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Add AudioContext.createConvolver(), createBiquadFilter() & OfflineAudioContext bindings
I am now doing quite a lot with WebAudio. The code base I was starting from is github.com/surikov/webaudiofont. It has constructed a node for reverb, which makes the sound seem bigger. (use the echo slider).
There does not appear to be bindings in exokit, but it is in the C code
The OfflineAudioContext is sort of nice to have, but it is also at the C level.
I might try to do this myself, but it you might make a few hints, it might be more doable.
Probably gonna start this right after thanksgiving. Making some notes for myself.
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The modulesio fork of LabSound is seriously older than where it originates from. From a bindings standpoint, can probably at least start with what is there, but there are probably fixes.
-
Probably should start by making a little table from the mozzilla docs of the API & the gaps. Something like:
Object / calls | In Exokit | In LabSound |
---|---|---|
Context | ||
createConvolver | No | Yes |
- A pass thru the issues shows these other audio related issues. Just noting for now https://github.com/exokitxr/exokit/issues/718 Panner node not connecting https://github.com/exokitxr/exokit/issues/678 PlaybackRate setValueAtTime ( need for sure ) https://github.com/exokitxr/exokit/issues/556 another offline support
Here are the various audio contexts. Do other objects in another comment.
BaseAudioContext | In Exokit | In Lab Sound | Notes |
---|---|---|---|
AudioWorklet | No | No | Do not need |
currentTime | Yes | Yes | |
destination | Yes | Yes | |
listener | Yes | Yes | |
sampleRate | Yes | Yes | |
state | No | No | Do not need |
onstatechange() | No | No | Do not need |
createAnalyser() | Yes | Yes | |
createBiquadFilter() | No | Yes | MUST HAVE |
createBuffer() | Yes | Yes | |
createBufferSource() | Yes | Yes | |
createConstantSource() | No | No | Do not need |
createChannelMerger() | No | Yes | Doing, but do not need. |
createChannelSplitter() | No | Yes | Doing, but do not need. |
createConvolver() | No | Yes | MUST HAVE |
createDelay() | No | Yes | Feed back loops might be nice |
createEmptyBuffer() | Yes | Yes | NOT IN API |
createDynamicsCompressor() | No | Yes | MUST HAVE |
createGain() | Yes | Yes | |
createIIRFilter() | No | No | Do not need |
createOscillator() | Yes | Yes | |
createPanner() | Yes | Yes | |
createPeriodicWave() | No | No | Do not need |
createScriptProcessor() | Yes | Yes | Deprecated in API |
createStereoPanner() | Yes | Yes | |
createWaveShaper() | No | Yes | Do not need |
decodeAudioData() | Yes | Yes |
AudioContext | In Exokit | In Lab Sound | Notes |
---|---|---|---|
baseLatency | No | No | Do not need |
outputLatency | No | No | Do not need |
close | Yes | Yes | Not in Offline |
createMediaElementSource() | Yes | Yes | |
createMediaStreamSource() | Yes | Yes | |
createMediaTrackSource() | Yes | Yes | |
getOutputTimestamp() | No | No | Do not need |
resume() | Yes | Yes | |
suspend() | Yes | Yes |
OfflineAudioContextt | In Exokit | In Lab Sound | Notes |
---|---|---|---|
/ length | No | Not Sure | calculate-able, but do not use |
/ startRendering | No | Yes | No point without having |
resume() | No | No | Do not need |
suspend() | No | No | Do not need |
Awesome, thanks for rounding up this info!
The modulesio fork of LabSound is seriously older than where it originates from. From a bindings standpoint, can probably at least start with what is there, but there are probably fixes.
I agree we could pull in from LabSound, and it should work. There are mostly just hardware bindings that are not upstream (I think the generic buffer-based binding for Android), and buffering that was used for the deprecated ScriptProcessorNode.
For most of the node types I mostly just created proxy objects for Javascript access which parallel the LabSound audio processing graph, without changing the LabSound code. The main difficulty in adding new node types is correct object linkage so that they are garbage collected at the right time.
Wonder if ASM is going to work on devices? Generally it should, though probably the best way to do decoding is to use whatever the platform is providing.
Started on BiquadFilter. Question, it has both regular AudioParams and something probably requiring a fake, type
. Can both be done at the same time? Here the labSound h file
// License: BSD 2 Clause
// Copyright (C) 2011, Google Inc. All rights reserved.
// Copyright (C) 2015+, The LabSound Authors. All rights reserved.
#ifndef BiquadFilterNode_h
#define BiquadFilterNode_h
#include "LabSound/core/AudioBasicProcessorNode.h"
namespace lab {
class AudioParam;
class BiquadProcessor;
class BiquadFilterNode : public AudioBasicProcessorNode
{
BiquadProcessor * biquadProcessor();
public:
enum
{
LOWPASS = 0,
HIGHPASS = 1,
BANDPASS = 2,
LOWSHELF = 3,
HIGHSHELF = 4,
PEAKING = 5,
NOTCH = 6,
ALLPASS = 7
};
BiquadFilterNode();
unsigned short type();
void setType(unsigned short type);
std::shared_ptr<AudioParam> frequency();
std::shared_ptr<AudioParam> q();
std::shared_ptr<AudioParam> gain();
std::shared_ptr<AudioParam> detune();
// Get the magnitude and phase response of the filter at the given
// set of frequencies (in Hz). The phase response is in radians.
void getFrequencyResponse(ContextRenderLock&, const std::vector<float>& frequencyHz, std::vector<float>& magResponse, std::vector<float>& phaseResponse);
};
} // namespace lab
#endif // BiquadFilterNode_h
Ok, I found an example of 2 type of these 'cons' in the Initialize()
of the obsolete ScriptProcessorNode
, so maybe not so obsolete after all.
Will pattern after that.
Now looking where LabSound might do decoding. They have it in extended. Am noticing that the source files here are very different in both call syntax and # of lines.
Your copy has 375 lines from 9/11/18.
The current from Lab Sound has 108 lines from 7/4/19 & different calls.
Think updating, your fork might be the next move.
I am using FFmpeg for the decoding; is there some reason to change that?
When I do a search of AudioContext.cpp, I do not get a hit for decodeAudioData
. I assumed that you were not doing any decoding. Maybe just implement that, but where are you doing the decoding?
I think any sort of Audio
node loading must be going through that path to load various file types on the Exokit side.
See this sound font I am using stores its data in js files for each instrument. There are multiple "files" within a font. Here is a piano. It uses that method.
I will look around elsewhere in the project for where this is.
Ok, it was not actually in C++, but in native-bindings.js. It works. Having a misunderstand & correcting is much preferred to having to write code!
Went to add Constant Source Node, but found it wasn't actually implemented by LabSound. Edited table above to reflect.