Hello again. I've been looking through the header of AudioScheduledSourceNode and found no mention of this event. When it ends, it ends, that's it.

However, there is a lot of (JS) code that expects onended calback to be called. For example in three.js it leads to the warning 'THREE.Audio: Audio is already playing.'.

If there actually is such an event, please point me in a right direction. And if there are no events in LabSound, then how those should be implemented?

Hi. I'm trying to fire-off a cmake on macOS and run immediately into cmake errors. Same goes when I try to fire-off cmake on Debian 9 or CentOS 7. I fire-off the cmake by issuing "cmake ." at the clone's root.

The problems I am seeing are:

On macOS:

1. The "clangdefaults.cmake" file is missing (work around: do a 'touch cmake/defaults/clangdefaults.cmake')
2. Get the error "error: /arch:AVX: No such file or directory" (work around: comment out the 'target_compile_options(${proj} PRIVATE /arch:AVX /Zi )' line from cmake/defaults/CXXhelpers)
3. With both these work-arounds in place, the makefiles get made and I issue a 'make' at the clone's root. I then get 42% into the build and get this compiler error:

[ 42%] Building C object CMakeFiles/libnyquist.dir/third_party/libnyquist/src/VorbisDependencies.c.o
In file included from /Users/jay/dev/repos/heynow/labsound/third_party/libnyquist/src/VorbisDependencies.c:62:
In file included from /Users/jay/dev/repos/heynow/labsound/third_party/libnyquist/third_party/libvorbis/src/vorbisenc.c:155:
/Users/jay/dev/repos/heynow/labsound/third_party/libnyquist/third_party/libvorbis/src/modes/setup_44.h:18:10: fatal error: 
      'modes/floor_all.h' file not found
#include "modes/floor_all.h"
         ^~~~~~~~~~~~~~~~~~~

Seems to me that the cmake didn't set things up fully for the third_party dependencies. What should I be doing differently here? Invoke cmake with some special arguments?

A similar set of problems when building on Linux occur:

1. The "gccdefaults" file is missing (work around: do a 'touch cmake/defaults/gccdefaults')
2. Get the error "error: /arch:AVX: No such file or directory" (work around: comment out the 'target_compile_options(${proj} PRIVATE /arch:AVX /Zi )' line from cmake/defaults/CXXhelpers)
3. With both of these work-arounds in place, the makefiles get made and I issue a 'make' at the clone's root. I then get 11% into the build and get this compiler error:

[ 12%] Building C object CMakeFiles/libopus.dir/third_party/libnyquist/third_party/opus/opusfile/src/info.c.o
In file included from /home/centos/LabSound/third_party/libnyquist/third_party/libogg/include/ogg/ogg.h:25:0,
                 from /home/centos/LabSound/third_party/libnyquist/third_party/opus/opusfile/include/opusfile.h:109,
                 from /home/centos/LabSound/third_party/libnyquist/third_party/opus/opusfile/src/include/internal.h:32,
                 from /home/centos/LabSound/third_party/libnyquist/third_party/opus/opusfile/src/info.c:16:
/home/centos/LabSound/third_party/libnyquist/third_party/libogg/include/ogg/os_types.h:143:32: fatal error: ogg/config_types.h: No such file or directory
 #  include <ogg/config_types.h>

Again, it seems like I'm missing something here using cmake. Or.. is it possible no one has recently done 'make' builds recently on this repository? Please advise, and, THANK YOU for your help.

/Jay

Hi there,

I've just cloned the master again and tried running the example (e_simple) but noticed how the sine wave is playing but the audio file is not.

I've also updated the library in a project that I am developing and have come across the exact same issue.

Here is the log produced by running the example project:

Toms-MacBook-Pro:bin tombutcher$ ./LabSoundExample 
11:28:03 TRACE /Users/tombutcher/Projects/xxxxx/lib/LabSound/src/extended/LabSound.cpp:33: MakeAudioDeviceList()
11:28:03 INFO  /Users/tombutcher/Projects/xxxxx/lib/LabSound/src/backends/RtAudio/AudioDevice_RtAudio.cpp:42: using rtaudio api macos_coreaudio
11:28:03 TRACE /Users/tombutcher/Projects/xxxxx/lib/LabSound/src/extended/LabSound.cpp:39: GetDefaultOutputAudioDeviceIndex()
11:28:03 TRACE /Users/tombutcher/Projects/xxxxx/lib/LabSound/src/extended/LabSound.cpp:45: GetDefaultInputAudioDeviceIndex()
11:28:03 TRACE /Users/tombutcher/Projects/xxxxx/lib/LabSound/src/extended/LabSound.cpp:51: MakeRealtimeAudioContext()
11:28:03 INFO  /Users/tombutcher/Projects/xxxxx/lib/LabSound/src/backends/RtAudio/AudioDevice_RtAudio.cpp:124: using output device idx: 1
11:28:03 INFO  /Users/tombutcher/Projects/xxxxx/lib/LabSound/src/backends/RtAudio/AudioDevice_RtAudio.cpp:125: using output device name: Apple Inc.: Built-in Output
11:28:03 INFO  /Users/tombutcher/Projects/xxxxx/lib/LabSound/src/backends/RtAudio/AudioDevice_RtAudio.cpp:130: using input device idx: -1
11:28:03 INFO  /Users/tombutcher/Projects/xxxxx/lib/LabSound/src/core/AudioHardwareDeviceNode.cpp:47: MakePlatformSpecificDevice() 
	* Sample Rate:     48000.000000 
	* Input Channels:  0 
	* Output Channels: 2   
11:28:03 TRACE /Users/tombutcher/Projects/xxxxx/lib/LabSound/src/core/AudioContext.cpp:287: Begin UpdateGraphThread

I've ensured that the samples folder copied from the assets directory is present:

Toms-MacBook-Pro:bin tombutcher$ ls
LabSoundExample	samples
Toms-MacBook-Pro:bin tombutcher$ 

The previous commit I used which worked was: https://github.com/LabSound/LabSound/commit/c0d778fef727432ad2280863e818280e6669c92e

Sorry for annoying you guys by constantly submitting issues. This is a really good library.

According to the spec such properties as AudioListener::positionX must be of type AudioParam with gettable/settable value.

In LabSound only direct void setPosition(float x, float y, float z) is available. And this is actually deprecated, because of reasons.

Is there any plan on converting those into AudioParams?

It seems that the samples do not play on Windows when no input device is available. If I enable/add an input device (stereo mix, mic etc.), the samples start playing.

Poking around the code it looks like AudioDestinationWin always tries to open the input device in AudioDestinationWin::configure: dac.openStream(&outputParams, &inputParams ... which fails and closes the stream. Later in AudioDestinationWin::start, dac.startStream fails and render is never called.

If I change AudioDestinationWin to not open or use input in render it fixes the issue, but it feels like a hack. Input should probably be optional on AudioDestination?

I'm trying to run Simple.h and I get this error:

Thomass-Air:bin thomasbutcher$ ./LabSoundExample [/Volumes/DataSSD/MultiWAV/src/mw4/MultiWAV/multiwav/lib/LabSound/src/core/SampledAudioNode.cpp @ 169] ASSERTION FAILED: isLengthGood [/Volumes/DataSSD/MultiWAV/src/mw4/MultiWAV/multiwav/lib/LabSound/src/core/OscillatorNode.cpp @ 213] ASSERTION FAILED: quantumFrameOffset <= framesToProcess Segmentation fault: 11

I'm trying to setup LabSound to build using gyp, and get it running on Windows, because I think it's a cool project. Unfortunately, it seems like a lot of the sources are looking for WebCore/WebCoreHeaderDetection.h, which is not in the repository. Am I configuring something incorrectly, or do you have some external dependencies that I can setup on my machine.

I can't schedule the playing time precisely

The time, in seconds, at which the sound should begin to play, in the same time coordinate system used by the AudioContext. https://developer.mozilla.org/en-US/docs/Web/API/AudioBufferSourceNode/start

So, speaking of WAA, they have this kind of interface:

interface BaseAudioContext : EventTarget {
...
    AudioBuffer            createBuffer (unsigned long numberOfChannels, unsigned long length, float sampleRate);
    Promise<AudioBuffer>   decodeAudioData (ArrayBuffer audioData, optional DecodeSuccessCallback successCallback, optional DecodeErrorCallback errorCallback);

And then those guys in threejs use it:

	Object.assign( AudioLoader.prototype, {

		load: function ( url, onLoad, onProgress, onError ) {

			var loader = new FileLoader( this.manager );
			loader.setResponseType( 'arraybuffer' );
			loader.load( url, function ( buffer ) {

				var context = AudioContext.getContext();

				context.decodeAudioData( buffer, function ( audioBuffer ) {

					onLoad( audioBuffer );

				} );

			}, onProgress, onError );

		}

	} );

Well, that boders me a lot, since I'm currently investigating ways to port LabSound onto Node.js. But as LabSound does not comply with WAA standard so far, I would have to patch it first. This is why I ask if there is any meaningful explanation for 934b6bfc9b129ceafcd7f7f38c3cf8f218a2f36e ? Probably there were some issues with this class, or it was just a matter of over-optimization?

Hello, I was trying to use LabSound in Android. I was encouraged when I saw the following in the README.md:

In the past, LabSound has been demonstrated to work on iOS, Android, and Linux via JACK.

However I'm quite new to crosscompilation and I must admit that I'm a little bit lost in the field. As extra information, I'm compiling from a 86x64 Linux (Ubuntu 18.04 distribution). Any advice or hint on how to compile this project for Android would be useful.

Thank you so much.

One thing I notice in this implementation is that you seem to have left in merely the Web Audio objects without any reference to a DOM or to the v8 JavaScript engine. When I considered doing a similar project, my original thinking was that it might have been handy to have the JavaScript core floating about as well, to permit high-level scripting and interconnection of elements, and also hopefully to facilitate development of editors and the like for the object interconnect formats. What are your thoughts on development and representation of exchangeable node interconnect formats for audio engineers who don't know scripting or C++?

Besides left-to-right, cone and doppler effects, Panner has a distance gain set by the difference between listener and panner positions, and with fade effects controlled by a few parameters, here's a modification to the ex_hrtf_spatialization example: ... panner->setDistanceModel(lab::PannerNode::LINEAR_DISTANCE);
panner->setRolloffFactor(10.0f); panner->setRefDistance(.1f); panner->setMaxDistance(20.0f); ... and to elliminate left-right motion effects and focus on distance gain panner->setPosition(.1f, x, 0.1f); x but I don't hear a volume / gain change x and std::cout << " " << panner->param("distanceGain").get()->value(); x shows all 1 on every slice

goal, to get forward-back motion change of volume like this web audio boom-box example: https://developer.mozilla.org/en-US/docs/Web/API/Web_Audio_API/Web_audio_spatialization_basics

Any hypotheses?

more.. I work on freewrl.sf.net which renders www.web3d.org 3D standard scene files, and they updated their standard recently to include web audio equivalent nodes. freewrl is flat C, and with a .cpp module I can interface to and have a bit of the labsound working, but not the distance gain. Rather than separate threads with Wait(), I visit the labsound nodes on each 3D scene rendering frame, and reset the parameters such as location.

The AudioParam class has a field m_units that doesn't correspond to anything in the Web Audio API specification and is not documented anywhere in LabSound. I would guess that it is meant to specify what unit is used for the parameter (hertz, decibels, etc.) but it is only an unsigned integer and I don't see a data structure that maps that integer to a name.

(the curve that's showing up is because I captured an image with Brown instead of White noise)

Not a LabSound issue... Here's the canonical WebAudio response curves; selecting the shelf variants, there doesn't seem to be any way to get a curve out of it, at least on Safari:

https://rtoy.github.io/webaudio-hacks/more/biquad/biquad-coef.html

The discussion here - https://github.com/WebAudio/web-audio-api-v2/issues/67 - seems to suggest that the behavior of the shelf modes is unsatisfactory.

So the shelves seem to be broken on WebAudio/WebKit as well. I think we either need proper shelving filters added as a switch inside the BiquadFilter, or we need a new node with a good implementation. Unless if I'm missing something obvious, and there's a parameter to set on the shelves to make it do .... anything at all

I found a possible way to run the "ma_device_init" method again.

I don't know how to get device change events via C++, so I have to trigger them manually for now.

https://github.com/xioxin/LabSound/commit/7d0b8deb47d2a268b87bde9cdcb1729baa321638

Is there a better solution?

Documentation is here - https://developer.mozilla.org/en-US/docs/Web/API/AudioContext/close

In summary -

Close releases any system audio resources that it uses. It is asynchronous, returning a promise object that can wait() until all AudioContext-creation-blocking resources have been released

Closed contexts cannot have new nodes created, but can decode audio data, create buffers, etc. It will forcibly release any system audio resources that might prevent additional AudioContexts from being created and used, suspend the progression of audio time in the audio context, and stop processing audio data.