update sdl Merge commit '4d48f9d23713d94b861da7b5d41baf2a41334994'
This commit is contained in:
@ -27,15 +27,18 @@
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#include <emscripten/emscripten.h>
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/* !!! FIXME: this currently expects that the audio callback runs in the main thread,
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!!! FIXME: in intervals when the application isn't running, but that may not be
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!!! FIXME: true always once pthread support becomes widespread. Revisit this code
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!!! FIXME: at some point and see what needs to be done for that! */
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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 void FeedAudioDevice(SDL_AudioDevice *_this, const void *buf, const int buflen)
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static Uint8 *EMSCRIPTENAUDIO_GetDeviceBuf(SDL_AudioDevice *device, int *buffer_size)
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{
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const int framelen = (SDL_AUDIO_BITSIZE(_this->spec.format) / 8) * _this->spec.channels;
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/* *INDENT-OFF* */ /* clang-format off */
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return device->hidden->mixbuf;
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}
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static void EMSCRIPTENAUDIO_PlayDevice(SDL_AudioDevice *device, const Uint8 *buffer, int buffer_size)
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{
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const int framelen = (SDL_AUDIO_BITSIZE(device->spec.format) / 8) * device->spec.channels;
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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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@ -46,65 +49,25 @@ static void FeedAudioDevice(SDL_AudioDevice *_this, const void *buf, const int b
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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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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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}, buf, buflen / framelen);
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/* *INDENT-ON* */ /* clang-format on */
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}, buffer, buffer_size / framelen);
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}
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static void HandleAudioProcess(SDL_AudioDevice *_this)
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static void HandleAudioProcess(SDL_AudioDevice *device) // this fires when the main thread is idle.
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{
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SDL_AudioCallback callback = _this->callbackspec.callback;
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const int stream_len = _this->callbackspec.size;
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/* Only do something if audio is enabled */
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if (!SDL_AtomicGet(&_this->enabled) || SDL_AtomicGet(&_this->paused)) {
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if (_this->stream) {
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SDL_ClearAudioStream(_this->stream);
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}
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SDL_memset(_this->work_buffer, _this->spec.silence, _this->spec.size);
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FeedAudioDevice(_this, _this->work_buffer, _this->spec.size);
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return;
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}
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if (_this->stream == NULL) { /* no conversion necessary. */
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SDL_assert(_this->spec.size == stream_len);
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callback(_this->callbackspec.userdata, _this->work_buffer, stream_len);
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} else { /* streaming/converting */
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int got;
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while (SDL_GetAudioStreamAvailable(_this->stream) < ((int)_this->spec.size)) {
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callback(_this->callbackspec.userdata, _this->work_buffer, stream_len);
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if (SDL_PutAudioStreamData(_this->stream, _this->work_buffer, stream_len) == -1) {
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SDL_ClearAudioStream(_this->stream);
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SDL_AtomicSet(&_this->enabled, 0);
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break;
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}
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}
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got = SDL_GetAudioStreamData(_this->stream, _this->work_buffer, _this->spec.size);
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SDL_assert((got < 0) || (got == _this->spec.size));
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if (got != _this->spec.size) {
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SDL_memset(_this->work_buffer, _this->spec.silence, _this->spec.size);
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}
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}
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FeedAudioDevice(_this, _this->work_buffer, _this->spec.size);
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SDL_OutputAudioThreadIterate(device);
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}
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static void HandleCaptureProcess(SDL_AudioDevice *_this)
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static void EMSCRIPTENAUDIO_FlushCapture(SDL_AudioDevice *device)
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{
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SDL_AudioCallback callback = _this->callbackspec.callback;
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const int stream_len = _this->callbackspec.size;
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// Do nothing, the new data will just be dropped.
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}
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/* Only do something if audio is enabled */
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if (!SDL_AtomicGet(&_this->enabled) || SDL_AtomicGet(&_this->paused)) {
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SDL_ClearAudioStream(_this->stream);
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return;
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}
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/* *INDENT-OFF* */ /* clang-format off */
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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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@ -114,7 +77,7 @@ static void HandleCaptureProcess(SDL_AudioDevice *_this)
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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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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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@ -124,33 +87,22 @@ static void HandleCaptureProcess(SDL_AudioDevice *_this)
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}
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}
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}
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}, _this->work_buffer, (_this->spec.size / sizeof(float)) / _this->spec.channels);
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/* *INDENT-ON* */ /* clang-format on */
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}, buffer, (buflen / sizeof(float)) / device->spec.channels);
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/* okay, we've got an interleaved float32 array in C now. */
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if (_this->stream == NULL) { /* no conversion necessary. */
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SDL_assert(_this->spec.size == stream_len);
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callback(_this->callbackspec.userdata, _this->work_buffer, stream_len);
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} else { /* streaming/converting */
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if (SDL_PutAudioStreamData(_this->stream, _this->work_buffer, _this->spec.size) == -1) {
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SDL_AtomicSet(&_this->enabled, 0);
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}
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while (SDL_GetAudioStreamAvailable(_this->stream) >= stream_len) {
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const int got = SDL_GetAudioStreamData(_this->stream, _this->work_buffer, stream_len);
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SDL_assert((got < 0) || (got == stream_len));
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if (got != stream_len) {
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SDL_memset(_this->work_buffer, _this->callbackspec.silence, stream_len);
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}
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callback(_this->callbackspec.userdata, _this->work_buffer, stream_len); /* Send it to the app. */
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}
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}
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return buflen;
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}
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static void EMSCRIPTENAUDIO_CloseDevice(SDL_AudioDevice *_this)
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static void HandleCaptureProcess(SDL_AudioDevice *device) // this fires when the main thread is idle.
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{
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/* *INDENT-OFF* */ /* clang-format off */
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SDL_CaptureAudioThreadIterate(device);
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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 ($0) {
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@ -188,29 +140,23 @@ static void EMSCRIPTENAUDIO_CloseDevice(SDL_AudioDevice *_this)
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SDL3.audioContext.close();
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SDL3.audioContext = undefined;
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}
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}, _this->iscapture);
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/* *INDENT-ON* */ /* clang-format on */
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}, device->iscapture);
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#if 0 /* !!! FIXME: currently not used. Can we move some stuff off the SDL3 namespace? --ryan. */
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SDL_free(_this->hidden);
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#endif
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SDL_free(device->hidden->mixbuf);
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SDL_free(device->hidden);
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device->hidden = NULL;
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SDL_AudioThreadFinalize(device);
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}
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EM_JS_DEPS(sdlaudio, "$autoResumeAudioContext,$dynCall");
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static int EMSCRIPTENAUDIO_OpenDevice(SDL_AudioDevice *_this, const char *devname)
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static int EMSCRIPTENAUDIO_OpenDevice(SDL_AudioDevice *device)
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{
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SDL_AudioFormat test_format;
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const SDL_AudioFormat *closefmts;
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SDL_bool iscapture = _this->iscapture;
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int result;
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// based on parts of library_sdl.js
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/* based on parts of library_sdl.js */
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/* *INDENT-OFF* */ /* clang-format off */
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/* create context */
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result = MAIN_THREAD_EM_ASM_INT({
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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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@ -232,57 +178,41 @@ static int EMSCRIPTENAUDIO_OpenDevice(SDL_AudioDevice *_this, const char *devnam
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}
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}
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return SDL3.audioContext === undefined ? -1 : 0;
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}, iscapture);
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/* *INDENT-ON* */ /* clang-format on */
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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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closefmts = SDL_ClosestAudioFormats(_this->spec.format);
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while ((test_format = *(closefmts++)) != 0) {
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switch (test_format) {
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case SDL_AUDIO_F32: /* web audio only supports floats */
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break;
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default:
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continue;
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}
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break;
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}
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device->spec.format = SDL_AUDIO_F32; // web audio only supports floats
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if (!test_format) {
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/* Didn't find a compatible format :( */
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return SDL_SetError("%s: Unsupported audio format", "emscripten");
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}
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_this->spec.format = test_format;
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/* Initialize all variables that we clean on shutdown */
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#if 0 /* !!! FIXME: currently not used. Can we move some stuff off the SDL3 namespace? --ryan. */
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_this->hidden = (struct SDL_PrivateAudioData *)SDL_malloc(sizeof(*_this->hidden));
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if (_this->hidden == NULL) {
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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 == NULL) {
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return SDL_OutOfMemory();
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}
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SDL_zerop(_this->hidden);
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#endif
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_this->hidden = (struct SDL_PrivateAudioData *)0x1;
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/* limit to native freq */
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_this->spec.freq = EM_ASM_INT({
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var SDL3 = Module['SDL3'];
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return SDL3.audioContext.sampleRate;
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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_CalculateAudioSpec(&_this->spec);
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SDL_UpdatedAudioDeviceFormat(device);
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/* *INDENT-OFF* */ /* clang-format off */
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if (iscapture) {
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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 == NULL) {
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return SDL_OutOfMemory();
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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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and forget about it.
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/* This should, strictly speaking, use MediaRecorder for capture, but
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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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@ -317,7 +247,7 @@ static int EMSCRIPTENAUDIO_OpenDevice(SDL_AudioDevice *_this, const char *devnam
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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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// 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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@ -332,9 +262,9 @@ static int EMSCRIPTENAUDIO_OpenDevice(SDL_AudioDevice *_this, const char *devnam
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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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}, _this->spec.channels, _this->spec.samples, HandleCaptureProcess, _this);
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}, device->spec.channels, device->sample_frames, HandleCaptureProcess, device);
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} else {
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/* setup a ScriptProcessorNode */
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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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@ -344,33 +274,29 @@ static int EMSCRIPTENAUDIO_OpenDevice(SDL_AudioDevice *_this, const char *devnam
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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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}, _this->spec.channels, _this->spec.samples, HandleAudioProcess, _this);
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}, device->spec.channels, device->sample_frames, HandleAudioProcess, device);
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}
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/* *INDENT-ON* */ /* clang-format on */
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return 0;
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}
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static void EMSCRIPTENAUDIO_LockOrUnlockDeviceWithNoMixerLock(SDL_AudioDevice *device)
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{
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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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/* Set the function pointers */
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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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/* no threads here */
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impl->LockDevice = impl->UnlockDevice = EMSCRIPTENAUDIO_LockOrUnlockDeviceWithNoMixerLock;
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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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/* *INDENT-OFF* */ /* clang-format off */
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/* check availability */
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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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@ -378,14 +304,12 @@ static SDL_bool EMSCRIPTENAUDIO_Init(SDL_AudioDriverImpl *impl)
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return true;
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}
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return false;
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});
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/* *INDENT-ON* */ /* clang-format on */
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}) ? SDL_TRUE : SDL_FALSE;
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if (!available) {
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SDL_SetError("No audio context available");
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}
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/* *INDENT-OFF* */ /* clang-format off */
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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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@ -393,8 +317,7 @@ static SDL_bool EMSCRIPTENAUDIO_Init(SDL_AudioDriverImpl *impl)
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return true;
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}
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return false;
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});
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/* *INDENT-ON* */ /* clang-format on */
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}) ? SDL_TRUE : SDL_FALSE;
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impl->HasCaptureSupport = capture_available ? SDL_TRUE : SDL_FALSE;
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impl->OnlyHasDefaultCaptureDevice = capture_available ? SDL_TRUE : SDL_FALSE;
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@ -406,4 +329,6 @@ AudioBootStrap EMSCRIPTENAUDIO_bootstrap = {
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"emscripten", "SDL emscripten audio driver", EMSCRIPTENAUDIO_Init, SDL_FALSE
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};
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#endif /* SDL_AUDIO_DRIVER_EMSCRIPTEN */
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/* *INDENT-ON* */ /* clang-format on */
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#endif // SDL_AUDIO_DRIVER_EMSCRIPTEN
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Reference in New Issue
Block a user