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'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.
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
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.
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).
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?
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'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 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 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..
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.
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 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
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 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 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.
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 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
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 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.
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.
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.
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
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
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..
While recording the flv is saved inside applications/stream/samples/audio.flv. But it is not working properly.[code]
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'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 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!
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)
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 installed Boot Camp 1.3 so I could run Windows XP on a partition (use it for music); no problem. Set up my Motu Ultralite on the OSX side, no problem. HOWEVER: The Motu UltraLite doesn't work on the XP partition, at all. It shows up, but no audio can be routed through it. It's a firewire audio card, if you're not familiar with it.
I have a newer imac and the WD 4TB raid drive, which i have seen use up about 70 MB/s max with large transfers. Problem is since i partitioned it and run time machine on it, it slows down anything on that drive while its backing up. So i was thinking of getting a 2TB drive for time machine and backups. Along with an M-Audio project mix audio interface when will it start affecting the firewire bandwidth Information: iMac Mac OS X (10.6.1)
Is it possible to connect my 27" Intel iMac to audio equipment (receiver, pre-amp, integrated amp, speakers) WITHOUT losing the audio from the built-in speakers (they make a nice center channel)? If so, how? Are there USB or FireWire devices that do the job? Information: 27" iMac v3.06 GHz Intel Core 2 Duo Mac OS X (10.6.3) 3 cats, i dog, 2 PBG4s,3 (I think) iMac Indigo DVs, I B&W G3
I am using the firewire sound card that uses Core Audio standard driver. The card is fully functional but volume control is not working even if I set it as default device for input/output. Volume control works for built-in audio. It would be nice to know how the OS X volume control is designed to work. Information: Mac OS X (10.5.7)
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.
Does anyone know how to get details from Mpeg videos like bit rate, frame rate, video size, audio details etc? I'm working on a front end type program for Windows Media Player. In addition to that I also need to get info from AVI's and WAV's (codec used, bit rate, video size etc). If anyone could help I would be very thankfull.Excremedies.
My audio was working well, but i decided to start recording find a way to improve my outbound audio quality (i'm an audiophile). I purchased a Fast Track Pro. Unlike other threads I've read, I'm not seeing any BSOD and its not crackling. My operating system will just stop functioning, or freeze for a half a second at a time. This can become frequent at about 10 times in a thirty second period requiring a restart. I had installed the latest x64 driver for my W7 x64 system from M-audio's website. I uninstalled the driver and restarted, the issue persisted. I then installed the on-board audio driver and restarted, no system lag is present. The audio device-driver is acting up and I'm not sure why. Because i bought it used M-audio's support is a no-go. Are there any additional steps I can take which may troubleshoot this issue?
I am experimenting with video using the alpha channel capability of the FlashCS3. I followed the tutorial on the gotoandlearn website on how to create chroma keyed video. I exported an avi video file having removed the background. Ive used Flash video encoder for VP6 and flash player 8 with encode alpha channel ticked. However my swf is not showing the video, just a white blank where it should be. It puts out the audio only. This is the log file. FILE SUCCESSFULLY ENCODED - Source file: C:downloadsflash greenscreenvideoscreentestalpha.avi - Output file: C:downloadsflash greenscreenvideoscreentestalpha2.flv - Video codec: On2 VP6 - Alpha channel encoded: yes - Deinterlace: no - Frame rate: 25 fps - Key frame interval: 50 frames - Video data rate: 40 kbps - Width: 720 pixels - Height: 576 pixels - Audio codec: MPEG Layer III (MP3) - Audio data rate: 16 kbps (mono) - FLV duration: 00:00:10 - Encoding time: 00:01:00
i just had this problem, i need urgent help! I am working on a sound for picture project.After working and mixing in logic studio 8 i bounced the track, all the settings (Fps, Bit and sample rate) are matched into final cuts own project settings but still theres an aprox. 15 frame offset. Its basically like an error on the sample rate config but is all correctly adjusted, i dont know what to do, Information: macbook pro Mac OS X (10.6.4)
Imported audio is way out of sync with the video. The audio and video is fine when viewing the tape in the camera. The audio was recorded at 16 bit. 'Abort capture on dropped frames' in preferences is checked, but there are no warning about dropped frames. 'Sync audio capture to video source' is checked as well. Capture preset is DV NTSC 48 kHz; device control is Firewire. I was importing audio and video on the same internal drive, so tried separating audio and video as separate tracks on two different internal drives. Audio is still out of sync. Computer is PowerMac G5 2GHz, 4.5GB RAM with 2 internal drives. FCP is v.6.06. Information: PowerMac G5 2GHz, 4.5GB RAM; 2 internal drives Mac OS X (10.5.8)
Is there a way to increase the sampling rate of Accelerometer values? I use the HIGHEST tag in the android and it gives me a sampling rate of 50Hz.Are there any new drivers or anything else that may allow me to go further, say till 100Hz?
I pulled in some of my DVD's from Handbrake for the family to watch on the iPad. I cannot for the life of me figure out why some will import, and some are refused by iTunes as not compatible. Even more annoying, when I use the Convert To iPad or Apple TV setting under Advanced, I get a white screen with audio as the new movie. Two for example that won't sync have these settings from Get Info: Kind: mp4 Size: 1.56GB Dimensions: 1280 x 688 (This imports as HD; one of the ones that end up with "white video" when converted) Codecs: H.264, AAC, Subtitle Duration 02:00:21 Audio channels: 2 Total bit rate: 1,727 Kind: mp4 Size: 1.05GB Dimensions: 853 x 480 Codecs: H.264, AAC Duration: 01:36:09 Audio channels: 2 Total bit rate: 1,450 This one for example WILL sync and has these settings from Get Info: Kind: mp4 Size: 1.77GB Dimensions: 1280 x 688 (this does not import as HD; syncs and plays fine) Codecs: H.264, MPEG 4 High Efficiency AAC Duration: 01:47:24 Total bit rate: 2,199 Information: MacBook Pro 15" 2.5GHZ Mac OS X (10.5.2)
hi every1, I know how to get the title, artist, u know, all the stuff for a MP3 TAG ( thanks to this web site :-)) Anyways, how can I know if a MP3 file is Sterio or Mono, the bitrate and the sample rate (KHz) and also, the lenght in seconds?How can i extract the decompressor type, Sample rate and bitrate from a WAVE file and the lenght in seconds?And the lenght in seconds of a MIDI file???Thanks for ur helpA+
I'm working on a program which requires recording via a mic which is played back.Using the MCI controls to record, the default sample rate is 8bit 11khz ish.I'd like to improve the recording quailty by changeing the sample rate to 16bit.I know its something to do with:mciSendString Lib "winmm.dll" Alias "mciSendStringA" (ByVal lpstrCommand As String, ByVal lpstrrtning As String, ByVal uReturnLength As Long, ByVal hwndCallback As Long) As Long But I can't work it out! Can anyone help please?
I am dealing with data sets from various instruments that have different sample rates. I am deleting data points I don't need from some of the sets with higher sample rates so that all the data is on the same time scale. The macro I have is super simple, but incredibly slow. I'm simply deleting every other cell down a column. VB: Sub OATcondense() Application.ScreenUpdating = False Do While ActiveCell <> "" ActiveCell.Offset(1, 0).Delete Shift:=xlUp ActiveCell.Offset(1, 0).Select Loop Application.ScreenUpdating = True End Sub
Getting this BSOD when starting it up in the morning. Afterwards the computer runs normally.I've tried to enable creating minidumps, but 3 days in (and 3 bsod's) it hasn't created one yet.I've copied info from the BSOD and Event Viewer below. Please let me know if any more info is needed.-Technical Information from the BSOD below: Code: *** STOP: 0x000000F4 (0x0000000000000003, 0xFFFFFA00E399B30, 0xFFFFFA800E399E10, 0xFFFFF80002F935F0)
I get the BSOD while seeding in uTorrent mutliple files. I have a stable notebook, never got an issue for years. Occasionally get some BSOD, but for reasons, now I have BSOD i cant explain for myself. uTorrent can seed for some time, but if I leave it ,i find it with the following BSOD. I doesnt do it right away, after sometime. check screenshot.
i usually started to got alot of BSOD after closing games like PES 2012 or S4 League and this is becoming annoying . I used WhoCrashed v3.06 (HOME EDITION ) to analyze my BSOD and this is what i got after my last BSOD error when i closed a game: [code] I can't understand what those two BSOD means , but i think they are the problem . I will attach the dump file i got and the files that i got from the SF Diagnostic Tool.
I have BSOD on a cold startup everyday. On most time after the first BSOD of the day the PC will run fine for that whole day. There will be two consecutive BSOD on certain occasions. The problem stops for a week after I updated the NEC USB 3.0 Host controller so it seems there is other problem causing the BSOD but now I am not so sure.
I'm trying to playback some sample data through the new real-time audio capabilities of Flash Player 10. I started with the example given at the bottom of this page on livedocs, which seems to work fine and plays a crystal clear tone.I assume that the two writeFloat's in the example write to the left and right audio channels respectively and that the data being written is 32 bit (because of the float). However. I seem to be having trouble converting my old 8 bit audio data to a format that is understood by this interface. When I playback my sample data I can vaguely hear the sound I'm expecting but it is massively distorted. My sample data consists of raw 8 bit samples that ranges from 0..255 where 127 is silence. I've been trying different conversion formulas but I seem to be missing some vital information regarding this conversion. UPDATE:The correct formula turns out to be: f = (sample.data.readByte() - 127) / 255
I have site with an audio player and a few galleries.The problem is that the audio starts loading at the beginning. And when you click on a gallery, the audio already uses all of the download speed and therefore the galleries load slow.Is there a way to limit the bit-rate flash the swf uses to load the sound?[code]