G5 Mac OS X (10.4.11)
I taped at 12bit audio and now while capturing to FCP after every clip I capture I get "Audio sample rated does not match source tape" I looked under my choices for Sequence presets and all I find is 16 bit. How can I switch this so I am supposedly synced and avoid the warning dialog box? My present settings are DV NTSC 48 kHz. I use a sony vx2000 with mini DV tapes. Of course next time I tape I will use 16bit audio to avoid this if I can't make the change.Information: Mac Pro 2.66 Quad Mac OS X (10.6.4)
I've set up a live in-place auditorium video recording system, which right now has the following paths: Mixer --> Beachtek adapter --> remote HDV camera --> Mac So, the audio is good quality, sent by balanced cables to the camera, and the complete audio/video is sent to the computer by firewire. The camera never actually uses a tape. This works pretty well, but relies on the camera's ADC. I can hear a little difference between the mixer sound and the sound coming back by firewire. Ideally, I could send the audio into the Mac's optical port instead. But, DV/HDV have a latency of about .8 seconds, making the audio and video out of sync. Information: MacBook Pro 2.2 Ghz Mac OS X (10.6.4)
I'm working on fixing a Toshiba Satellite laptop. It is a very years old and I reformatted it and tried Windows XP and Windows 7. I installed the Realtek audio drivers straight from the website (HD Audio Driver) but no luck. The Windows Update driver gives me a RTKHDAudio BSOD and I tried fixing that with some advice online to no avail.
I have 1TB Seagate Freeagent external (15mb buffer / 7200rpm) connected to MBP. Transfering .avi files using firewire 800 takes about 20 seconds per gb so around 50mb a second - Is that not incredibly slow considering firewire 800 is rated upto 800 mb/s?
Control-click I get three options in a box Reload Page/Save Page As/Print Page File Save As I get "audio-sample-2.webarchive" could not be exported as "audio-sample-2" I'm a transcription and need to receive audio files from clients in various ways. Info: MacBook Pro, Mac OS X (10.6.8), User of Express Scribe Pro
Does anybody now how to retrieve information from mpg video files. I mean information like video and audio codecs, bitrate, sampling rate, the number of channels, etc. I'm right now using a reference to the activemovie control library (quartz.dll), but I only figured out to retrieve frame size and the video length.I hope somebody knows how to retrieve the other information.
I am executing a computeSpectrum function every 1/10 of a second (the speed I should be sampling the sound frequency to obtain a data bit). However the computeSpectrum seems to be giving me the same output every two or three times I call it. Take a look at the source code and check the Binary read. Notice the binary read is repeating numbers, and the lack of single '0' or '1' bits? There should be some single '0's and '1's in there! I even made arrays of all of the input to check if the uncompressed ByteArray data was repeating and sure enough, the samples were quite often IDENTICAL every 2 or 3 calls... PS: The audio sample data [More audio samples available there] I'm using is Peceptive Development's 'How to talk to Tin Can' [More info on reading audio as data there]. The audio sample provided in the archive is 'Low speed', which means the data is being sent at 100 baud (100 baud / 1000 ms = 10 bits per second)
I have managed to achieve the BSOD quite a lot, and have tried re-installing Windows 7 x64 on my computer three times today. I have received the BSOD: after trying to reformat my F: drive. Uninstalling my Nividia drivers for a new one After using Firefox for a while it just won't open and BSOD follows shortly after I believe after installing Windows Live applications it startes to give me BSOD After the first BSOD I start to get the error message: Check filesums or somehting about the computed sums is not correct.
I wish to add an external drive to my iMac for time machine backup (at least 1TB) of my movies and music. My firewire 400 port is for my audio interface and the USBs are for printer, camera etc. I work with Final Cut Express and Logic Studio so my hard drive is quickly filling up! I assume the firewire 800 will be the one to choose for speed - any thought or recommendations? Information: iMac 2.66 GHz Intel Core 2 Duo, 4G Ram Mac OS X (10.5.8) Logic Studio, FCE, iPod Touch, Apple TV
I do alot of recording and music editing, and i'd hate to get noise from the fans into my recordings. I've read that FireWire devices (Audio Interfaces) causes the processor to get 10 degrees C higher (50 F?), wich means more fan noise, is this still true?Should i let this be the deciding factor on wether to get the Retina or a regular, or isn't it as bad as they say? Im also reading that people are having trouble with Cubase on the new Retina, and FireWire adapter isn't even out yet. Info: MacBook
My Pavilion dv8333ea's (specs here) GPU died the other day It was getting old (3.5 years) but I'd never had a problem with it before and aside from the GPU it still works fine (except the screen looks like it may have been programmed by Picasso (random pixels everywhere!). If this were any other laptop, I'd just replace it, but there was something special about this one: It had a firewire port with a Texas Instruments chipset (this is the Holy Grail of firewire audio recording). Plus, it genuinely was a good laptop
My computer has been stable for years, however recently I have been getting BSOD when doing any task. Windows will sometimes load to the desktop and I am able to watch video, run windows score, check email, etc before it will BSOD. Other times, it will BSOD before getting to the windows login screen.When I use safe mode, it logs in and stays there - no BSOD.I have tried removing all extra hardware in the machine, however this has had no affect. I have also tried one stick of RAM, then another, both BSOD.
My MacBook Pro (OSX 10.6.8) can not find the FireWire port. Until now there hasn't been any problems using the FireWire, but now it just deny any knowledge of it, so to speak.The audio device lights up when I connect it to the MBP (yes, I did connect it when in off-mode), so it seems to work in a way.
I have hooked up[ an m audio 2626 to my imac. drivers have been downloaded, followed all the steps. computer is not seeing a connection to the firewire port. When i connect to my macbook pro, no problem. How do I get a connection to the firewire on my imac
With the report that the new mac pros don't play nicely with Pro Tools and some unsubstantiated claims on a forum or two that Logic Pro doesn't handle as well with the new mac pros in comparison with the 08 models. Can anyone with a new model answer if the new models use the TI firewire chipset, or the troublesome Agere chipset. I'd think that a lot of Audio folk would love to know, especially with the recent macbook unibody no firewire, MBP reverting to the Agere chipset hoopla.
I'm trying to setup an audio-only, on-demand HLS stream in FMS 3.5. I have no problems streaming the sample f4v files via HLS, nor do I have any issues streaming the mp3 files via RTMP to a Flash client. However, when I try to stream a sample mp3 via HLS (the mp3 file is located in the same directory as the sample f4v's), I get a 404 error. I can't find anything in the documentation about streaming audio via HLS on-demand.
I'd like to capture the audio data from an RTMFP stream to which the client is subscribed (so I get a bytearray of audio samples).The presence of the audioSampleAccess propery on the NetStream class certainly makes that sounds possibe: For RTMFP connections, specifies whether peer-to-peer subscribers on this NetStream are allowed to capture the audio stream. When FALSE, subscriber attempts to capture the audio stream show permission errors.[code]But in the case of audio, I dont know how to address the audio data to get it into a bytearray.My instinct said this wasnt possible, but the presence of the 'audioSampleAccess' property makes me think it might be..
i was downloading mp3s in limewire again and in the process of my friends changing the name of my external hd osx for some reason just created its own name for the device. so now in limewire when i look in the volumes folder theres a "firewire HD" and a "firewire HD 1" which all the files are now being saved to... and i can't access it. how do i get into the volumes folder to transfer all the new mp3's to the original firewire hd and delete the new firewire hd 1 folder?
While recording the flv is saved inside applications/stream/samples/audio.flv. But it is not working properly.[code]
I'm trying to have Logic chase my Yamaha AW4416 and am having troubles. I'm using an old PowerBook G4 with Logic Pro 8.0.2 and an M-Audio Firewire 410 interface. I have the MIDI Out of my AW4416 connected to the MIDI In of the Firewire 410. I've set the frame rate to 25 for both Logic and the AW4416 (both support MTC). I enable Sync in Logic and when I press play on the AW4416, the Play transport button in Logic turns green, the rightmost digits of the SMPTE and bar position change constantly (as though cycling), and the MIDI out value shows a constant "1 64 xx" where "xx" also cycles constantly. There's no play head. Am I missing something basic? This is the first time I've tried to use Logic as a slave to chase an MTC master. Does the MIDI interface (M-Audio Firewire 410 in this case) need to specifically support MTC? (If yes, I'm not sure if the Firewire 410 does.) Information: PB G4 1.25GHz Mac OS X (10.5.8)
I'm new to this forum, just registered and you've helped me before with some of my mac issues.. Recently, I was offered a Powermac G5 2.3 Ghz dual core. I'm interested but I want to know if the machine is worth paying for, he was asking $1700.00 AUD. Here are the specs. POWERMAC CPU * 2.3GHz Dual PowerPC G5 processor * 1.15GHz frontside bus * 512k L2 cache * 2.5 Ghz RAM (400MHz DDR2 PC3200U DDR SDRAM Memory) * 300GB Serial ATA hard drive * 16x SuperDrive (double-layer) DVD Burner * 4 Firewire port PCI-X Expansion card * Two open PCI-X expansion slots * ATI RADEON 9650 Video Card with 256MB of GDDR SDRAM PORTS & BAYS * 1x FireWire 800 port * 2x FireWire 400 ports (one on front) * 4x USB 2.0 ports (one on front) * 2x internal SATA hard drive bays BUILT IN AUDIO * Optical digital audio input/output * Analog line-level input/output * Front headphone minijack and speaker Upgradable to 8Gb RAM and 2-3 Terabyte of Hard Drive I was told its a good, non-problematic machine. Are they telling the truth. I am also planning to check it out, what should I check or watch out for.
i'm a user of pro video apps and i have a question that i'm having trouble finding answers to. I'm trying to decipher all the different ways data rate or bit rate is reported in different info screens in OSX. Case study: exported an mpeg 4 of a 2 minute video i edited. was told by web designer the specs and followed exactly. Used the MPEG 4 preset in FCP>export using quicktime conversion. vid size was 480x360 bit rate was set to 500, and honestly i don't remember what the audio settings were. video came out to 30 megs which seemed a little high to me. and the web designer suggested i lower the bit rate to get the files size down bc when he viewd the info in QT it said it was around 1500. now here's my confusion. I encoded the vid at 500kbps. QT7 reports the "data rate" as 1687.21 kbps, putting the vid in iTunes the info is "bit rate" 101 kbps and "total bit rate" 1685, opening info from Finder, "total bit rate" is 604. what is one supposed to make of this info. i realize that total bit rate includes the audio as a sum of the video and audio rates. and i realize that it's reported as an average with VBR formats, but why the differing "total bit rates" btw Finder and QT? where does itunes get the 101 kbps from. does this mean that even though the vid was encoded at 500 kbps that the audo rate was around 1000 kbps? Information: MacBook Pr0 2.33 intel Mac OS X (10.6.4)
I have a FW-1082 mixer/sound device connected to a firewire pcie card, and my problem is that whenever i try to shutdown or restart computer i get a bsod. after some research on the net, i have found that i can make a net stop .bat file to stop "AudioSrv" and "AudioEndpointBuilder". However when i run the batch file for the "AudioSrv" i get a System error 5 has occurred, Access is denied, message. Can anyone please tell me how i can resolve this?
Windows 7 . . .- x64- the original installed OS on the system? YES- Retail- 1.5 - 2 year old system- Original OS install (have not had to re-install Windows 7)I'm about to swap the motherboard out for an M5A99X Evo but want to make sure it's the MB that's bad, first... I'm sure I'm having some driver issues, at least with my Alesis I|O26 FireWire. Lots of BSOD dumps in that zip file attached!
Even after ticking "Convert higher bit rate songs to 128kbps AAC" under "Options" in "Summary" of the iPhone page, I cannot seem to synchronise any audio tracks higher than the 16-bit, 44.1kHz format. For example, I own high-resolution audio tracks (i.e. 24-bit, 192kHz sampling rate) which plays fine on iTunes, but iTunes doesn't seem to have converted these to 128kbps AAC although it appeared that downconversion was going on. Can I just assume that iTunes gives a false impression that it's converting these high-resolution tracks to AAC and syncing them to my iPhone 3G, although nothing happened in reality? Information: 13" 2.53 GHz MacBook Pro Mac OS X (10.6.4) 4 GB RAM, 500 GB 7200 RPM HD
hi guys ,i have two tables in a access databaseTable A have this two datarate rate id 1.5 A2.4 B3.2 C4.5 DTable B have this datas rate(this is taken from rate id) /working hrs/ nameA 8 KenA 8 JoeB 8 Dan The Problem is that whenever is display the data from table b ,i am able to get the correct name , working hrs , but i want to display the rate as 1.5 , 2.4 instead of A,B...D in table B.So anyone has idea how this shld be done or any samples ?Thnxs In Advance
I brought my blackberry in May of this year and could download large mp3s in excess of 70 MB at a rate high rate over 70 kB/s. In the past month or so I am unable to download even small audio files which download at a rate of less than 1 kB/s which ends with a downoad failed message.