352 lines
14 KiB
C
352 lines
14 KiB
C
/*
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Simple DirectMedia Layer
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Copyright (C) 1997-2024 Sam Lantinga <slouken@libsdl.org>
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This software is provided 'as-is', without any express or implied
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warranty. In no event will the authors be held liable for any damages
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arising from the use of this software.
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Permission is granted to anyone to use this software for any purpose,
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including commercial applications, and to alter it and redistribute it
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freely, subject to the following restrictions:
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1. The origin of this software must not be misrepresented; you must not
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claim that you wrote the original software. If you use this software
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in a product, an acknowledgment in the product documentation would be
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appreciated but is not required.
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2. Altered source versions must be plainly marked as such, and must not be
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misrepresented as being the original software.
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3. This notice may not be removed or altered from any source distribution.
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*/
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#include "SDL_internal.h"
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#ifdef SDL_AUDIO_DRIVER_EMSCRIPTEN
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#include "../SDL_sysaudio.h"
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#include "SDL_emscriptenaudio.h"
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#include <emscripten/emscripten.h>
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// just turn off clang-format for this whole file, this INDENT_OFF stuff on
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// each EM_ASM section is ugly.
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/* *INDENT-OFF* */ /* clang-format off */
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static Uint8 *EMSCRIPTENAUDIO_GetDeviceBuf(SDL_AudioDevice *device, int *buffer_size)
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{
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return device->hidden->mixbuf;
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}
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static int EMSCRIPTENAUDIO_PlayDevice(SDL_AudioDevice *device, const Uint8 *buffer, int buffer_size)
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{
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const int framelen = SDL_AUDIO_FRAMESIZE(device->spec);
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MAIN_THREAD_EM_ASM({
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var SDL3 = Module['SDL3'];
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var numChannels = SDL3.audio.currentOutputBuffer['numberOfChannels'];
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for (var c = 0; c < numChannels; ++c) {
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var channelData = SDL3.audio.currentOutputBuffer['getChannelData'](c);
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if (channelData.length != $1) {
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throw 'Web Audio output buffer length mismatch! Destination size: ' + channelData.length + ' samples vs expected ' + $1 + ' samples!';
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}
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for (var j = 0; j < $1; ++j) {
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channelData[j] = HEAPF32[$0 + ((j*numChannels + c) << 2) >> 2]; // !!! FIXME: why are these shifts here?
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}
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}
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}, buffer, buffer_size / framelen);
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return 0;
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}
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static void EMSCRIPTENAUDIO_FlushCapture(SDL_AudioDevice *device)
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{
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// Do nothing, the new data will just be dropped.
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}
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static int EMSCRIPTENAUDIO_CaptureFromDevice(SDL_AudioDevice *device, void *buffer, int buflen)
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{
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MAIN_THREAD_EM_ASM({
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var SDL3 = Module['SDL3'];
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var numChannels = SDL3.capture.currentCaptureBuffer.numberOfChannels;
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for (var c = 0; c < numChannels; ++c) {
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var channelData = SDL3.capture.currentCaptureBuffer.getChannelData(c);
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if (channelData.length != $1) {
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throw 'Web Audio capture buffer length mismatch! Destination size: ' + channelData.length + ' samples vs expected ' + $1 + ' samples!';
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}
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if (numChannels == 1) { // fastpath this a little for the common (mono) case.
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for (var j = 0; j < $1; ++j) {
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setValue($0 + (j * 4), channelData[j], 'float');
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}
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} else {
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for (var j = 0; j < $1; ++j) {
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setValue($0 + (((j * numChannels) + c) * 4), channelData[j], 'float');
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}
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}
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}
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}, buffer, (buflen / sizeof(float)) / device->spec.channels);
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return buflen;
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}
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static void EMSCRIPTENAUDIO_CloseDevice(SDL_AudioDevice *device)
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{
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if (!device->hidden) {
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return;
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}
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MAIN_THREAD_EM_ASM({
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var SDL3 = Module['SDL3'];
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if ($0) {
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if (SDL3.capture.silenceTimer !== undefined) {
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clearInterval(SDL3.capture.silenceTimer);
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}
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if (SDL3.capture.stream !== undefined) {
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var tracks = SDL3.capture.stream.getAudioTracks();
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for (var i = 0; i < tracks.length; i++) {
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SDL3.capture.stream.removeTrack(tracks[i]);
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}
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}
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if (SDL3.capture.scriptProcessorNode !== undefined) {
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SDL3.capture.scriptProcessorNode.onaudioprocess = function(audioProcessingEvent) {};
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SDL3.capture.scriptProcessorNode.disconnect();
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}
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if (SDL3.capture.mediaStreamNode !== undefined) {
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SDL3.capture.mediaStreamNode.disconnect();
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}
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SDL3.capture = undefined;
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} else {
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if (SDL3.audio.scriptProcessorNode != undefined) {
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SDL3.audio.scriptProcessorNode.disconnect();
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}
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if (SDL3.audio.silenceTimer !== undefined) {
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clearInterval(SDL3.audio.silenceTimer);
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}
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SDL3.audio = undefined;
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}
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if ((SDL3.audioContext !== undefined) && (SDL3.audio === undefined) && (SDL3.capture === undefined)) {
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SDL3.audioContext.close();
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SDL3.audioContext = undefined;
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}
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}, device->iscapture);
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SDL_free(device->hidden->mixbuf);
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SDL_free(device->hidden);
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device->hidden = NULL;
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SDL_AudioThreadFinalize(device);
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}
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EM_JS_DEPS(sdlaudio, "$autoResumeAudioContext,$dynCall");
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static int EMSCRIPTENAUDIO_OpenDevice(SDL_AudioDevice *device)
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{
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// based on parts of library_sdl.js
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// create context
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const int result = MAIN_THREAD_EM_ASM_INT({
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if (typeof(Module['SDL3']) === 'undefined') {
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Module['SDL3'] = {};
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}
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var SDL3 = Module['SDL3'];
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if (!$0) {
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SDL3.audio = {};
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} else {
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SDL3.capture = {};
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}
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if (!SDL3.audioContext) {
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if (typeof(AudioContext) !== 'undefined') {
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SDL3.audioContext = new AudioContext();
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} else if (typeof(webkitAudioContext) !== 'undefined') {
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SDL3.audioContext = new webkitAudioContext();
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}
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if (SDL3.audioContext) {
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if ((typeof navigator.userActivation) === 'undefined') { // Firefox doesn't have this (as of August 2023), use autoResumeAudioContext instead.
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autoResumeAudioContext(SDL3.audioContext);
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}
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}
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}
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return SDL3.audioContext === undefined ? -1 : 0;
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}, device->iscapture);
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if (result < 0) {
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return SDL_SetError("Web Audio API is not available!");
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}
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device->spec.format = SDL_AUDIO_F32; // web audio only supports floats
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// Initialize all variables that we clean on shutdown
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device->hidden = (struct SDL_PrivateAudioData *)SDL_calloc(1, sizeof(*device->hidden));
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if (!device->hidden) {
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return -1;
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}
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// limit to native freq
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device->spec.freq = EM_ASM_INT({ return Module['SDL3'].audioContext.sampleRate; });
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SDL_UpdatedAudioDeviceFormat(device);
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if (!device->iscapture) {
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device->hidden->mixbuf = (Uint8 *)SDL_malloc(device->buffer_size);
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if (!device->hidden->mixbuf) {
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return -1;
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}
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SDL_memset(device->hidden->mixbuf, device->silence_value, device->buffer_size);
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}
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if (device->iscapture) {
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/* The idea is to take the capture media stream, hook it up to an
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audio graph where we can pass it through a ScriptProcessorNode
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to access the raw PCM samples and push them to the SDL app's
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callback. From there, we "process" the audio data into silence
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and forget about it.
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This should, strictly speaking, use MediaRecorder for capture, but
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this API is cleaner to use and better supported, and fires a
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callback whenever there's enough data to fire down into the app.
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The downside is that we are spending CPU time silencing a buffer
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that the audiocontext uselessly mixes into any output. On the
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upside, both of those things are not only run in native code in
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the browser, they're probably SIMD code, too. MediaRecorder
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feels like it's a pretty inefficient tapdance in similar ways,
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to be honest. */
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MAIN_THREAD_EM_ASM({
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var SDL3 = Module['SDL3'];
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var have_microphone = function(stream) {
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//console.log('SDL audio capture: we have a microphone! Replacing silence callback.');
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if (SDL3.capture.silenceTimer !== undefined) {
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clearInterval(SDL3.capture.silenceTimer);
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SDL3.capture.silenceTimer = undefined;
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SDL3.capture.silenceBuffer = undefined
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}
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SDL3.capture.mediaStreamNode = SDL3.audioContext.createMediaStreamSource(stream);
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SDL3.capture.scriptProcessorNode = SDL3.audioContext.createScriptProcessor($1, $0, 1);
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SDL3.capture.scriptProcessorNode.onaudioprocess = function(audioProcessingEvent) {
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if ((SDL3 === undefined) || (SDL3.capture === undefined)) { return; }
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audioProcessingEvent.outputBuffer.getChannelData(0).fill(0.0);
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SDL3.capture.currentCaptureBuffer = audioProcessingEvent.inputBuffer;
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dynCall('vi', $2, [$3]);
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};
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SDL3.capture.mediaStreamNode.connect(SDL3.capture.scriptProcessorNode);
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SDL3.capture.scriptProcessorNode.connect(SDL3.audioContext.destination);
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SDL3.capture.stream = stream;
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};
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var no_microphone = function(error) {
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//console.log('SDL audio capture: we DO NOT have a microphone! (' + error.name + ')...leaving silence callback running.');
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};
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// we write silence to the audio callback until the microphone is available (user approves use, etc).
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SDL3.capture.silenceBuffer = SDL3.audioContext.createBuffer($0, $1, SDL3.audioContext.sampleRate);
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SDL3.capture.silenceBuffer.getChannelData(0).fill(0.0);
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var silence_callback = function() {
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SDL3.capture.currentCaptureBuffer = SDL3.capture.silenceBuffer;
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dynCall('vi', $2, [$3]);
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};
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SDL3.capture.silenceTimer = setInterval(silence_callback, ($1 / SDL3.audioContext.sampleRate) * 1000);
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if ((navigator.mediaDevices !== undefined) && (navigator.mediaDevices.getUserMedia !== undefined)) {
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navigator.mediaDevices.getUserMedia({ audio: true, video: false }).then(have_microphone).catch(no_microphone);
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} else if (navigator.webkitGetUserMedia !== undefined) {
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navigator.webkitGetUserMedia({ audio: true, video: false }, have_microphone, no_microphone);
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}
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}, device->spec.channels, device->sample_frames, SDL_CaptureAudioThreadIterate, device);
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} else {
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// setup a ScriptProcessorNode
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MAIN_THREAD_EM_ASM({
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var SDL3 = Module['SDL3'];
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SDL3.audio.scriptProcessorNode = SDL3.audioContext['createScriptProcessor']($1, 0, $0);
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SDL3.audio.scriptProcessorNode['onaudioprocess'] = function (e) {
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if ((SDL3 === undefined) || (SDL3.audio === undefined)) { return; }
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// if we're actually running the node, we don't need the fake callback anymore, so kill it.
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if (SDL3.audio.silenceTimer !== undefined) {
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clearInterval(SDL3.audio.silenceTimer);
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SDL3.audio.silenceTimer = undefined;
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SDL3.audio.silenceBuffer = undefined;
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}
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SDL3.audio.currentOutputBuffer = e['outputBuffer'];
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dynCall('vi', $2, [$3]);
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};
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SDL3.audio.scriptProcessorNode['connect'](SDL3.audioContext['destination']);
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if (SDL3.audioContext.state === 'suspended') { // uhoh, autoplay is blocked.
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SDL3.audio.silenceBuffer = SDL3.audioContext.createBuffer($0, $1, SDL3.audioContext.sampleRate);
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SDL3.audio.silenceBuffer.getChannelData(0).fill(0.0);
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var silence_callback = function() {
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if ((typeof navigator.userActivation) !== 'undefined') { // Almost everything modern except Firefox (as of August 2023)
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if (navigator.userActivation.hasBeenActive) {
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SDL3.audioContext.resume();
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}
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}
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// the buffer that gets filled here just gets ignored, so the app can make progress
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// and/or avoid flooding audio queues until we can actually play audio.
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SDL3.audio.currentOutputBuffer = SDL3.audio.silenceBuffer;
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dynCall('vi', $2, [$3]);
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SDL3.audio.currentOutputBuffer = undefined;
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};
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SDL3.audio.silenceTimer = setInterval(silence_callback, ($1 / SDL3.audioContext.sampleRate) * 1000);
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}
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}, device->spec.channels, device->sample_frames, SDL_OutputAudioThreadIterate, device);
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}
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return 0;
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}
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static SDL_bool EMSCRIPTENAUDIO_Init(SDL_AudioDriverImpl *impl)
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{
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SDL_bool available, capture_available;
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impl->OpenDevice = EMSCRIPTENAUDIO_OpenDevice;
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impl->CloseDevice = EMSCRIPTENAUDIO_CloseDevice;
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impl->GetDeviceBuf = EMSCRIPTENAUDIO_GetDeviceBuf;
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impl->PlayDevice = EMSCRIPTENAUDIO_PlayDevice;
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impl->FlushCapture = EMSCRIPTENAUDIO_FlushCapture;
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impl->CaptureFromDevice = EMSCRIPTENAUDIO_CaptureFromDevice;
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impl->OnlyHasDefaultOutputDevice = SDL_TRUE;
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// technically, this is just runs in idle time in the main thread, but it's close enough to a "thread" for our purposes.
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impl->ProvidesOwnCallbackThread = SDL_TRUE;
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// check availability
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available = MAIN_THREAD_EM_ASM_INT({
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if (typeof(AudioContext) !== 'undefined') {
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return true;
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} else if (typeof(webkitAudioContext) !== 'undefined') {
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return true;
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}
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return false;
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});
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if (!available) {
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SDL_SetError("No audio context available");
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}
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capture_available = available && MAIN_THREAD_EM_ASM_INT({
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if ((typeof(navigator.mediaDevices) !== 'undefined') && (typeof(navigator.mediaDevices.getUserMedia) !== 'undefined')) {
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return true;
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} else if (typeof(navigator.webkitGetUserMedia) !== 'undefined') {
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return true;
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}
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return false;
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});
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impl->HasCaptureSupport = capture_available;
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impl->OnlyHasDefaultCaptureDevice = capture_available;
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return available;
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}
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AudioBootStrap EMSCRIPTENAUDIO_bootstrap = {
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"emscripten", "SDL emscripten audio driver", EMSCRIPTENAUDIO_Init, SDL_FALSE
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};
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/* *INDENT-ON* */ /* clang-format on */
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#endif // SDL_AUDIO_DRIVER_EMSCRIPTEN
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