src/audio/emscripten/SDL_emscriptenaudio.c - external/github.com/libsdl-org/SDL - Git at Google

 /*
   Simple DirectMedia Layer
   Copyright (C) 1997-2021 Sam Lantinga <slouken@libsdl.org>

   This software is provided 'as-is', without any express or implied
   warranty.  In no event will the authors be held liable for any damages
   arising from the use of this software.

   Permission is granted to anyone to use this software for any purpose,
   including commercial applications, and to alter it and redistribute it
   freely, subject to the following restrictions:

   1. The origin of this software must not be misrepresented; you must not
      claim that you wrote the original software. If you use this software
      in a product, an acknowledgment in the product documentation would be
      appreciated but is not required.
   2. Altered source versions must be plainly marked as such, and must not be
      misrepresented as being the original software.
   3. This notice may not be removed or altered from any source distribution.
 */
 #include "../../SDL_internal.h"

 #if SDL_AUDIO_DRIVER_EMSCRIPTEN

 #include "SDL_audio.h"
 #include "../SDL_audio_c.h"
 #include "SDL_emscriptenaudio.h"

 #include <emscripten/emscripten.h>

 static void
 FeedAudioDevice(_THIS, const void *buf, const int buflen)
 {
     const int framelen = (SDL_AUDIO_BITSIZE(this->spec.format) / 8) * this->spec.channels;
     EM_ASM_ARGS({
         var SDL2 = Module['SDL2'];
         var numChannels = SDL2.audio.currentOutputBuffer['numberOfChannels'];
         for (var c = 0; c < numChannels; ++c) {
             var channelData = SDL2.audio.currentOutputBuffer['getChannelData'](c);
             if (channelData.length != $1) {
                 throw 'Web Audio output buffer length mismatch! Destination size: ' + channelData.length + ' samples vs expected ' + $1 + ' samples!';
             }

             for (var j = 0; j < $1; ++j) {
                 channelData[j] = HEAPF32[$0 + ((j*numChannels + c) << 2) >> 2];  /* !!! FIXME: why are these shifts here? */
             }
         }
     }, buf, buflen / framelen);
 }

 static void
 HandleAudioProcess(_THIS)
 {
     SDL_AudioCallback callback = this->callbackspec.callback;
     const int stream_len = this->callbackspec.size;

     /* Only do something if audio is enabled */
     if (!SDL_AtomicGet(&this->enabled) || SDL_AtomicGet(&this->paused)) {
         if (this->stream) {
             SDL_AudioStreamClear(this->stream);
         }
         return;
     }

     if (this->stream == NULL) {  /* no conversion necessary. */
         SDL_assert(this->spec.size == stream_len);
         callback(this->callbackspec.userdata, this->work_buffer, stream_len);
     } else {  /* streaming/converting */
         int got;
         while (SDL_AudioStreamAvailable(this->stream) < ((int) this->spec.size)) {
             callback(this->callbackspec.userdata, this->work_buffer, stream_len);
             if (SDL_AudioStreamPut(this->stream, this->work_buffer, stream_len) == -1) {
                 SDL_AudioStreamClear(this->stream);
                 SDL_AtomicSet(&this->enabled, 0);
                 break;
             }
         }

         got = SDL_AudioStreamGet(this->stream, this->work_buffer, this->spec.size);
         SDL_assert((got < 0) || (got == this->spec.size));
         if (got != this->spec.size) {
             SDL_memset(this->work_buffer, this->spec.silence, this->spec.size);
         }
     }

     FeedAudioDevice(this, this->work_buffer, this->spec.size);
 }

 static void
 HandleCaptureProcess(_THIS)
 {
     SDL_AudioCallback callback = this->callbackspec.callback;
     const int stream_len = this->callbackspec.size;

     /* Only do something if audio is enabled */
     if (!SDL_AtomicGet(&this->enabled) || SDL_AtomicGet(&this->paused)) {
         SDL_AudioStreamClear(this->stream);
         return;
     }

     EM_ASM_ARGS({
         var SDL2 = Module['SDL2'];
         var numChannels = SDL2.capture.currentCaptureBuffer.numberOfChannels;
         for (var c = 0; c < numChannels; ++c) {
             var channelData = SDL2.capture.currentCaptureBuffer.getChannelData(c);
             if (channelData.length != $1) {
                 throw 'Web Audio capture buffer length mismatch! Destination size: ' + channelData.length + ' samples vs expected ' + $1 + ' samples!';
             }

             if (numChannels == 1) {  /* fastpath this a little for the common (mono) case. */
                 for (var j = 0; j < $1; ++j) {
                     setValue($0 + (j * 4), channelData[j], 'float');
                 }
             } else {
                 for (var j = 0; j < $1; ++j) {
                     setValue($0 + (((j * numChannels) + c) * 4), channelData[j], 'float');
                 }
             }
         }
     }, this->work_buffer, (this->spec.size / sizeof (float)) / this->spec.channels);

     /* okay, we've got an interleaved float32 array in C now. */

     if (this->stream == NULL) {  /* no conversion necessary. */
         SDL_assert(this->spec.size == stream_len);
         callback(this->callbackspec.userdata, this->work_buffer, stream_len);
     } else {  /* streaming/converting */
         if (SDL_AudioStreamPut(this->stream, this->work_buffer, this->spec.size) == -1) {
             SDL_AtomicSet(&this->enabled, 0);
         }

         while (SDL_AudioStreamAvailable(this->stream) >= stream_len) {
             const int got = SDL_AudioStreamGet(this->stream, this->work_buffer, stream_len);
             SDL_assert((got < 0) || (got == stream_len));
             if (got != stream_len) {
                 SDL_memset(this->work_buffer, this->callbackspec.silence, stream_len);
             }
             callback(this->callbackspec.userdata, this->work_buffer, stream_len);  /* Send it to the app. */
         }
     }
 }


 static void
 EMSCRIPTENAUDIO_CloseDevice(_THIS)
 {
     EM_ASM_({
         var SDL2 = Module['SDL2'];
         if ($0) {
             if (SDL2.capture.silenceTimer !== undefined) {
                 clearTimeout(SDL2.capture.silenceTimer);
             }
             if (SDL2.capture.stream !== undefined) {
                 var tracks = SDL2.capture.stream.getAudioTracks();
                 for (var i = 0; i < tracks.length; i++) {
                     SDL2.capture.stream.removeTrack(tracks[i]);
                 }
                 SDL2.capture.stream = undefined;
             }
             if (SDL2.capture.scriptProcessorNode !== undefined) {
                 SDL2.capture.scriptProcessorNode.onaudioprocess = function(audioProcessingEvent) {};
                 SDL2.capture.scriptProcessorNode.disconnect();
                 SDL2.capture.scriptProcessorNode = undefined;
             }
             if (SDL2.capture.mediaStreamNode !== undefined) {
                 SDL2.capture.mediaStreamNode.disconnect();
                 SDL2.capture.mediaStreamNode = undefined;
             }
             if (SDL2.capture.silenceBuffer !== undefined) {
                 SDL2.capture.silenceBuffer = undefined
             }
             SDL2.capture = undefined;
         } else {
             if (SDL2.audio.scriptProcessorNode != undefined) {
                 SDL2.audio.scriptProcessorNode.disconnect();
                 SDL2.audio.scriptProcessorNode = undefined;
             }
             SDL2.audio = undefined;
         }
         if ((SDL2.audioContext !== undefined) && (SDL2.audio === undefined) && (SDL2.capture === undefined)) {
             SDL2.audioContext.close();
             SDL2.audioContext = undefined;
         }
     }, this->iscapture);

 #if 0  /* !!! FIXME: currently not used. Can we move some stuff off the SDL2 namespace? --ryan. */
     SDL_free(this->hidden);
 #endif
 }

 static int
 EMSCRIPTENAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
 {
     SDL_bool valid_format = SDL_FALSE;
     SDL_AudioFormat test_format;
     int result;

     /* based on parts of library_sdl.js */

     /* create context */
     result = EM_ASM_INT({
         if(typeof(Module['SDL2']) === 'undefined') {
             Module['SDL2'] = {};
         }
         var SDL2 = Module['SDL2'];
         if (!$0) {
             SDL2.audio = {};
         } else {
             SDL2.capture = {};
         }

         if (!SDL2.audioContext) {
             if (typeof(AudioContext) !== 'undefined') {
                 SDL2.audioContext = new AudioContext();
             } else if (typeof(webkitAudioContext) !== 'undefined') {
                 SDL2.audioContext = new webkitAudioContext();
             }
             if (SDL2.audioContext) {
                 autoResumeAudioContext(SDL2.audioContext);
             }
         }
         return SDL2.audioContext === undefined ? -1 : 0;
     }, iscapture);
     if (result < 0) {
         return SDL_SetError("Web Audio API is not available!");
     }

     test_format = SDL_FirstAudioFormat(this->spec.format);
     while ((!valid_format) && (test_format)) {
         switch (test_format) {
         case AUDIO_F32: /* web audio only supports floats */
             this->spec.format = test_format;

             valid_format = SDL_TRUE;
             break;
         }
         test_format = SDL_NextAudioFormat();
     }

     if (!valid_format) {
         /* Didn't find a compatible format :( */
         return SDL_SetError("No compatible audio format!");
     }

     /* Initialize all variables that we clean on shutdown */
 #if 0  /* !!! FIXME: currently not used. Can we move some stuff off the SDL2 namespace? --ryan. */
     this->hidden = (struct SDL_PrivateAudioData *)
         SDL_malloc((sizeof *this->hidden));
     if (this->hidden == NULL) {
         return SDL_OutOfMemory();
     }
     SDL_zerop(this->hidden);
 #endif
     this->hidden = (struct SDL_PrivateAudioData *)0x1;

     /* limit to native freq */
     this->spec.freq = EM_ASM_INT_V({
       var SDL2 = Module['SDL2'];
       return SDL2.audioContext.sampleRate;
     });

     SDL_CalculateAudioSpec(&this->spec);

     if (iscapture) {
         /* The idea is to take the capture media stream, hook it up to an
            audio graph where we can pass it through a ScriptProcessorNode
            to access the raw PCM samples and push them to the SDL app's
            callback. From there, we "process" the audio data into silence
            and forget about it. */

         /* This should, strictly speaking, use MediaRecorder for capture, but
            this API is cleaner to use and better supported, and fires a
            callback whenever there's enough data to fire down into the app.
            The downside is that we are spending CPU time silencing a buffer
            that the audiocontext uselessly mixes into any output. On the
            upside, both of those things are not only run in native code in
            the browser, they're probably SIMD code, too. MediaRecorder
            feels like it's a pretty inefficient tapdance in similar ways,
            to be honest. */

         EM_ASM_({
             var SDL2 = Module['SDL2'];
             var have_microphone = function(stream) {
                 //console.log('SDL audio capture: we have a microphone! Replacing silence callback.');
                 if (SDL2.capture.silenceTimer !== undefined) {
                     clearTimeout(SDL2.capture.silenceTimer);
                     SDL2.capture.silenceTimer = undefined;
                 }
                 SDL2.capture.mediaStreamNode = SDL2.audioContext.createMediaStreamSource(stream);
                 SDL2.capture.scriptProcessorNode = SDL2.audioContext.createScriptProcessor($1, $0, 1);
                 SDL2.capture.scriptProcessorNode.onaudioprocess = function(audioProcessingEvent) {
                     if ((SDL2 === undefined) || (SDL2.capture === undefined)) { return; }
                     audioProcessingEvent.outputBuffer.getChannelData(0).fill(0.0);
                     SDL2.capture.currentCaptureBuffer = audioProcessingEvent.inputBuffer;
                     dynCall('vi', $2, [$3]);
                 };
                 SDL2.capture.mediaStreamNode.connect(SDL2.capture.scriptProcessorNode);
                 SDL2.capture.scriptProcessorNode.connect(SDL2.audioContext.destination);
                 SDL2.capture.stream = stream;
             };

             var no_microphone = function(error) {
                 //console.log('SDL audio capture: we DO NOT have a microphone! (' + error.name + ')...leaving silence callback running.');
             };

             /* we write silence to the audio callback until the microphone is available (user approves use, etc). */
             SDL2.capture.silenceBuffer = SDL2.audioContext.createBuffer($0, $1, SDL2.audioContext.sampleRate);
             SDL2.capture.silenceBuffer.getChannelData(0).fill(0.0);
             var silence_callback = function() {
                 SDL2.capture.currentCaptureBuffer = SDL2.capture.silenceBuffer;
                 dynCall('vi', $2, [$3]);
             };

             SDL2.capture.silenceTimer = setTimeout(silence_callback, ($1 / SDL2.audioContext.sampleRate) * 1000);

             if ((navigator.mediaDevices !== undefined) && (navigator.mediaDevices.getUserMedia !== undefined)) {
                 navigator.mediaDevices.getUserMedia({ audio: true, video: false }).then(have_microphone).catch(no_microphone);
             } else if (navigator.webkitGetUserMedia !== undefined) {
                 navigator.webkitGetUserMedia({ audio: true, video: false }, have_microphone, no_microphone);
             }
         }, this->spec.channels, this->spec.samples, HandleCaptureProcess, this);
     } else {
         /* setup a ScriptProcessorNode */
         EM_ASM_ARGS({
             var SDL2 = Module['SDL2'];
             SDL2.audio.scriptProcessorNode = SDL2.audioContext['createScriptProcessor']($1, 0, $0);
             SDL2.audio.scriptProcessorNode['onaudioprocess'] = function (e) {
                 if ((SDL2 === undefined) || (SDL2.audio === undefined)) { return; }
                 SDL2.audio.currentOutputBuffer = e['outputBuffer'];
                 dynCall('vi', $2, [$3]);
             };
             SDL2.audio.scriptProcessorNode['connect'](SDL2.audioContext['destination']);
         }, this->spec.channels, this->spec.samples, HandleAudioProcess, this);
     }

     return 0;
 }

 static int
 EMSCRIPTENAUDIO_Init(SDL_AudioDriverImpl * impl)
 {
     int available;
     int capture_available;

     /* Set the function pointers */
     impl->OpenDevice = EMSCRIPTENAUDIO_OpenDevice;
     impl->CloseDevice = EMSCRIPTENAUDIO_CloseDevice;

     impl->OnlyHasDefaultOutputDevice = 1;

     /* no threads here */
     impl->SkipMixerLock = 1;
     impl->ProvidesOwnCallbackThread = 1;

     /* check availability */
     available = EM_ASM_INT_V({
         if (typeof(AudioContext) !== 'undefined') {
             return 1;
         } else if (typeof(webkitAudioContext) !== 'undefined') {
             return 1;
         }
         return 0;
     });

     if (!available) {
         SDL_SetError("No audio context available");
     }

     capture_available = available && EM_ASM_INT_V({
         if ((typeof(navigator.mediaDevices) !== 'undefined') && (typeof(navigator.mediaDevices.getUserMedia) !== 'undefined')) {
             return 1;
         } else if (typeof(navigator.webkitGetUserMedia) !== 'undefined') {
             return 1;
         }
         return 0;
     });

     impl->HasCaptureSupport = capture_available ? SDL_TRUE : SDL_FALSE;
     impl->OnlyHasDefaultCaptureDevice = capture_available ? SDL_TRUE : SDL_FALSE;

     return available;
 }

 AudioBootStrap EMSCRIPTENAUDIO_bootstrap = {
     "emscripten", "SDL emscripten audio driver", EMSCRIPTENAUDIO_Init, 0
 };

 #endif /* SDL_AUDIO_DRIVER_EMSCRIPTEN */

 /* vi: set ts=4 sw=4 expandtab: */
	/*
	Simple DirectMedia Layer
	Copyright (C) 1997-2021 Sam Lantinga <slouken@libsdl.org>

	This software is provided 'as-is', without any express or implied
	warranty. In no event will the authors be held liable for any damages
	arising from the use of this software.

	Permission is granted to anyone to use this software for any purpose,
	including commercial applications, and to alter it and redistribute it
	freely, subject to the following restrictions:

	1. The origin of this software must not be misrepresented; you must not
	claim that you wrote the original software. If you use this software
	in a product, an acknowledgment in the product documentation would be
	appreciated but is not required.
	2. Altered source versions must be plainly marked as such, and must not be
	misrepresented as being the original software.
	3. This notice may not be removed or altered from any source distribution.
	*/
	#include "../../SDL_internal.h"

	#if SDL_AUDIO_DRIVER_EMSCRIPTEN

	#include "SDL_audio.h"
	#include "../SDL_audio_c.h"
	#include "SDL_emscriptenaudio.h"

	#include <emscripten/emscripten.h>

	static void
	FeedAudioDevice(_THIS, const void *buf, const int buflen)
	{
	const int framelen = (SDL_AUDIO_BITSIZE(this->spec.format) / 8) * this->spec.channels;
	EM_ASM_ARGS({
	var SDL2 = Module['SDL2'];
	var numChannels = SDL2.audio.currentOutputBuffer['numberOfChannels'];
	for (var c = 0; c < numChannels; ++c) {
	var channelData = SDL2.audio.currentOutputBuffer['getChannelData'](c);
	if (channelData.length != $1) {
	throw 'Web Audio output buffer length mismatch! Destination size: ' + channelData.length + ' samples vs expected ' + $1 + ' samples!';
	}

	for (var j = 0; j < $1; ++j) {
	channelData[j] = HEAPF32[$0 + ((jnumChannels + c) << 2) >> 2]; / !!! FIXME: why are these shifts here? */
	}
	}
	}, buf, buflen / framelen);
	}

	static void
	HandleAudioProcess(_THIS)
	{
	SDL_AudioCallback callback = this->callbackspec.callback;
	const int stream_len = this->callbackspec.size;

	/* Only do something if audio is enabled */
	if (!SDL_AtomicGet(&this->enabled) \|\| SDL_AtomicGet(&this->paused)) {
	if (this->stream) {
	SDL_AudioStreamClear(this->stream);
	}
	return;
	}

	if (this->stream == NULL) { /* no conversion necessary. */
	SDL_assert(this->spec.size == stream_len);
	callback(this->callbackspec.userdata, this->work_buffer, stream_len);
	} else { /* streaming/converting */
	int got;
	while (SDL_AudioStreamAvailable(this->stream) < ((int) this->spec.size)) {
	callback(this->callbackspec.userdata, this->work_buffer, stream_len);
	if (SDL_AudioStreamPut(this->stream, this->work_buffer, stream_len) == -1) {
	SDL_AudioStreamClear(this->stream);
	SDL_AtomicSet(&this->enabled, 0);
	break;
	}
	}

	got = SDL_AudioStreamGet(this->stream, this->work_buffer, this->spec.size);
	SDL_assert((got < 0) \|\| (got == this->spec.size));
	if (got != this->spec.size) {
	SDL_memset(this->work_buffer, this->spec.silence, this->spec.size);
	}
	}

	FeedAudioDevice(this, this->work_buffer, this->spec.size);
	}

	static void
	HandleCaptureProcess(_THIS)
	{
	SDL_AudioCallback callback = this->callbackspec.callback;
	const int stream_len = this->callbackspec.size;

	/* Only do something if audio is enabled */
	if (!SDL_AtomicGet(&this->enabled) \|\| SDL_AtomicGet(&this->paused)) {
	SDL_AudioStreamClear(this->stream);
	return;
	}

	EM_ASM_ARGS({
	var SDL2 = Module['SDL2'];
	var numChannels = SDL2.capture.currentCaptureBuffer.numberOfChannels;
	for (var c = 0; c < numChannels; ++c) {
	var channelData = SDL2.capture.currentCaptureBuffer.getChannelData(c);
	if (channelData.length != $1) {
	throw 'Web Audio capture buffer length mismatch! Destination size: ' + channelData.length + ' samples vs expected ' + $1 + ' samples!';
	}

	if (numChannels == 1) { /* fastpath this a little for the common (mono) case. */
	for (var j = 0; j < $1; ++j) {
	setValue($0 + (j * 4), channelData[j], 'float');
	}
	} else {
	for (var j = 0; j < $1; ++j) {
	setValue($0 + (((j * numChannels) + c) * 4), channelData[j], 'float');
	}
	}
	}
	}, this->work_buffer, (this->spec.size / sizeof (float)) / this->spec.channels);

	/* okay, we've got an interleaved float32 array in C now. */

	if (this->stream == NULL) { /* no conversion necessary. */
	SDL_assert(this->spec.size == stream_len);
	callback(this->callbackspec.userdata, this->work_buffer, stream_len);
	} else { /* streaming/converting */
	if (SDL_AudioStreamPut(this->stream, this->work_buffer, this->spec.size) == -1) {
	SDL_AtomicSet(&this->enabled, 0);
	}

	while (SDL_AudioStreamAvailable(this->stream) >= stream_len) {
	const int got = SDL_AudioStreamGet(this->stream, this->work_buffer, stream_len);
	SDL_assert((got < 0) \|\| (got == stream_len));
	if (got != stream_len) {
	SDL_memset(this->work_buffer, this->callbackspec.silence, stream_len);
	}
	callback(this->callbackspec.userdata, this->work_buffer, stream_len); /* Send it to the app. */
	}
	}
	}


	static void
	EMSCRIPTENAUDIO_CloseDevice(_THIS)
	{
	EM_ASM_({
	var SDL2 = Module['SDL2'];
	if ($0) {
	if (SDL2.capture.silenceTimer !== undefined) {
	clearTimeout(SDL2.capture.silenceTimer);
	}
	if (SDL2.capture.stream !== undefined) {
	var tracks = SDL2.capture.stream.getAudioTracks();
	for (var i = 0; i < tracks.length; i++) {
	SDL2.capture.stream.removeTrack(tracks[i]);
	}
	SDL2.capture.stream = undefined;
	}
	if (SDL2.capture.scriptProcessorNode !== undefined) {
	SDL2.capture.scriptProcessorNode.onaudioprocess = function(audioProcessingEvent) {};
	SDL2.capture.scriptProcessorNode.disconnect();
	SDL2.capture.scriptProcessorNode = undefined;
	}
	if (SDL2.capture.mediaStreamNode !== undefined) {
	SDL2.capture.mediaStreamNode.disconnect();
	SDL2.capture.mediaStreamNode = undefined;
	}
	if (SDL2.capture.silenceBuffer !== undefined) {
	SDL2.capture.silenceBuffer = undefined
	}
	SDL2.capture = undefined;
	} else {
	if (SDL2.audio.scriptProcessorNode != undefined) {
	SDL2.audio.scriptProcessorNode.disconnect();
	SDL2.audio.scriptProcessorNode = undefined;
	}
	SDL2.audio = undefined;
	}
	if ((SDL2.audioContext !== undefined) && (SDL2.audio === undefined) && (SDL2.capture === undefined)) {
	SDL2.audioContext.close();
	SDL2.audioContext = undefined;
	}
	}, this->iscapture);

	#if 0 /* !!! FIXME: currently not used. Can we move some stuff off the SDL2 namespace? --ryan. */
	SDL_free(this->hidden);
	#endif
	}

	static int
	EMSCRIPTENAUDIO_OpenDevice(_THIS, void handle, const char devname, int iscapture)
	{
	SDL_bool valid_format = SDL_FALSE;
	SDL_AudioFormat test_format;
	int result;

	/* based on parts of library_sdl.js */

	/* create context */
	result = EM_ASM_INT({
	if(typeof(Module['SDL2']) === 'undefined') {
	Module['SDL2'] = {};
	}
	var SDL2 = Module['SDL2'];
	if (!$0) {
	SDL2.audio = {};
	} else {
	SDL2.capture = {};
	}

	if (!SDL2.audioContext) {
	if (typeof(AudioContext) !== 'undefined') {
	SDL2.audioContext = new AudioContext();
	} else if (typeof(webkitAudioContext) !== 'undefined') {
	SDL2.audioContext = new webkitAudioContext();
	}
	if (SDL2.audioContext) {
	autoResumeAudioContext(SDL2.audioContext);
	}
	}
	return SDL2.audioContext === undefined ? -1 : 0;
	}, iscapture);
	if (result < 0) {
	return SDL_SetError("Web Audio API is not available!");
	}

	test_format = SDL_FirstAudioFormat(this->spec.format);
	while ((!valid_format) && (test_format)) {
	switch (test_format) {
	case AUDIO_F32: /* web audio only supports floats */
	this->spec.format = test_format;

	valid_format = SDL_TRUE;
	break;
	}
	test_format = SDL_NextAudioFormat();
	}

	if (!valid_format) {
	/* Didn't find a compatible format :( */
	return SDL_SetError("No compatible audio format!");
	}

	/* Initialize all variables that we clean on shutdown */
	#if 0 /* !!! FIXME: currently not used. Can we move some stuff off the SDL2 namespace? --ryan. */
	this->hidden = (struct SDL_PrivateAudioData *)
	SDL_malloc((sizeof *this->hidden));
	if (this->hidden == NULL) {
	return SDL_OutOfMemory();
	}
	SDL_zerop(this->hidden);
	#endif
	this->hidden = (struct SDL_PrivateAudioData *)0x1;

	/* limit to native freq */
	this->spec.freq = EM_ASM_INT_V({
	var SDL2 = Module['SDL2'];
	return SDL2.audioContext.sampleRate;
	});

	SDL_CalculateAudioSpec(&this->spec);

	if (iscapture) {
	/* The idea is to take the capture media stream, hook it up to an
	audio graph where we can pass it through a ScriptProcessorNode
	to access the raw PCM samples and push them to the SDL app's
	callback. From there, we "process" the audio data into silence
	and forget about it. */

	/* This should, strictly speaking, use MediaRecorder for capture, but
	this API is cleaner to use and better supported, and fires a
	callback whenever there's enough data to fire down into the app.
	The downside is that we are spending CPU time silencing a buffer
	that the audiocontext uselessly mixes into any output. On the
	upside, both of those things are not only run in native code in
	the browser, they're probably SIMD code, too. MediaRecorder
	feels like it's a pretty inefficient tapdance in similar ways,
	to be honest. */

	EM_ASM_({
	var SDL2 = Module['SDL2'];
	var have_microphone = function(stream) {
	//console.log('SDL audio capture: we have a microphone! Replacing silence callback.');
	if (SDL2.capture.silenceTimer !== undefined) {
	clearTimeout(SDL2.capture.silenceTimer);
	SDL2.capture.silenceTimer = undefined;
	}
	SDL2.capture.mediaStreamNode = SDL2.audioContext.createMediaStreamSource(stream);
	SDL2.capture.scriptProcessorNode = SDL2.audioContext.createScriptProcessor($1, $0, 1);
	SDL2.capture.scriptProcessorNode.onaudioprocess = function(audioProcessingEvent) {
	if ((SDL2 === undefined) \|\| (SDL2.capture === undefined)) { return; }
	audioProcessingEvent.outputBuffer.getChannelData(0).fill(0.0);
	SDL2.capture.currentCaptureBuffer = audioProcessingEvent.inputBuffer;
	dynCall('vi', $2, [$3]);
	};
	SDL2.capture.mediaStreamNode.connect(SDL2.capture.scriptProcessorNode);
	SDL2.capture.scriptProcessorNode.connect(SDL2.audioContext.destination);
	SDL2.capture.stream = stream;
	};

	var no_microphone = function(error) {
	//console.log('SDL audio capture: we DO NOT have a microphone! (' + error.name + ')...leaving silence callback running.');
	};

	/* we write silence to the audio callback until the microphone is available (user approves use, etc). */
	SDL2.capture.silenceBuffer = SDL2.audioContext.createBuffer($0, $1, SDL2.audioContext.sampleRate);
	SDL2.capture.silenceBuffer.getChannelData(0).fill(0.0);
	var silence_callback = function() {
	SDL2.capture.currentCaptureBuffer = SDL2.capture.silenceBuffer;
	dynCall('vi', $2, [$3]);
	};

	SDL2.capture.silenceTimer = setTimeout(silence_callback, ($1 / SDL2.audioContext.sampleRate) * 1000);

	if ((navigator.mediaDevices !== undefined) && (navigator.mediaDevices.getUserMedia !== undefined)) {
	navigator.mediaDevices.getUserMedia({ audio: true, video: false }).then(have_microphone).catch(no_microphone);
	} else if (navigator.webkitGetUserMedia !== undefined) {
	navigator.webkitGetUserMedia({ audio: true, video: false }, have_microphone, no_microphone);
	}
	}, this->spec.channels, this->spec.samples, HandleCaptureProcess, this);
	} else {
	/* setup a ScriptProcessorNode */
	EM_ASM_ARGS({
	var SDL2 = Module['SDL2'];
	SDL2.audio.scriptProcessorNode = SDL2.audioContext['createScriptProcessor']($1, 0, $0);
	SDL2.audio.scriptProcessorNode['onaudioprocess'] = function (e) {
	if ((SDL2 === undefined) \|\| (SDL2.audio === undefined)) { return; }
	SDL2.audio.currentOutputBuffer = e['outputBuffer'];
	dynCall('vi', $2, [$3]);
	};
	SDL2.audio.scriptProcessorNode['connect'](SDL2.audioContext['destination']);
	}, this->spec.channels, this->spec.samples, HandleAudioProcess, this);
	}

	return 0;
	}

	static int
	EMSCRIPTENAUDIO_Init(SDL_AudioDriverImpl * impl)
	{
	int available;
	int capture_available;

	/* Set the function pointers */
	impl->OpenDevice = EMSCRIPTENAUDIO_OpenDevice;
	impl->CloseDevice = EMSCRIPTENAUDIO_CloseDevice;

	impl->OnlyHasDefaultOutputDevice = 1;

	/* no threads here */
	impl->SkipMixerLock = 1;
	impl->ProvidesOwnCallbackThread = 1;

	/* check availability */
	available = EM_ASM_INT_V({
	if (typeof(AudioContext) !== 'undefined') {
	return 1;
	} else if (typeof(webkitAudioContext) !== 'undefined') {
	return 1;
	}
	return 0;
	});

	if (!available) {
	SDL_SetError("No audio context available");
	}

	capture_available = available && EM_ASM_INT_V({
	if ((typeof(navigator.mediaDevices) !== 'undefined') && (typeof(navigator.mediaDevices.getUserMedia) !== 'undefined')) {
	return 1;
	} else if (typeof(navigator.webkitGetUserMedia) !== 'undefined') {
	return 1;
	}
	return 0;
	});

	impl->HasCaptureSupport = capture_available ? SDL_TRUE : SDL_FALSE;
	impl->OnlyHasDefaultCaptureDevice = capture_available ? SDL_TRUE : SDL_FALSE;

	return available;
	}

	AudioBootStrap EMSCRIPTENAUDIO_bootstrap = {
	"emscripten", "SDL emscripten audio driver", EMSCRIPTENAUDIO_Init, 0
	};

	#endif /* SDL_AUDIO_DRIVER_EMSCRIPTEN */

	/* vi: set ts=4 sw=4 expandtab: */