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emscriptenaudio: Updated for SDL3 audio API.

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This commit is contained in:
Ryan C. Gordon
2023-07-05 11:51:06 -04:00
parent 79cc29ba35
commit a0528cd5ed
2 changed files with 74 additions and 153 deletions
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225
src/audio/emscripten/SDL_emscriptenaudio.c
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@@ -27,15 +27,18 @@
#include <emscripten/emscripten.h> #include <emscripten/emscripten.h>
/* !!! FIXME: this currently expects that the audio callback runs in the main thread, // just turn off clang-format for this whole file, this INDENT_OFF stuff on
!!! FIXME: in intervals when the application isn't running, but that may not be // each EM_ASM section is ugly.
!!! FIXME: true always once pthread support becomes widespread. Revisit this code /* *INDENT-OFF* */ /* clang-format off */
!!! FIXME: at some point and see what needs to be done for that! */
static void FeedAudioDevice(SDL_AudioDevice *_this, const void *buf, const int buflen) static Uint8 *EMSCRIPTENAUDIO_GetDeviceBuf(SDL_AudioDevice *device, int *buffer_size)
{ {
const int framelen = (SDL_AUDIO_BITSIZE(_this->spec.format) / 8) * _this->spec.channels; return device->hidden->mixbuf;
/* *INDENT-OFF* */ /* clang-format off */ }
static void EMSCRIPTENAUDIO_PlayDevice(SDL_AudioDevice *device, const Uint8 *buffer, int buffer_size)
{
const int framelen = (SDL_AUDIO_BITSIZE(device->spec.format) / 8) * device->spec.channels;
MAIN_THREAD_EM_ASM({ MAIN_THREAD_EM_ASM({
var SDL3 = Module['SDL3']; var SDL3 = Module['SDL3'];
var numChannels = SDL3.audio.currentOutputBuffer['numberOfChannels']; var numChannels = SDL3.audio.currentOutputBuffer['numberOfChannels'];
@@ -46,65 +49,25 @@ static void FeedAudioDevice(SDL_AudioDevice *_this, const void *buf, const int b
} }
for (var j = 0; j < $1; ++j) { for (var j = 0; j < $1; ++j) {
channelData[j] = HEAPF32[$0 + ((j*numChannels + c) << 2) >> 2]; /* !!! FIXME: why are these shifts here? */ channelData[j] = HEAPF32[$0 + ((j*numChannels + c) << 2) >> 2]; // !!! FIXME: why are these shifts here?
} }
} }
}, buf, buflen / framelen); }, buffer, buffer_size / framelen);
/* *INDENT-ON* */ /* clang-format on */
} }
static void HandleAudioProcess(SDL_AudioDevice *_this) static void HandleAudioProcess(SDL_AudioDevice *device) // this fires when the main thread is idle.
{ {
SDL_AudioCallback callback = _this->callbackspec.callback; SDL_OutputAudioThreadIterate(device);
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_ClearAudioStream(_this->stream);
}
SDL_memset(_this->work_buffer, _this->spec.silence, _this->spec.size);
FeedAudioDevice(_this, _this->work_buffer, _this->spec.size);
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_GetAudioStreamAvailable(_this->stream) < ((int)_this->spec.size)) {
callback(_this->callbackspec.userdata, _this->work_buffer, stream_len);
if (SDL_PutAudioStreamData(_this->stream, _this->work_buffer, stream_len) == -1) {
SDL_ClearAudioStream(_this->stream);
SDL_AtomicSet(&_this->enabled, 0);
break;
}
}
got = SDL_GetAudioStreamData(_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(SDL_AudioDevice *_this)
static void EMSCRIPTENAUDIO_FlushCapture(SDL_AudioDevice *device)
{ {
SDL_AudioCallback callback = _this->callbackspec.callback; // Do nothing, the new data will just be dropped.
const int stream_len = _this->callbackspec.size; }
/* Only do something if audio is enabled */ static int EMSCRIPTENAUDIO_CaptureFromDevice(SDL_AudioDevice *device, void *buffer, int buflen)
if (!SDL_AtomicGet(&_this->enabled) || SDL_AtomicGet(&_this->paused)) { {
SDL_ClearAudioStream(_this->stream);
return;
}
/* *INDENT-OFF* */ /* clang-format off */
MAIN_THREAD_EM_ASM({ MAIN_THREAD_EM_ASM({
var SDL3 = Module['SDL3']; var SDL3 = Module['SDL3'];
var numChannels = SDL3.capture.currentCaptureBuffer.numberOfChannels; var numChannels = SDL3.capture.currentCaptureBuffer.numberOfChannels;
@@ -114,7 +77,7 @@ static void HandleCaptureProcess(SDL_AudioDevice *_this)
throw 'Web Audio capture buffer length mismatch! Destination size: ' + channelData.length + ' samples vs expected ' + $1 + ' samples!'; 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. */ if (numChannels == 1) { // fastpath this a little for the common (mono) case.
for (var j = 0; j < $1; ++j) { for (var j = 0; j < $1; ++j) {
setValue($0 + (j * 4), channelData[j], 'float'); setValue($0 + (j * 4), channelData[j], 'float');
} }
@@ -124,33 +87,18 @@ static void HandleCaptureProcess(SDL_AudioDevice *_this)
} }
} }
} }
}, _this->work_buffer, (_this->spec.size / sizeof(float)) / _this->spec.channels); }, buffer, (buflen / sizeof(float)) / device->spec.channels);
/* *INDENT-ON* */ /* clang-format on */
/* okay, we've got an interleaved float32 array in C now. */ return buflen;
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_PutAudioStreamData(_this->stream, _this->work_buffer, _this->spec.size) == -1) {
SDL_AtomicSet(&_this->enabled, 0);
}
while (SDL_GetAudioStreamAvailable(_this->stream) >= stream_len) {
const int got = SDL_GetAudioStreamData(_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(SDL_AudioDevice *_this) static void HandleCaptureProcess(SDL_AudioDevice *device) // this fires when the main thread is idle.
{
SDL_CaptureAudioThreadIterate(device);
}
static void EMSCRIPTENAUDIO_CloseDevice(SDL_AudioDevice *device)
{ {
/* *INDENT-OFF* */ /* clang-format off */
MAIN_THREAD_EM_ASM({ MAIN_THREAD_EM_ASM({
var SDL3 = Module['SDL3']; var SDL3 = Module['SDL3'];
if ($0) { if ($0) {
@@ -188,29 +136,23 @@ static void EMSCRIPTENAUDIO_CloseDevice(SDL_AudioDevice *_this)
SDL3.audioContext.close(); SDL3.audioContext.close();
SDL3.audioContext = undefined; SDL3.audioContext = undefined;
} }
}, _this->iscapture); }, device->iscapture);
/* *INDENT-ON* */ /* clang-format on */
#if 0 /* !!! FIXME: currently not used. Can we move some stuff off the SDL3 namespace? --ryan. */ if (!device->hidden) {
SDL_free(_this->hidden); SDL_free(device->hidden->mixbuf);
#endif SDL_free(device->hidden);
device->hidden = NULL;
}
} }
EM_JS_DEPS(sdlaudio, "$autoResumeAudioContext,$dynCall"); EM_JS_DEPS(sdlaudio, "$autoResumeAudioContext,$dynCall");
static int EMSCRIPTENAUDIO_OpenDevice(SDL_AudioDevice *_this, const char *devname) static int EMSCRIPTENAUDIO_OpenDevice(SDL_AudioDevice *device)
{ {
SDL_AudioFormat test_format; // based on parts of library_sdl.js
const SDL_AudioFormat *closefmts;
SDL_bool iscapture = _this->iscapture;
int result;
/* based on parts of library_sdl.js */ // create context
const int result = MAIN_THREAD_EM_ASM_INT({
/* *INDENT-OFF* */ /* clang-format off */
/* create context */
result = MAIN_THREAD_EM_ASM_INT({
if (typeof(Module['SDL3']) === 'undefined') { if (typeof(Module['SDL3']) === 'undefined') {
Module['SDL3'] = {}; Module['SDL3'] = {};
} }
@@ -232,57 +174,41 @@ static int EMSCRIPTENAUDIO_OpenDevice(SDL_AudioDevice *_this, const char *devnam
} }
} }
return SDL3.audioContext === undefined ? -1 : 0; return SDL3.audioContext === undefined ? -1 : 0;
}, iscapture); }, device->iscapture);
/* *INDENT-ON* */ /* clang-format on */
if (result < 0) { if (result < 0) {
return SDL_SetError("Web Audio API is not available!"); return SDL_SetError("Web Audio API is not available!");
} }
closefmts = SDL_ClosestAudioFormats(_this->spec.format); device->spec.format = SDL_AUDIO_F32; // web audio only supports floats
while ((test_format = *(closefmts++)) != 0) {
switch (test_format) {
case SDL_AUDIO_F32: /* web audio only supports floats */
break;
default:
continue;
}
break;
}
if (!test_format) { // Initialize all variables that we clean on shutdown
/* Didn't find a compatible format :( */ device->hidden = (struct SDL_PrivateAudioData *)SDL_calloc(1, sizeof(*device->hidden));
return SDL_SetError("%s: Unsupported audio format", "emscripten"); if (device->hidden == NULL) {
}
_this->spec.format = test_format;
/* Initialize all variables that we clean on shutdown */
#if 0 /* !!! FIXME: currently not used. Can we move some stuff off the SDL3 namespace? --ryan. */
_this->hidden = (struct SDL_PrivateAudioData *)SDL_malloc(sizeof(*_this->hidden));
if (_this->hidden == NULL) {
return SDL_OutOfMemory(); return SDL_OutOfMemory();
} }
SDL_zerop(_this->hidden);
#endif
_this->hidden = (struct SDL_PrivateAudioData *)0x1;
/* limit to native freq */ // limit to native freq
_this->spec.freq = EM_ASM_INT({ device->spec.freq = EM_ASM_INT({ return Module['SDL3'].audioContext.sampleRate; });
var SDL3 = Module['SDL3'];
return SDL3.audioContext.sampleRate;
});
SDL_CalculateAudioSpec(&_this->spec); SDL_UpdatedAudioDeviceFormat(device);
/* *INDENT-OFF* */ /* clang-format off */ if (!device->iscapture) {
if (iscapture) { device->hidden->mixbuf = (Uint8 *)SDL_malloc(device->buffer_size);
if (device->hidden->mixbuf == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(device->hidden->mixbuf, device->silence_value, device->buffer_size);
}
if (device->iscapture) {
/* The idea is to take the capture media stream, hook it up to an /* The idea is to take the capture media stream, hook it up to an
audio graph where we can pass it through a ScriptProcessorNode audio graph where we can pass it through a ScriptProcessorNode
to access the raw PCM samples and push them to the SDL app's to access the raw PCM samples and push them to the SDL app's
callback. From there, we "process" the audio data into silence callback. From there, we "process" the audio data into silence
and forget about it. */ and forget about it.
/* This should, strictly speaking, use MediaRecorder for capture, but This should, strictly speaking, use MediaRecorder for capture, but
this API is cleaner to use and better supported, and fires a this API is cleaner to use and better supported, and fires a
callback whenever there's enough data to fire down into the app. callback whenever there's enough data to fire down into the app.
The downside is that we are spending CPU time silencing a buffer The downside is that we are spending CPU time silencing a buffer
@@ -317,7 +243,7 @@ static int EMSCRIPTENAUDIO_OpenDevice(SDL_AudioDevice *_this, const char *devnam
//console.log('SDL audio capture: we DO NOT have a microphone! (' + error.name + ')...leaving silence callback running.'); //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). */ // we write silence to the audio callback until the microphone is available (user approves use, etc).
SDL3.capture.silenceBuffer = SDL3.audioContext.createBuffer($0, $1, SDL3.audioContext.sampleRate); SDL3.capture.silenceBuffer = SDL3.audioContext.createBuffer($0, $1, SDL3.audioContext.sampleRate);
SDL3.capture.silenceBuffer.getChannelData(0).fill(0.0); SDL3.capture.silenceBuffer.getChannelData(0).fill(0.0);
var silence_callback = function() { var silence_callback = function() {
@@ -332,9 +258,9 @@ static int EMSCRIPTENAUDIO_OpenDevice(SDL_AudioDevice *_this, const char *devnam
} else if (navigator.webkitGetUserMedia !== undefined) { } else if (navigator.webkitGetUserMedia !== undefined) {
navigator.webkitGetUserMedia({ audio: true, video: false }, have_microphone, no_microphone); navigator.webkitGetUserMedia({ audio: true, video: false }, have_microphone, no_microphone);
} }
}, _this->spec.channels, _this->spec.samples, HandleCaptureProcess, _this); }, device->spec.channels, device->sample_frames, HandleCaptureProcess, device);
} else { } else {
/* setup a ScriptProcessorNode */ // setup a ScriptProcessorNode
MAIN_THREAD_EM_ASM({ MAIN_THREAD_EM_ASM({
var SDL3 = Module['SDL3']; var SDL3 = Module['SDL3'];
SDL3.audio.scriptProcessorNode = SDL3.audioContext['createScriptProcessor']($1, 0, $0); SDL3.audio.scriptProcessorNode = SDL3.audioContext['createScriptProcessor']($1, 0, $0);
@@ -344,33 +270,29 @@ static int EMSCRIPTENAUDIO_OpenDevice(SDL_AudioDevice *_this, const char *devnam
dynCall('vi', $2, [$3]); dynCall('vi', $2, [$3]);
}; };
SDL3.audio.scriptProcessorNode['connect'](SDL3.audioContext['destination']); SDL3.audio.scriptProcessorNode['connect'](SDL3.audioContext['destination']);
}, _this->spec.channels, _this->spec.samples, HandleAudioProcess, _this); }, device->spec.channels, device->sample_frames, HandleAudioProcess, device);
} }
/* *INDENT-ON* */ /* clang-format on */
return 0; return 0;
} }
static void EMSCRIPTENAUDIO_LockOrUnlockDeviceWithNoMixerLock(SDL_AudioDevice *device)
{
}
static SDL_bool EMSCRIPTENAUDIO_Init(SDL_AudioDriverImpl *impl) static SDL_bool EMSCRIPTENAUDIO_Init(SDL_AudioDriverImpl *impl)
{ {
SDL_bool available, capture_available; SDL_bool available, capture_available;
/* Set the function pointers */
impl->OpenDevice = EMSCRIPTENAUDIO_OpenDevice; impl->OpenDevice = EMSCRIPTENAUDIO_OpenDevice;
impl->CloseDevice = EMSCRIPTENAUDIO_CloseDevice; impl->CloseDevice = EMSCRIPTENAUDIO_CloseDevice;
impl->GetDeviceBuf = EMSCRIPTENAUDIO_GetDeviceBuf;
impl->PlayDevice = EMSCRIPTENAUDIO_PlayDevice;
impl->FlushCapture = EMSCRIPTENAUDIO_FlushCapture;
impl->CaptureFromDevice = EMSCRIPTENAUDIO_CaptureFromDevice;
impl->OnlyHasDefaultOutputDevice = SDL_TRUE; impl->OnlyHasDefaultOutputDevice = SDL_TRUE;
/* no threads here */ // technically, this is just runs in idle time in the main thread, but it's close enough to a "thread" for our purposes.
impl->LockDevice = impl->UnlockDevice = EMSCRIPTENAUDIO_LockOrUnlockDeviceWithNoMixerLock;
impl->ProvidesOwnCallbackThread = SDL_TRUE; impl->ProvidesOwnCallbackThread = SDL_TRUE;
/* *INDENT-OFF* */ /* clang-format off */ // check availability
/* check availability */
available = MAIN_THREAD_EM_ASM_INT({ available = MAIN_THREAD_EM_ASM_INT({
if (typeof(AudioContext) !== 'undefined') { if (typeof(AudioContext) !== 'undefined') {
return true; return true;
@@ -378,14 +300,12 @@ static SDL_bool EMSCRIPTENAUDIO_Init(SDL_AudioDriverImpl *impl)
return true; return true;
} }
return false; return false;
}); }) ? SDL_TRUE : SDL_FALSE;
/* *INDENT-ON* */ /* clang-format on */
if (!available) { if (!available) {
SDL_SetError("No audio context available"); SDL_SetError("No audio context available");
} }
/* *INDENT-OFF* */ /* clang-format off */
capture_available = available && MAIN_THREAD_EM_ASM_INT({ capture_available = available && MAIN_THREAD_EM_ASM_INT({
if ((typeof(navigator.mediaDevices) !== 'undefined') && (typeof(navigator.mediaDevices.getUserMedia) !== 'undefined')) { if ((typeof(navigator.mediaDevices) !== 'undefined') && (typeof(navigator.mediaDevices.getUserMedia) !== 'undefined')) {
return true; return true;
@@ -393,8 +313,7 @@ static SDL_bool EMSCRIPTENAUDIO_Init(SDL_AudioDriverImpl *impl)
return true; return true;
} }
return false; return false;
}); }) ? SDL_TRUE : SDL_FALSE;
/* *INDENT-ON* */ /* clang-format on */
impl->HasCaptureSupport = capture_available ? SDL_TRUE : SDL_FALSE; impl->HasCaptureSupport = capture_available ? SDL_TRUE : SDL_FALSE;
impl->OnlyHasDefaultCaptureDevice = capture_available ? SDL_TRUE : SDL_FALSE; impl->OnlyHasDefaultCaptureDevice = capture_available ? SDL_TRUE : SDL_FALSE;
@@ -406,4 +325,6 @@ AudioBootStrap EMSCRIPTENAUDIO_bootstrap = {
"emscripten", "SDL emscripten audio driver", EMSCRIPTENAUDIO_Init, SDL_FALSE "emscripten", "SDL emscripten audio driver", EMSCRIPTENAUDIO_Init, SDL_FALSE
}; };
#endif /* SDL_AUDIO_DRIVER_EMSCRIPTEN */ /* *INDENT-ON* */ /* clang-format on */
#endif // SDL_AUDIO_DRIVER_EMSCRIPTEN

2
src/audio/emscripten/SDL_emscriptenaudio.h
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@@ -27,7 +27,7 @@
struct SDL_PrivateAudioData struct SDL_PrivateAudioData
{ {
int unused; Uint8 *mixbuf;
}; };
#endif /* SDL_emscriptenaudio_h_ */ #endif /* SDL_emscriptenaudio_h_ */