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• Video and Audio Drift

  - by Cenoc

  Hey everyone, I was wondering, how much does recorded audio and video drift from their actual recording time usually? I'm recording both separately (into unsigned 8 bit PCM (44100 Hz) and raw image data files) and I was wondering how much I can expect each to drift. Thanks in advance!

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• Audio Recording in C++

  - by Cenoc

  Hey, I was wondering, what was a good cross-platform utility for doing audio recording/ playback/ seeking in C++? I was thinking going the route of ALUT (OpenAL), but is there a better way? If not, do you guys know of any good tutorials/sample code for ALUT?

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• Audio Panning using RtAudio

  - by user1801724

  I use Rtaudio library. I would like to implement an audio program where I can control the panning (e.g. shifting the sound from the left channel to the right channel). In my specific case, I use a duplex mode (you can find an example here: duplex mode). It means that I link the microphone input to the speaker output. I seek on the web, but I did not find anything useful. Should I apply a filter on the output buffer? What kind of filter? Can anyone help me? Thanks

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• Extract music files from a Audio CD [closed]

  - by Jatin

  Possible Duplicate: What good, free audio CD ripping/extraction tools exist for Windows, and supporting multiple formats? I have an audio cd, which has audio files with the file format as .cda ( CD Audio Track ). Each one of these files have a size of 1 KB each, and the rest of the CD has nothing else. Is there a way that I can get the audio files from the CD and then convert it into mp3 format and then play it in any other devices as I like.

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• Multiple Audio I/P and O/P simultaneaously

  - by Raj Naveen

  hi (1) i saw in one of your posts that it is possible to get different outputs in windows 7. i am eager to know more. Is there any way i can create a 2 or more virtual cable between two softwares simultaneously. so that simultaneously, two or more audio inputs will be routed to equal no of audio analysers receivers, and then the audio analysers send back a filtered audio back to respective audio inputs... Please reply to email id: [email protected]

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• Nyquist won't play audio

  - by erjiang

  I downloaded Nyquist, and am having trouble playing sounds from it. If I run it normally, I get: Nyquist -- A Language for Sound Synthesis and Composition Copyright (c) 1991,1992,1995 by Roger B. Dannenberg Version 2.29 > (play (osc 60)) Saving sound file to ./eric-temp.wav error: snd_save -- could not open audio output > If I wrap it by running padsp ny, the sound plays fine for about half a second, and then I get garbage fed to my speakers. Any solutions?

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• Multiple Audio listeners in Scene

  - by Kevin Jensen Petersen

  THIS IS UNITY Im trying to make a FPS game over networking, it works fine. But now, when im trying to implement sound, it won't work. My guess would be, to add a Audio listener to the prefab, that gets instansiated whenever a player connects to the server, however the problem about this is that each player's audiolistener have been switched out which the other player(s), so the AudioSource won't play at the player, but at someone else in the game. Any suggestions ?

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• Audio Stutters at gdm

  - by Allan

  Ok I have a problem every 2 times out of 3 I login (I cant be specific it fairly random) I get a Stuttering GDM warning (not the login sound just the Bell sound to wake you up) the only way to stop it is to login I have a Fujitsu Siemens Amilo 1718 with a 2gig of memory (only hardware mod) using 10.10 Maverick and I have disabled KMS as my system was freezing as per the release notes. The only time this has happened before on the same machine was when I gave Kubuntu a try when 10.04 came out then it happened at the login screen and at random times while listening to music in any program. By the way audio is fine as is almost everything else once I have logged in. I would like an answer to this as I am an advocate of Ubuntu and its kind of embarrassing when the first thing that happens is *bing*. as requested Daniel alsa-info Pulse verbose log Not sure how useful the pulse log will be as I cant replicate the bug with a terminal open but I wouldnt be asking the question if I knew the answer so..... Edit 24/12/2010 ......been living on cocktail sausages and pickled onions for five days now made a make shift splint with cocktail sticks..... oops so updated the alsa drivers but I still get the same message in the dmesg No response from codec, disabling MSI: last cmd=0x10a90000 googleing it brings up a forum post from some other distro with a green logo the only common denominator seems to be graphics ie ATI Radeon XPRESS 200M which is why I have had to turn of kms as the chip is so old that small mice try to eat the "kernel" ;) funnily enough following the bug link at the end of the post, I found a comment about "Ubuntu Black Magic" so mabey I am coming at this from the wrong angle...... Bad Joo Joo any one. I will try the second part of Daniels Fix and Update with the result. The final Edit: (Plays air guitar) In the end neither of these solved the problem as such However I have given Roland a tick for reminding me of the solution and I gave Daniel the Bounty for the effort in trying to solve the problem. The answer for future readers was the enable the correct HD Audio Model I found the answer back when using Karmic Koala 9.10 in this forum post Amilo Li1718 Skype - Can't get it working... the model is options snd-hda-intel model=3stack position_fix=1 enable=yes which can be added to the end of alsa-base.conf thanks all for helping and hope anyone with a similar problem will find the answer here.

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• Synchronizing audio with scrolling text

  - by mr yoshida

  I am trying to have a website that vertically scrolls about 5 paragraphs of text with a matching audio file that reads along with it. It doesn't need to be synchronized word for word such as highlighting each spoken word but an accurate start and stop time. I've searched for quite a bit on the most efficient way of doing this but can't seem to find any answers. I tried Flash but really don't want to use it. Thanks in advance.

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• Create Audio file on iPhone/iPad from many other audio files (mixer)

  - by Brian

  I am trying to create something similar like Piano app on the iPhone. When people tap a key, it play a piano note. Basically, there will have only 7 notes (C) at the moment. Each note is a .caf file and its length is 5 seconds. I do not know if there is any way to save the song user played and export to mp3/caf format? The AVAudioRecord seems only record from the microphone input. Many thanks

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• Problems with MediaRecorder class setting audio source - setAudioSource() - unsupported parameter

  - by arakn0

  Hello everybody, I'm new in Android development and I have the next question/problem. I'm playing around with the MediaRecorder class to record just audio from the microphone. I'm following the steps indicated in the official site: http://developer.android.com/reference/android/media/MediaRecorder.html So I have a method that initializes and configure the MediaRecorder object in order to start recording. Here you have the code: this.mr = new MediaRecorder(); this.mr.setAudioSource(MediaRecorder.AudioSource.MIC); this.mr.setOutputFormat(MediaRecorder.OutputFormat.THREE_GPP); this.mr.setAudioEncoder(MediaRecorder.AudioEncoder.AMR_NB); this.mr.setOutputFile(this.path + this.fileName); try { this.mr.prepare(); } catch (IllegalStateException e) { Log.d("Syso", e.toString()); e.printStackTrace(); } catch (IOException e) { Log.d("Syso", e.toString()); e.printStackTrace(); } When I execute this code in the simulator, thanks to logcat, I can see that the method setAudioSource(MediaRecorder.AudioSource.MIC) gives the next error (with the tag audio_ipunt) when it is called: ERROR/audio_input(34): unsupported parameter: x-pvmf/media-input-node/cap-config-interface;valtype=key_specific_value ERROR/audio_input(34): VerifyAndSetParameter failed And then when the method prepare() is called, I get the another error again: ERROR/PVOMXEncNode(34): PVMFOMXEncNode-Audio_AMRNB::DoPrepare(): Got Component OMX.PV.amrencnb handle If I start to record bycalling the method start()... I get lots of messages saying: AudioFlinger(34):RecordThread: buffer overflow Then...after stop and release,.... I can see that a file has been created, but it doesn't seem that it been well recorderd. Anway, if i try this in a real device I can record with no problems, but I CAN'T play what I just recorded. I gues that the key is in these errors that I've mentioned before. How can I fix them? Any suggestion or help?? Thanks in advanced!!

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• Looking for an application to record audio and video on a linux "embedded" device

  - by Luke404

  I am working with a linux x86 device with limited CPU resources (as a prototype we just use a pentium-m netbook). We'd like to record video from one V4L2 device (we'll probably end up using just USB Video Class devices like all modern webcams) and one audio stream from an ALSA source. The thing will not have screen and keyboard, and obviously no X11 environment. Goals are: do as little work as possible to cope with little cpu resources - for example I'd like to record video in the native MJPEG I get out of the UVC devices encoding audio to MPEG3 Layer-2 (aka mp2) is ok since it let us save a lot of space (compared to raw pcm samples) and does use little cpu power I don't mind loosing some video frames here and there (UVC devices do that) as long as I can get audio and video streams syncronized not require user input to start the thing (a python script takes care of initialization, startup, shutdown, etc...) be able to open the resulting files for postprocessing without too much effort (ie, if mplayer or vlc can play it, it's fine) So far the only app I found that could be started from command line and record V4L2 video + ALSA audio is mencoder but I'm having some difficulties with it. It should be able to do that but I cannot record audio and video together - just one of the two. And if I use two different processes to record to two different files I have no means to get them in sync (audio is more or less always correct, but video framerate will vary over time and it seems to lack timestamps to correctly play it back to the correct time). Long story short, how do you record an unconverted MJPEG stream (from an UVC device) and an audio stream (from an ALSA device, possibly encoding to any standard format) using a command line tool, to a single file (MPEG or any other container), keeping audio and video in sync?

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• BluRay audio/video stuttering with PowerDVD 11, WinDVD 11 Pro, etc? Xonar/Auzen HD audio option?

  - by jrista

  I recently upgraded my Windows 7 MediaCenter HTPC due to a motherboard failure (really old motherboard and cpu, it was on its last legs.) I chose to upgrade to an i5 system with everything built into the motherboard. I did my due diligence, researched, and found some hardware that was within my budget. I ended up with: Core i5 2500K (3.3Ghz) Corsair XMS3 2x2Gb DDR3 (4Gb) ASUS P8H 61-M LE/CSM MicroCenter 64Gb SSD (Previous BluRay player, forget the brand) The system is pretty awesome, and plays everything I have perfectly. I almost went with an Atom solution, however there have been numerous notes that they do not play NetFlix Instant Watch well...and I am a heavy Netflix IW user. High definition BluRay rips work well, although they usually contain lower audio quality than the BluRay's they were ripped from. The real problem I am encountering is playing back BluRay video from discs. For some reason, I am encountering rather terrible stuttering problems with both the audio and video. The stuttering is synchronous in both, and occurs at seemingly random intervals. I've used PowerDVD 9, PowerDVD 11 trial, and WinDVD 11 Pro trial. All three have stuttering problems, although PowerDVD 11 seems to have the least. Watching system resource usage, CPU load is never above 20%, and memory usage tends to be a constant 1/3rd the total available system memory. When playback is fine, its superb...the video is crystal clear. The audio quality is ok, certainly not what I would expect from a BluRay disc. I did some research, and it seems that playing BluRay from a PC causes a downsampling of the audio? I am curious if the audio is my primary problem here, the cause of the stuttering I am encountering? When stuttering occurs, the audio gets REALLY bad, while the video just pauses momentarily every second until for whatever reason everything picks up and runs fine (usually after a few seconds to a couple minutes.) The audio chipset is a Realtek HD ALC887 8-channel, supposedly designed to support BluRay playback. Has anyone encountered any issues like this playing back bluray discs on a PC (namely with PowerDVD...WinDVD was FAR worse, and seemed to have real trouble even reading the discs, and I have no interest in fiddling with it further.) Is there any reason to suspect the video decoding as the problem?(Given how bad the audio gets during a stutter, and how clean the video remains, I am inclined to think the issue boils down to audio.) Is it even remotely possible that the motherboard, cpu, or ram are causing the stuttering (all three are pretty blazing fast...faster than the hardware that I replaced, which seemed to play BluRay fine with PowerDVD 9.) I've read a bit about the Asus Xonar HDAV 1.3 and the Auzen X-Fi HomeTheater HD home theater hi-fi audio cards. Seems they are the only way to get true full-quality, uncompressed BluRay audio bitstreaming over HDMI on a PC. None of the usual suspects seem to have these cards in stock, however. Are these cards worth getting? Are they even still available, or have they been discontinued (if so, that would indeed be sad...they sound simply fantastic.)

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• FFmpeg not recording audio during screen capture

  - by King

  I'm using the script below to run FFmpeg on Ubuntu 10.10. I followed these instructions to install FFmpeg & x264. While ffmpeg does capture the screen it does not capture the mic audio. I've checked that the mic works via "System Preferences". Anyone have any ideas on what the problem(s) could be and suggestions on how to resolve this issue? Thanks. ffmpeg -f alsa -ac 2 -i hw:0,0 -f x11grab -r 30 -s $(xwininfo -root | grep 'geometry' | awk '{print $2;}') -i :0.0 -acodec pcm_s16le -vcodec libx264 -vpre lossless_ultrafast -threads 0 -y screen-capture.mkv

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• Looking for Non Hosted Audio & Video Podcasting Solution for Church Websites

  - by motboys

  I am looking for a solution that will do the following: User uploads audio and/or video files with title, desc. image etc Solution embeds info into ID3 tags Solution generates RSS feed Solution embeds new content in our website Content on website is searchable This is for a couple of church websites I manage. I am looking for the ability to do the above with a sermon mp3 and also a video. At the moment we are doing it with multiple steps / people involved and I want to automate the process. I can't seem to find a solution that does all of the above. Thank you!

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• Which API for cross platform mobile audio?

  - by deft_code

  This question focuses on the API's available on phones. I'd been planning to use OpenAL in my game for maximum portability. It runs great on Linux so I can quickly develop the Game and leverage it's superior debugging tools. However I've recently heard that Android doesn't support OpenAL well. Instead they've gone with a OpenSL ES library. What I'm looking for is a free Audio library that I can use with minimal custom code on iPhone, Android, and my Linux desktop. Does such an API exists? Some extra details: The game is written in C++ with custom minimal front ends. ObjC for iPhone, Java for Android, and SFML for Desktops. I'm using OpenGL ES for portability as iPhone doesn't support the more advanced OpenGL APIs.

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• Virtual audio driver for Windows?

  - by Ognjen

  Is there any (possibly free or open-source) virtual WDM audio driver for Windows, with additional processing plugins, which would add one more layer between windows applications and actual sound card's WDM audio driver, allowing to: Add software DSPs to general audio output. I would like to be able to use custom effects, like compressor, or stereophonic-to-binaural converter for listening online's streaming media on headphones, etc. Connect its output to some custom buffer instead of the sound card. For example, to be able to record audio, or to send audio via wireless connection to some other wireless source? Virtual audio driver was just my idea how to solve these issues - if you know other way, please share your knowledge. I need this for Windows 7 and/or Windows XP.

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• Audio doesn't work on Windows XP guest (WS 7.0)

  - by Mads

  I can't get audio to work with on a Windows XP guest running on VMware Workstation 7.0 and Ubuntu 9.10 host. Windows fails to produce any audio output and the Windows device manager says the Multimedia Audio Controller is not working properly. Audio is working fine in the host OS. When I open Multimedia Audio Controller properties it says: Device status: The drivers for this device are not installed (Code 28) If I try to reinstall the driver I get the following error message: Cannot Install this Hardware There was a problem installing this hardware: Multimedia Audio Controller An Error occurred during the installation of the device Driver is not intended for this platform Has anyone else experienced this problem?

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• Bluetooth Audio and SoftPhone Audio Input/Output

  - by o7th Web Design

  I have a Voip Softphone software that I would like to start using on my Ubuntu 14.04 box. Here's the thing. My system sound right now goes through my HDMI to my speaker system so I can play music all day ;-) I have a bluetooth headset connected to the machine as well. What I am wondering is if there is a way to: Auto-mute the music when a call comes in Auto-switch the sound devices when a call comes in, from my hdmi sound device, to my headset Auto-switch back when the call ends, and auto-un-mute the music Or even just an auto-switch to the headset? I can always pause the music ;)

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• iOS 5 Audio Alarms Don't Sound Without kAudioSessionProperty_OverrideCategoryMixWithOthers On

  - by coneybeare

  I have an audio app that is having some problems with the way iOS 5 has changed audio behaviors. When my app's audio is playing (AVAudioSessionCategoryPlayback), and a Clock.app alarm or timer is fired from the OS, the UIAlertView notification pops up, but without the audio alert. My application sound ducks fine to get out of the way of the audio alert, but the alarm app's audio alert does not sound. Naturally, tons of support requests poured in over the iOS 5 change. I have solved this temporarily by setting kAudioSessionProperty_OverrideCategoryMixWithOthers which lets the alarm audio come through, but there are a few very undesirable side-effects when doing this: Other app's audio can play with/over mine. The remote control events are not routed to my app, but to iPod.app. None of the above drawbacks are acceptable for my app's requirements. I have been hacking away at this for some time now but haven't been able to crack it. How can I setup my audio such that: My app's audio still uses the AVAudioSessionCategoryPlayback category for background audio. The Clock.app alarms still have their audio alerts make sound The app still responds to remote control notifications

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• record output sound in python

  - by aaronstacy

  i want to programatically record sound coming out of my laptop in python. i found PyAudio and came up with the following program that accomplishes the task: import pyaudio, wave, sys chunk = 1024 FORMAT = pyaudio.paInt16 CHANNELS = 1 RATE = 44100 RECORD_SECONDS = 5 WAVE_OUTPUT_FILENAME = sys.argv[1] p = pyaudio.PyAudio() channel_map = (0, 1) stream_info = pyaudio.PaMacCoreStreamInfo( flags = pyaudio.PaMacCoreStreamInfo.paMacCorePlayNice, channel_map = channel_map) stream = p.open(format = FORMAT, rate = RATE, input = True, input_host_api_specific_stream_info = stream_info, channels = CHANNELS) all = [] for i in range(0, RATE / chunk * RECORD_SECONDS): data = stream.read(chunk) all.append(data) stream.close() p.terminate() data = ''.join(all) wf = wave.open(WAVE_OUTPUT_FILENAME, 'wb') wf.setnchannels(CHANNELS) wf.setsampwidth(p.get_sample_size(FORMAT)) wf.setframerate(RATE) wf.writeframes(data) wf.close() the problem is i have to connect the headphone jack to the microphone jack. i tried replacing these lines: input = True, input_host_api_specific_stream_info = stream_info, with these: output = True, output_host_api_specific_stream_info = stream_info, but then i get this error: Traceback (most recent call last): File "./test.py", line 25, in data = stream.read(chunk) File "/Library/Python/2.5/site-packages/pyaudio.py", line 562, in read paCanNotReadFromAnOutputOnlyStream) IOError: [Errno Not input stream] -9975 is there a way to instantiate the PyAudio stream so that it inputs from the computer's output and i don't have to connect the headphone jack to the microphone? is there a better way to go about this? i'd prefer to stick w/ a python app and avoid cocoa.

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• pitchbend (varispeed) audio with iPhone SDK's AudioUnit

  - by fetzig

  hi, I'm trying to manipulate the speed (and pitch) of a sound while playing. so i played around with iphone sdk's AudioUnit. downloaded iPhoneMultichannelMixerTest and tried to add an AUComponent to the graph (in this case a formatconverter). but i get (pretty soon) following error when building: #import <AudioToolbox/AudioToolbox.h> #import <AudioUnit/AudioUnit.h> ... AUComponentDescription varispeed_desc(kAudioUnitType_FormatConverter, kAudioUnitSubType_Varispeed, kAudioUnitManufacturer_Apple); ^^ error: 'kAudioUnitSubType_Varispeed' was not declared in this scope. any ideas why? the documentation on this topic doesn't help me at all (just api doc isn't very helpful when having no clue about the concept behind). there are no examples on how to wire these effects together and manipulating there properties...so maybe i'm totally wrong, anyway any hint is great. thx for help.

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• combining live recording audio in another audio file iphone sdk?

  - by Rahul Vyas

  Does someone knows how we can combine our current recording into another background playing music? I need some sample code for this. I want to create something like karaoke application in which we can record our voice over an instrumental music.

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• Can Flash player play .m3u audio files from a remote location

  - by undefined

  Is it possible to play a .m3u file streamed from a remote location through Flash Player in a browser? I have a player that loads and plays .mp3 files but also want to be able to play .m3u files. I have looked at the as3plsreader on google code but I think this is only for AIR and desktop files. anyone tried this or know where I should start looking for an answer? If I wasnt to use flash, what other ways could I get remote m3u files to play in a browser?

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• How can I get it the Free Music Archive audio player or is there a better alternative?

  - by Dennis Hodapp

  I'm looking at free streaming audio players for web browsers that I can use in a project. I really like the audio player used on http://freemusicarchive.org/. Are they using an open source audio player and can I get a hold of it? Or is it closed source? Also if there are any open-source audio players that anybody knows about I'd love to know about them (preferable to have one with no flash). Last thing...is HTML5 going to be able to replace audio streaming players?

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