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• Client-side framework for web-app with good audio support

  - by Poita_

  I'm trying to create a client-side web app that generates music procedurally using some user-input parameters, so I'm looking for a framework (e.g. Flash, Silverlight etc.) that has the capability to play audio at a specified pitch. Whether it is playing a WAV/MP3 file, using MIDI output, or just playing beeps doesn't really matter -- I just need something that will enable me to generate arbitrary music client-side. I've done a bit of searching and it appears that Flash might have the ability to change pitch with the help of a third-part plugin, but I couldn't find anything similar for Silverlight. I can go a try all them out manually if need be, but I thought I'd ask here first just in case anyone had tried something like this before. Thanks in advance

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• iPhone game audio and background music

  - by Boon

  Have a few questions related to adding sounds to my game, specifically intro music (for splash), background music (loop) and button event sounds. Hope you can share your knowledge on this. 1) Should I use compressed sounds or uncompressed sounds? Or perhaps a combination of the two? Are there any limitations on the iPhone hardware that I should be aware of -- for example, the ability to play multiple compressed sounds? 2) What's the best audio format for my purpose? 3) For background music, I am thinking of using AVAudioPlayer. For button event sounds, I am thinking of using AudioServicesPlaySystemSound, what do you think? 4) Any other issues I should be aware of? Thank you!

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• directx audio video error message in debugmode

  - by clamp

  I have a c#/winforms application that uses directx to play some video and audio. whenever i start my application in debugmode i get this annoying message. i can click "continue" and everything seems to work fine. but i still want to get rid of this message. it does not show up in releasemode. Managed Debugging Assistant 'LoaderLock' has detected a problem in 'C:\pathtoexe.exe'. Additional Information: DLL 'C:\WINDOWS\assembly\GAC\Microsoft.DirectX.AudioVideoPlayback\1.0.2902.0__31bf3856ad364e35\Microsoft.DirectX.AudioVideoPlayback.dll' is attempting managed execution inside OS Loader lock. Do not attempt to run managed code inside a DllMain or image initialization function since doing so can cause the application to hang.

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• Naudio - putting audio stream into values [-1,1]

  - by denonth

  Hi all I need to put my audio stream into values of [-1,1]. Can someone tell me a good approach. I was reading byte array and float array from stream but I don't know what to do next. Here is my code: float[] bytes=new float[stream.Length]; float biggest= 0; for (int i = 0; i < stream.Length; i++) { bytes[i] = (byte)stream.ReadByte(); if (bytes[i] > biggest) { biggest=bytes[i]; } } and I don't know how to put values into stream. Because byte is only positive values. And I need to have from [-1,1] for (int i = 0; i < bytes.Count(); i++) { bytes[i] = (byte)(bytes[i] * (1 / biggest)); }

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• audio power on AudioQueue

  - by Tomoyuki

  Hi everyone. I'm now creating an Application using speech recognition.To check the Audio Power coming in through the microphone, I wrote a method as follows. -(void)checkPower(AudioqueRef)queue{ UInt32 expectedSize= sizeof(AudioQueueLevelMeterState); AudioQueueGetProperty(queue, kAudioQueueProperty_CurrentLevelMeter, audioLevels, expectedSize); NSLog(@"average:%f peak:%f",audioLevels.mAveragePower,audioLevels.mPeakPower); } I found that sometimes mAveragePower was larger than mPeakPower, and when mAveragePower was 1.0, in other words, averagePower is regarded as max, mPeakPower was lower than 1.0. I think that generally this result is inpossible. please Let me know if you have any information about sound power on CoreAudio. thanks.

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• High level audio crossfading library for python

  - by tcoopman

  I am looking for a high level audio library that supports crossfading for python (and that works in linux). In fact crossfading a song and saving it is about the only thing I need. I tried pyechonest but I find it really slow. Working with multiple songs at the same time is hard on memory too (I tried to crossfade about 10 songs in one, but I got out of memory errors and my script was using 1.4Gb of memory). So now I'm looking for something else that works with python. I have no idea if there exists anything like that, if not, are there good command line tools for this, I could write a wrapper for the tool.

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• Toggling audio on click?

  - by angela

  please look at this fiddle http://jsfiddle.net/rabelais/yLdkj/1/ The above fiddle shows three bars that on hover play audios. How do I change this so the music plays and pauses on click instead. Also if one audio is playing and another is clicked how can the already playing song pause? $("#one").mouseenter(function () { $('#sound-1').get(0).play(); }); $("#one").mouseleave(function () { $('#sound-1').get(0).pause(); }); $("#two").mouseenter(function () { $('#sound-2').get(0).play(); }); $("#two").mouseleave(function () { $('#sound-2').get(0).pause(); }); $("#three").mouseenter(function () { $('#sound-3').get(0).play(); }); $("#three").mouseleave(function () { $('#sound-3').get(0).pause(); });

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• audio stream sampling rate in linux

  - by farhan

  Im trying read and store samples from an audio microphone in linux using C/C++. Using PCM ioctls i setup the device to have a certain sampling rate say 10Khz using the SOUND_PCM_WRITE_RATE ioctl etc. The device gets setup correctly and im able to read back from the device after setup using the "read". int got = read(itsFd, b.getDataPtr(), b.sizeBytes()); The problem i have is that after setting the appropriate sampling rate i have a thread that continuously reads from /dev/dsp1 and stores these samples, but the number of samples that i get for 1 second of recording are way off the sampling rate and always orders of magnitude more than the set sampling rate. Any ideas where to begin on figuring out what might be the problem?

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• audio error in vmware running mac os x

  - by PenguinSource

  simple synchronous loading of an audio file (.mp3) in a cocos2d app makes my vmware disconnect the sound. the error is display bottom right, saying 'error in creating sound stream; sound is disconnected' i read that it might be cause of my vmware's version (mine is 8) but I'm looking for a fix, not to downgrade to another version. before i get that error, the sound on the system works just fine (youtube, etc) the exact code im calling is.. [CDSoundEngine setMixerSampleRate: CD_SAMPLE_RATE_MID]; [[CDAudioManager sharedManager] setResignBehavior: kAMRBStopPlay autoHandle:Yes]; soundEngine = [SimpleAudioEngine sharedEngine]; [soundEngine preloadBackgroundMusic:@"somemp3.mp3"]; [soundEngine playBackgroundMusic:@"somemp3.mp3"]; maybe the bit rate is too high .. ? thanks

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• How to calculate the audio file duration in core audio?

  - by mystify

  I have this info variable which is of this type: struct AudioStreamBasicDescription { Float64 mSampleRate; UInt32 mFormatID; UInt32 mFormatFlags; UInt32 mBytesPerPacket; UInt32 mFramesPerPacket; UInt32 mBytesPerFrame; UInt32 mChannelsPerFrame; UInt32 mBitsPerChannel; UInt32 mReserved; }; How could I calculate the total duration of the audio file, in seconds?

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• Seeking not working in HTML5 audio tag

  - by lord_wilmore

  I have a lighttpd server running locally. If I load a static file on the server (through an html5 audio tag), it plays and seeks fine. However, seeking doesn't work when running a dev server (web.py/CherryPy) or if I return the bytes via a defined action url instead of as a static file. It won't load the duration either. According to the "HTTP byte range requests" section in this Opera Page it's something to do with support for byte range requests/partial content responses. The content is treated as streaming instead. What I don't understand is: If the browser has the whole file downloaded surely it can display the duration, and surely it can seek. What I need to do on the web server to enable byte range requests (for non-static urls). Any advice would be most gratefully received.

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• Advice for building a browser-based audio mixer up to 32 tracks

  - by Jonathan P.

  As a personal hobby I am looking to build an online audio mixer where I can upload individual instrument tracks, control individual volumes of each track, and export the mixed down version. I've been trying (and have come pretty close) with javascript. I really would like to stay away from flash if possible, but I'm really looking for suggestions for technologies to try. If anyone has any suggestions on languages that are good at stuff like this or libraries that I am missing, please let me know! I have a test environment that I have been using: http://driverstestpractice.com/sandbox Currently all tracks on the site are set to the click track in order to test the track sync (which as you can tell is a little off)! Thanks!

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• Compare two audio files of beat/tempo and rating in iphone

  - by Senthil Kumar

  Hello, I want to develop iPhone application should have the ability to count the number of phrases that are received when user sing on mic. This application should also have the ability to decipher whether the users phrases are in or out of cadence with a preset beat.When user sing on mic Instrumental music only play. So I have to merge the User Recorded voice with Instrumental music this is one Audio file.Already i have on original Song file.I have to compare both and give the Rating to users. [Note: Instrumental music is without vocal of Original Song file] Can you please help me?. Thanks Vadivelu

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• Capturing Mac OS X System Audio output with Python

  - by richbs

  Hello, I've been trying to "hijack" the Mac OS X system audio using PyAudio and save to a wav in python. That is, I do not want to record from an input device such as a microphone. I want to grab the sound output from any or all applications. I have followed the tutorials on the PyAudio site but these do not appear to cover my use case and when I try to read from the output stream I unsurprisingly get the paCanNotReadFromAnOutputOnlyStream exception. Fair enough! Is there a way to do what I am proposing with the PyAudio or other FOSS Python Library?

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• zoom a JPanel and all of its components

  - by user253530

  I have a big JPanel which holds many TileMapping objects (TileMapping extends JPanel). I want to zoom the big JPanel and all of its components to be scaled aswell. A TileMapping has an image which of course has to be zoomed as well. here is my TileMapping class public class TileMapping extends JPanel { private double zoom = 1.0; private Mapping mapping; private int height; private int width; private int xCoord; private int yCoord; TileMapping(Mapping m, int w, int h, int x, int y) { super(); mapping = m; height = h; width = w; this.xCoord = x; this.yCoord = y; this.setPreferredSize(new Dimension(w, h)); } public int getXcoord() { return xCoord; } public int getYcoord() { return yCoord; } public Mapping getMapping() { return mapping; } public void setMapping(Mapping m) { mapping = m; } public void originalSize() { zoom = 1; } public void zoomIn() { zoom += 0.2; } public void zoomOut() { zoom -= 0.2; } @Override public void paintComponent(Graphics g) { Graphics2D g2d = (Graphics2D) g; if (mapping.getImage() != null) { BufferedImage bg = mapping.getImage(); g2d.scale(zoom, zoom); g2d.drawImage(bg, 0, 0, width, height, this); } else { g2d.scale(zoom, zoom); this.setBackground(Color.white); } } } I need to know how i should continue this on the main JPanel which is populated using this code: This is part of a MapWindow class. if you carefully read the code you will understand what i'm talking about. Basically this is a Tile MapEditor and I am simulating a Tile as a JPanel which holds the mapping object (mapping is a binding between an ASCII character and an image) you place. private void populateMapPanel(Map map) { GridBagConstraints c = new GridBagConstraints(); mapPanel.setLayout(new GridBagLayout()); for (int i = 0; i < map.getMapGrid().getRows(); i++) { for (int j = 0; j < map.getMapGrid().getColumns(); j++) { tile = new TileMapping(map.getMapItem(i, j), cellSize, cellSize, i, j); tile.setBorder(new LineBorder(Color.black)); tile.addMouseListener(this); tile.addMouseMotionListener(this); c.gridx = i; c.gridy = j; mapPanel.add(tile, c); } } mapPanel.validate(); mapPanel.repaint(); } I need to know how to do the following: 1. When the user presses Zoom In the mapPanel should zoom in and also all the images in the smaller panels. 2. when the user click and drag over the TileMappings an image should be painted on them and this image should get scaled of course if the mapPanel is zoomed. Pseudo code, ideas, hints, java code ...anything would be helpful. Thank you

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• Perl to Ruby conversion (multidimensional arrays)

  - by Alex

  I'm just trying to get my head around a multidimensional array creation from a perl script i'm currently converting to Ruby, I have 0 experience in Perl, as in i opened my first Perl script this morning. Here is the original loop: my $tl = {}; for my $zoom ($zoommin..$zoommax) { my $txmin = lon2tilex($lonmin, $zoom); my $txmax = lon2tilex($lonmax, $zoom); # Note that y=0 is near lat=+85.0511 and y=max is near # lat=-85.0511, so lat2tiley is monotonically decreasing. my $tymin = lat2tiley($latmax, $zoom); my $tymax = lat2tiley($latmin, $zoom); my $ntx = $txmax - $txmin + 1; my $nty = $tymax - $tymin + 1; printf "Schedule %d (%d x %d) tiles for zoom level %d for download ...\n", $ntx*$nty, $ntx, $nty, $zoom unless $opt{quiet}; $tl->{$zoom} = []; for my $tx ($txmin..$txmax) { for my $ty ($tymin..$tymax) { push @{$tl->{$zoom}}, { xyz => [ $tx, $ty, $zoom ] }; } } } and what i have so far in Ruby: tl = [] for zoom in zoommin..zoommax txmin = cm.tiles.xtile(lonmin,zoom) txmax = cm.tiles.xtile(lonmax,zoom) tymin = cm.tiles.ytile(latmax,zoom) tymax = cm.tiles.ytile(latmin,zoom) ntx = txmax - txmin + 1 nty = tymax - tymin + 1 tl[zoom] = [] for tx in txmin..txmax for ty in tymin..tymax tl[zoom] << xyz = [tx,ty,zoom] puts tl end end end The part i'm unsure of is nested right at the root of the loops, push @{$tl->{$zoom}},{ xyz => [ $tx, $ty, $zoom ] }; I'm sure this will be very simple for a seasoned Perl programmer, thanks! `

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• C#: Streaming an Audio file from a Server to a Client

  - by Andreas Grech

  I am currently writing an application that will allow a user to install some form of an application (maybe a Windows Service) that will open a port on it's PC and given a particular destination on the hard disk, will then be able to stream mp3 files. I will then have another application that will connect to the server (being the user's pc) and be able to browse the hosted data by connecting to that PC (remotely ofcourse) given the port, and stream mp3 files from the server to the application I have found some tutorials online but most of them are about File Servers in C# and they download allow you to download a whole file. What I want is to stream an mp3 file so that it starts playing when a certain number of bytes are download (ie, whilst it is being buffered) How do I go about in accomplishing such a task? What I need to know specifically is how to write this application (that I will turn into a Windows Service later on) that will listen on a specified port a stream files, so that I can then access the files by something of the sort: http://<serverip>:65000/acdc/wholelottarosie.mp3 and hopefully be able to stream that file in a WPF MediaPlayer. [Update] I was following this tutorial about building a file server and sending the file from the server to the client. Is what I have to do something of the sort? [Update] Currently reading this post: Play Audio from a Stream using C# and I think it looks very promising as to how I can play streamed files; but I still don't know how I can actually stream the files from the server.

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• How to play simultaneous multiply audio sources in Silverlight

  - by Shurup

  I want to play simultaneous multiply audio sources in Silverlight. So I've created a prototype in Silverlight 4 that should play a two mp3 files containing the same ticks sound with an intervall 1 second. So these files must be sounded as one sound if they will be played together with any whole second offsets (0 and 1, 0 and 2, 1 and 1 seconds, etc.) I my prototype I use two MediaElement (me and me2) objects. DateTime startTime; private void Play_Clicked(object sender, RoutedEventArgs e) { me.SetSource(new FileStream(file1), FileMode.Open))); me2.SetSource(new FileStream(file2), FileMode.Open))); var timer = new DispatcherTimer { Interval = TimeSpan.FromMilliseconds(1) }; timer.Tick += RefreshData; timer.Start(); } First file should be played at 00:00 sec. and the second in 00:02 second. void RefreshData(object sender, EventArgs e) { if(me.CurrentState != MediaElementState.Playing) { startTime = DateTime.Now; me.Play(); return; } var elapsed = DateTime.Now - startTime; if(me2.CurrentState != MediaElementState.Playing && elapsed >= TimeSpan.FromSeconds(2)) { me2.Play(); ((DispatcherTimer)sender).Stop(); } } The tracks played every time different and not simultaneous as they should (as one sound).

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• Procesing 16bit sample audio

  - by user2431088

  Right now i have an audio file (2 Channels, 44.1kHz Sample Rate, 16bit Sample size, WAV) I would like to pass it into this method but i am not sure of any way to convert the WAV file to a byte array. /// <summary> /// Process 16 bit sample /// </summary> /// <param name="wave"></param> public void Process(ref byte[] wave) { _waveLeft = new double[wave.Length / 4]; _waveRight = new double[wave.Length / 4]; if (_isTest == false) { // Split out channels from sample int h = 0; for (int i = 0; i < wave.Length; i += 4) { _waveLeft[h] = (double)BitConverter.ToInt16(wave, i); _waveRight[h] = (double)BitConverter.ToInt16(wave, i + 2); h++; } } else { // Generate artificial sample for testing _signalGenerator = new SignalGenerator(); _signalGenerator.SetWaveform("Sine"); _signalGenerator.SetSamplingRate(44100); _signalGenerator.SetSamples(16384); _signalGenerator.SetFrequency(5000); _signalGenerator.SetAmplitude(32768); _waveLeft = _signalGenerator.GenerateSignal(); _waveRight = _signalGenerator.GenerateSignal(); } // Generate frequency domain data in decibels _fftLeft = FourierTransform.FFTDb(ref _waveLeft); _fftRight = FourierTransform.FFTDb(ref _waveRight); }

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• AS3 microphone recording/saving works, in-flash PCM playback double speed

  - by Lowgain

  I have a working mic recording script in AS3 which I have been able to successfully use to save .wav files to a server through AMF. These files playback fine in any audio player with no weird effects. For reference, here is what I am doing to capture the mic's ByteArray: (within a class called AudioRecorder) public function startRecording():void { _rawData = new ByteArray(); _microphone.addEventListener(SampleDataEvent.SAMPLE_DATA, _samplesCaptured, false, 0, true); } private function _samplesCaptured(e:SampleDataEvent):void { _rawData.writeBytes(e.data); } This works with no problems. After the recording is complete I can take the _rawData variable and run it through a WavWriter class, etc. However, if I run this same ByteArray as a sound using the following code which I adapted from the adobe cookbook: (within a class called WavPlayer) public function playSound(data:ByteArray):void { _wavData = data; _wavData.position = 0; _sound.addEventListener(SampleDataEvent.SAMPLE_DATA, _playSoundHandler); _channel = _sound.play(); _channel.addEventListener(Event.SOUND_COMPLETE, _onPlaybackComplete, false, 0, true); } private function _playSoundHandler(e:SampleDataEvent):void { if(_wavData.bytesAvailable <= 0) return; for(var i:int = 0; i < 8192; i++) { var sample:Number = 0; if(_wavData.bytesAvailable > 0) sample = _wavData.readFloat(); e.data.writeFloat(sample); } } The audio file plays at double speed! I checked recording bitrates and such and am pretty sure those are all correct, and I tried changing the buffer size and whatever other numbers I could think of. Could it be a mono vs stereo thing? Hope I was clear enough here, thanks!

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• Our Flash Streaming Player Occasionally Stutters like a Skipping CD after a Period of Time

  - by Jonathan Fritz

  We offer a streaming player for a number of our clients, who are responsible for their providing us with their own audio streams. We have written a very simple flash player that can play all of the streams that we support (icecast/shoutcast/live365/mp3 over http/etc). Unfortunately, we have found that when listening, our player sometimes begins to stutter (like a skipping cd), sometimes after only 10 minutes, and sometimes after an hour of listening. We have noticed this behaviour in firefox on both linux and windows. Does anybody know anything about this problem? We know that flash isn't ideal for infinite streams of audio, but it's about all that we can find that's on every platform out there. If anybody can suggest a solution to our problem, I'll be your friend forever. Here is a link to the live player: http://cr-jf.jfritz.02.dev.wecreate.com/streaming/player_v5/ Note that you'll need to test in a browser that isn't IE, because we use WMP in IE, and that the JavaScript on the page will cause the player to unload and re-load once an hour because of memory issues. Because I can only put one hyperlink in a post, I'll add a link to the player source code as a comment. Thanks all!

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• Where is /dev/dsp or /dev/audio?

  - by YumYumYum

  I have to apply sudo chmod a+r /dev/dsp or /dev/audio but in my Ubuntu 12.10 i do not have such. Where is then the PCM sound file for ssh? chmod: cannot access `/dev/dsp': No such file or directory chmod: cannot access `/dev/audio': No such file or directory Follow up: http://superuser.com/questions/244173/missing-dev-dsp-under-ubuntu I want to stream the sound output and input. So that i can capture any audio in/out to a file for recording.

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• AAC.js : le décodeur audio JavaScript open source supporte le profile Low Complexity

  AAC.js : le dernier décodeur audio JavaScript de Official.fm Labs qui supporte le profile Low Complexity [IMG]http://media.tumblr.com/tumblr_m6wpozHbxB1qbis4g.png[/IMG] L'équipe de Official.fm Labs vient de sortir un codec audio qui pourrait d'ailleurs être le prochain codec le plus utilisé après le MP3, voire le surpasser. AAC.js est entièrement codé en JavaScript avec le framework Aurora.js qui facilite l'écriture de codecs. AAC, qui signifie Advanced Audio Codec, est l'un des codecs les plus courants et des noms comm...

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• Amnesia doesn't start due to audio problems

  - by james

  I have a problem with amnesia game. After Intro and clicking continue button few times, when game is supposed to start it crashes. Here is console output: ALSA lib pcm_dmix.c:1018:(snd_pcm_dmix_open) unable to open slave ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side ALSA lib audio/pcm_bluetooth.c:1614:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib audio/pcm_bluetooth.c:1614:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib audio/pcm_bluetooth.c:1614:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib audio/pcm_bluetooth.c:1614:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib pcm_dmix.c:957:(snd_pcm_dmix_open) The dmix plugin supports only playback stream ALSA lib pcm_dmix.c:1018:(snd_pcm_dmix_open) unable to open slave Cannot connect to server socket err = No such file or directory Cannot connect to server socket jack server is not running or cannot be started I should mention I have integrated both graphic and sound card.

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• asus n550jv audio problem: no sound from notebook' speakers

  - by skywalker

  Ubuntu 13.10. The problem is: the internal speakers don't work. I have no problem when I'm using the headphones. There is no hardware issue since in windows 8 everything works perfectly(external subwoofer included). I'm trying to modify /etc/modprobe.d/alsa-base.conf but I can't find the correct model to put into: options snd-hda-intel model= The file HD-Audio-Models.txt doesn't contain the model for ALC668. Some info: :~sudo aplay -l **** List of PLAYBACK Hardware Devices **** card 0: MID [HDA Intel MID], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: MID [HDA Intel MID], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: MID [HDA Intel MID], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 0: ALC668 Analog [ALC668 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 :~$ sudo lspci -v | grep -A7 -i "audio" 00:03.0 Audio device: Intel Corporation Xeon E3-1200 v3/4th Gen Core Processor HD Audio Controller (rev 06) Subsystem: Intel Corporation Device 2010 Flags: bus master, fast devsel, latency 0, IRQ 52 Memory at f7a14000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit- Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 Kernel driver in use: snd_hda_intel -- 00:1b.0 Audio device: Intel Corporation 8 Series/C220 Series Chipset High Definition Audio Controller (rev 04) Subsystem: ASUSTeK Computer Inc. Device 11cd Flags: bus master, fast devsel, latency 0, IRQ 53 Memory at f7a10000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit+ Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 Capabilities: [100] Virtual Channel PS info :~$ amixer -c 0 Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958',1 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958',2 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] :~$ pacmd dump-volumes Welcome to PulseAudio! Use "help" for usage information. Sink 0: reference = 0: 76% 1: 76%, real = 0: 76% 1: 76%, soft = 0: 100% 1: 100%, current_hw = 0: 76% 1: 76%, save = yes Input 8: volume = 0: 100% 1: 100%, reference_ratio = 0: 100% 1: 100%, real_ratio = 0: 100% 1: 100%, soft = 0: 100% 1: 100%, volume_factor = 0: 100% 1: 100%, volume_factor_sink = 0: 100% 1: 100%, save = no Source 0: reference = 0: 100% 1: 100%, real = 0: 100% 1: 100%, soft = 0: 100% 1: 100%, current_hw = 0: 100% 1: 100%, save = no Source 1: reference = 0: 16% 1: 16%, real = 0: 16% 1: 16%, soft = 0: 100% 1: 100%, current_hw = 0: 16% 1: 16%, save = yes

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