mirror of
https://github.com/thunderbrewhq/thunderbrew
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feat(gx): add incomplete 'CGxDeviceGLSDL' (#2)
* chore(build): add vendored SDL 3.0.0 library * chore(build): add vendored glew-cmake-2.2.0 library * feat(console): in the presence of -opengl launch flag, change GxApi to OpenGl * feat(gx): add uncompleted CGxDeviceGLSDL targeting Windows and Linux * chore(build): change SDL3 linkage from shared (bad) to to static (good)
This commit is contained in:
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parent
934e0fb600
commit
706c8903a1
2043 changed files with 663533 additions and 5 deletions
450
vendor/sdl-3.0.0/src/audio/qnx/SDL_qsa_audio.c
vendored
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450
vendor/sdl-3.0.0/src/audio/qnx/SDL_qsa_audio.c
vendored
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@ -0,0 +1,450 @@
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/*
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Simple DirectMedia Layer
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Copyright (C) 1997-2021 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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// !!! FIXME: can this target support hotplugging?
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#include "../../SDL_internal.h"
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#ifdef SDL_AUDIO_DRIVER_QNX
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#include <errno.h>
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#include <unistd.h>
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#include <fcntl.h>
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#include <signal.h>
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#include <sys/types.h>
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#include <sys/time.h>
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#include <sched.h>
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#include <sys/select.h>
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#include <sys/neutrino.h>
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#include <sys/asoundlib.h>
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#include "SDL3/SDL_timer.h"
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#include "SDL3/SDL_audio.h"
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#include "../../core/unix/SDL_poll.h"
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#include "../SDL_sysaudio.h"
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#include "SDL_qsa_audio.h"
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// default channel communication parameters
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#define DEFAULT_CPARAMS_RATE 44100
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#define DEFAULT_CPARAMS_VOICES 1
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#define DEFAULT_CPARAMS_FRAG_SIZE 4096
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#define DEFAULT_CPARAMS_FRAGS_MIN 1
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#define DEFAULT_CPARAMS_FRAGS_MAX 1
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#define QSA_MAX_NAME_LENGTH 81+16 // Hardcoded in QSA, can't be changed
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static int QSA_SetError(const char *fn, int status)
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{
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return SDL_SetError("QSA: %s() failed: %s", fn, snd_strerror(status));
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}
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// !!! FIXME: does this need to be here? Does the SDL version not work?
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static void QSA_ThreadInit(SDL_AudioDevice *device)
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{
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// Increase default 10 priority to 25 to avoid jerky sound
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struct sched_param param;
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if (SchedGet(0, 0, ¶m) != -1) {
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param.sched_priority = param.sched_curpriority + 15;
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SchedSet(0, 0, SCHED_NOCHANGE, ¶m);
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}
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}
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// PCM channel parameters initialize function
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static void QSA_InitAudioParams(snd_pcm_channel_params_t * cpars)
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{
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SDL_zerop(cpars);
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cpars->channel = SND_PCM_CHANNEL_PLAYBACK;
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cpars->mode = SND_PCM_MODE_BLOCK;
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cpars->start_mode = SND_PCM_START_DATA;
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cpars->stop_mode = SND_PCM_STOP_STOP;
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cpars->format.format = SND_PCM_SFMT_S16_LE;
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cpars->format.interleave = 1;
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cpars->format.rate = DEFAULT_CPARAMS_RATE;
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cpars->format.voices = DEFAULT_CPARAMS_VOICES;
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cpars->buf.block.frag_size = DEFAULT_CPARAMS_FRAG_SIZE;
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cpars->buf.block.frags_min = DEFAULT_CPARAMS_FRAGS_MIN;
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cpars->buf.block.frags_max = DEFAULT_CPARAMS_FRAGS_MAX;
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}
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// This function waits until it is possible to write a full sound buffer
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static int QSA_WaitDevice(SDL_AudioDevice *device)
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{
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// Setup timeout for playing one fragment equal to 2 seconds
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// If timeout occurred than something wrong with hardware or driver
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// For example, Vortex 8820 audio driver stucks on second DAC because
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// it doesn't exist !
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const int result = SDL_IOReady(device->hidden->audio_fd,
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device->iscapture ? SDL_IOR_READ : SDL_IOR_WRITE,
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2 * 1000);
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switch (result) {
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case -1:
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SDL_LogError(SDL_LOG_CATEGORY_AUDIO, "QSA: SDL_IOReady() failed: %s", strerror(errno));
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return -1;
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case 0:
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device->hidden->timeout_on_wait = SDL_TRUE; // !!! FIXME: Should we just disconnect the device in this case?
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break;
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default:
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device->hidden->timeout_on_wait = SDL_FALSE;
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break;
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}
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return 0;
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}
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static int QSA_PlayDevice(SDL_AudioDevice *device, const Uint8 *buffer, int buflen)
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{
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if (SDL_AtomicGet(&device->shutdown) || !device->hidden) {
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return 0;
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}
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int towrite = buflen;
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// Write the audio data, checking for EAGAIN (buffer full) and underrun
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while ((towrite > 0) && !SDL_AtomicGet(&device->shutdown));
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const int bw = snd_pcm_plugin_write(device->hidden->audio_handle, buffer, towrite);
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if (bw != towrite) {
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// Check if samples playback got stuck somewhere in hardware or in the audio device driver
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if ((errno == EAGAIN) && (bw == 0)) {
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if (device->hidden->timeout_on_wait) {
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return 0; // oh well, try again next time. !!! FIXME: Should we just disconnect the device in this case?
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}
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}
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// Check for errors or conditions
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if ((errno == EAGAIN) || (errno == EWOULDBLOCK)) {
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SDL_Delay(1); // Let a little CPU time go by and try to write again
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// if we wrote some data
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towrite -= bw;
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buffer += bw * device->spec.channels;
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continue;
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} else if ((errno == EINVAL) || (errno == EIO)) {
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snd_pcm_channel_status_t cstatus;
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SDL_zero(cstatus);
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cstatus.channel = device->iscapture ? SND_PCM_CHANNEL_CAPTURE : SND_PCM_CHANNEL_PLAYBACK;
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int status = snd_pcm_plugin_status(device->hidden->audio_handle, &cstatus);
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if (status < 0) {
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QSA_SetError("snd_pcm_plugin_status", status);
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return -1;
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} else if ((cstatus.status == SND_PCM_STATUS_UNDERRUN) || (cstatus.status == SND_PCM_STATUS_READY)) {
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status = snd_pcm_plugin_prepare(device->hidden->audio_handle, device->iscapture ? SND_PCM_CHANNEL_CAPTURE : SND_PCM_CHANNEL_PLAYBACK);
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if (status < 0) {
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QSA_SetError("snd_pcm_plugin_prepare", status);
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return -1;
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}
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}
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continue;
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} else {
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return -1;
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}
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} else {
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// we wrote all remaining data
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towrite -= bw;
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buffer += bw * device->spec.channels;
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}
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}
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// If we couldn't write, assume fatal error for now
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return (towrite != 0) ? -1 : 0;
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}
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static Uint8 *QSA_GetDeviceBuf(SDL_AudioDevice *device, int *buffer_size)
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{
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return device->hidden->pcm_buf;
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}
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static void QSA_CloseDevice(SDL_AudioDevice *device)
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{
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if (device->hidden) {
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if (device->hidden->audio_handle != NULL) {
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#if _NTO_VERSION < 710
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// Finish playing available samples or cancel unread samples during capture
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snd_pcm_plugin_flush(device->hidden->audio_handle, device->iscapture ? SND_PCM_CHANNEL_CAPTURE : SND_PCM_CHANNEL_PLAYBACK);
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#endif
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snd_pcm_close(device->hidden->audio_handle);
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}
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SDL_free(device->hidden->pcm_buf);
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SDL_free(device->hidden);
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device->hidden = NULL;
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}
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}
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static int QSA_OpenDevice(SDL_AudioDevice *device)
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{
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if (device->iscapture) {
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return SDL_SetError("SDL capture support isn't available on QNX atm"); // !!! FIXME: most of this code has support for capture devices, but there's no CaptureFromDevice, etc functions. Fill them in!
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}
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SDL_assert(device->handle != NULL); // NULL used to mean "system default device" in SDL2; it does not mean that in SDL3.
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const Uint32 sdlhandle = (Uint32) ((size_t) device->handle);
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const uint32_t cardno = (uint32_t) (sdlhandle & 0xFFFF);
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const uint32_t deviceno = (uint32_t) ((sdlhandle >> 16) & 0xFFFF);
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const SDL_bool iscapture = device->iscapture;
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int status = 0;
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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 (struct SDL_PrivateAudioData)));
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if (device->hidden == NULL) {
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return SDL_OutOfMemory();
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}
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// Initialize channel transfer parameters to default
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snd_pcm_channel_params_t cparams;
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QSA_InitAudioParams(&cparams);
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// Open requested audio device
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status = snd_pcm_open(&device->hidden->audio_handle, cardno, deviceno, iscapture ? SND_PCM_OPEN_CAPTURE : SND_PCM_OPEN_PLAYBACK);
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if (status < 0) {
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device->hidden->audio_handle = NULL;
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return QSA_SetError("snd_pcm_open", status);
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}
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// Try for a closest match on audio format
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SDL_AudioFormat test_format = 0;
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const SDL_AudioFormat *closefmts = SDL_ClosestAudioFormats(device->spec.format);
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while ((test_format = *(closefmts++)) != 0) {
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// if match found set format to equivalent QSA format
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switch (test_format) {
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#define CHECKFMT(sdlfmt, qsafmt) case SDL_AUDIO_##sdlfmt: cparams.format.format = SND_PCM_SFMT_##qsafmt; break
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CHECKFMT(U8, U8);
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CHECKFMT(S8, S8);
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CHECKFMT(S16LSB, S16_LE);
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CHECKFMT(S16MSB, S16_BE);
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CHECKFMT(S32LSB, S32_LE);
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CHECKFMT(S32MSB, S32_BE);
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CHECKFMT(F32LSB, FLOAT_LE);
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CHECKFMT(F32MSB, FLOAT_BE);
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#undef CHECKFMT
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default: continue;
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}
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break;
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}
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// assumes test_format not 0 on success
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if (test_format == 0) {
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return SDL_SetError("QSA: Couldn't find any hardware audio formats");
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}
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device->spec.format = test_format;
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// Set mono/stereo/4ch/6ch/8ch audio
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cparams.format.voices = device->spec.channels;
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// Set rate
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cparams.format.rate = device->spec.freq;
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// Setup the transfer parameters according to cparams
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status = snd_pcm_plugin_params(device->hidden->audio_handle, &cparams);
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if (status < 0) {
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return QSA_SetError("snd_pcm_plugin_params", status);
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}
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// Make sure channel is setup right one last time
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snd_pcm_channel_setup_t csetup;
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SDL_zero(csetup);
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csetup.channel = iscapture ? SND_PCM_CHANNEL_CAPTURE : SND_PCM_CHANNEL_PLAYBACK;
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if (snd_pcm_plugin_setup(device->hidden->audio_handle, &csetup) < 0) {
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return SDL_SetError("QSA: Unable to setup channel");
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}
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device->sample_frames = csetup.buf.block.frag_size;
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// Calculate the final parameters for this audio specification
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SDL_UpdatedAudioDeviceFormat(device);
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device->hidden->pcm_buf = (Uint8 *) SDL_malloc(device->buffer_size);
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if (device->hidden->pcm_buf == NULL) {
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return SDL_OutOfMemory();
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}
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SDL_memset(device->hidden->pcm_buf, device->silence_value, device->buffer_size);
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// get the file descriptor
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device->hidden->audio_fd = snd_pcm_file_descriptor(device->hidden->audio_handle, csetup.channel);
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if (device->hidden->audio_fd < 0) {
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return QSA_SetError("snd_pcm_file_descriptor", device->hidden->audio_fd);
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}
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// Prepare an audio channel
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status = snd_pcm_plugin_prepare(device->hidden->audio_handle, csetup.channel)
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if (status < 0) {
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return QSA_SetError("snd_pcm_plugin_prepare", status);
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}
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return 0; // We're really ready to rock and roll. :-)
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}
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static SDL_AudioFormat QnxFormatToSDLFormat(const int32_t qnxfmt)
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{
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switch (qnxfmt) {
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#define CHECKFMT(sdlfmt, qsafmt) case SND_PCM_SFMT_##qsafmt: return SDL_AUDIO_##sdlfmt
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CHECKFMT(U8, U8);
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CHECKFMT(S8, S8);
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CHECKFMT(S16LSB, S16_LE);
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CHECKFMT(S16MSB, S16_BE);
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CHECKFMT(S32LSB, S32_LE);
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CHECKFMT(S32MSB, S32_BE);
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CHECKFMT(F32LSB, FLOAT_LE);
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CHECKFMT(F32MSB, FLOAT_BE);
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#undef CHECKFMT
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default: break;
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}
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return SDL_AUDIO_S16; // oh well.
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}
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static void QSA_DetectDevices(SDL_AudioDevice **default_output, SDL_AudioDevice **default_capture)
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{
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// Detect amount of available devices
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// this value can be changed in the runtime
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int num_cards = 0;
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(void) snd_cards_list(NULL, 0, &alloc_num_cards);
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SDL_bool isstack = SDL_FALSE;
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int *cards = SDL_small_alloc(int, num_cards, &isstack);
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if (!cards) {
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return; // we're in trouble.
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}
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int overflow_cards = 0;
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const int total_num_cards = snd_cards_list(cards, num_cards, &overflow_cards);
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// if overflow_cards > 0 or total_num_cards > num_cards, it changed at the last moment; oh well, we lost some.
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num_cards = SDL_min(num_cards, total_num_cards); // ...but make sure it didn't _shrink_.
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// If io-audio manager is not running we will get 0 as number of available audio devices
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if (num_cards == 0) { // not any available audio devices?
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SDL_small_free(cards, isstack);
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return;
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}
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// Find requested devices by type
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for (int it = 0; it < num_cards; it++) {
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const int card = cards[it];
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for (uint32_t deviceno = 0; ; deviceno++) {
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int32_t status;
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char name[QSA_MAX_NAME_LENGTH];
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status = snd_card_get_longname(card, name, sizeof (name));
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if (status == EOK) {
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snd_pcm_t *handle;
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// Add device number to device name
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char fullname[QSA_MAX_NAME_LENGTH + 32];
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SDL_snprintf(fullname, sizeof (fullname), "%s d%d", name, (int) deviceno);
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// Check if this device id could play anything
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SDL_bool iscapture = SDL_FALSE;
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status = snd_pcm_open(&handle, card, deviceno, SND_PCM_OPEN_PLAYBACK);
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if (status != EOK) { // no? See if it's a capture device instead.
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#if 0 // !!! FIXME: most of this code has support for capture devices, but there's no CaptureFromDevice, etc functions. Fill them in!
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status = snd_pcm_open(&handle, card, deviceno, SND_PCM_OPEN_CAPTURE);
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if (status == EOK) {
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iscapture = SDL_TRUE;
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}
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#endif
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}
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if (status == EOK) {
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SDL_AudioSpec spec;
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SDL_AudioSpec *pspec = &spec;
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snd_pcm_channel_setup_t csetup;
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SDL_zero(csetup);
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csetup.channel = iscapture ? SND_PCM_CHANNEL_CAPTURE : SND_PCM_CHANNEL_PLAYBACK;
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if (snd_pcm_plugin_setup(device->hidden->audio_handle, &csetup) < 0) {
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pspec = NULL; // go on without spec info.
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} else {
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spec.format = QnxFormatToSDLFormat(csetup.format.format);
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spec.channels = csetup.format.channels;
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spec.freq = csetup.format.rate;
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}
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status = snd_pcm_close(handle);
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if (status == EOK) {
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// !!! FIXME: I'm assuming each of these values are way less than 0xFFFF. Fix this if not.
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SDL_assert(card <= 0xFFFF);
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SDL_assert(deviceno <= 0xFFFF);
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const Uint32 sdlhandle = ((Uint32) card) | (((Uint32) deviceno) << 16);
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SDL_AddAudioDevice(iscapture, fullname, pspec, (void *) ((size_t) sdlhandle));
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}
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} else {
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// Check if we got end of devices list
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if (status == -ENOENT) {
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break;
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}
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}
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} else {
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break;
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}
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}
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}
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SDL_small_free(cards, isstack);
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// Try to open the "preferred" devices, which will tell us the card/device pairs for the default devices.
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snd_pcm_t handle;
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int cardno, deviceno;
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if (snd_pcm_open_preferred(&handle, &cardno, &deviceno, SND_PCM_OPEN_PLAYBACK) == 0) {
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snd_pcm_close(handle);
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// !!! FIXME: I'm assuming each of these values are way less than 0xFFFF. Fix this if not.
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SDL_assert(cardno <= 0xFFFF);
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SDL_assert(deviceno <= 0xFFFF);
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const Uint32 sdlhandle = ((Uint32) card) | (((Uint32) deviceno) << 16);
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*default_output = SDL_FindPhysicalAudioDeviceByHandle((void *) ((size_t) sdlhandle));
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}
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if (snd_pcm_open_preferred(&handle, &cardno, &deviceno, SND_PCM_OPEN_CAPTURE) == 0) {
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snd_pcm_close(handle);
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// !!! FIXME: I'm assuming each of these values are way less than 0xFFFF. Fix this if not.
|
||||
SDL_assert(cardno <= 0xFFFF);
|
||||
SDL_assert(deviceno <= 0xFFFF);
|
||||
const Uint32 sdlhandle = ((Uint32) card) | (((Uint32) deviceno) << 16);
|
||||
*default_capture = SDL_FindPhysicalAudioDeviceByHandle((void *) ((size_t) sdlhandle));
|
||||
}
|
||||
}
|
||||
|
||||
static void QSA_Deinitialize(void)
|
||||
{
|
||||
// nothing to do here atm.
|
||||
}
|
||||
|
||||
static SDL_bool QSA_Init(SDL_AudioDriverImpl * impl)
|
||||
{
|
||||
impl->DetectDevices = QSA_DetectDevices;
|
||||
impl->OpenDevice = QSA_OpenDevice;
|
||||
impl->ThreadInit = QSA_ThreadInit;
|
||||
impl->WaitDevice = QSA_WaitDevice;
|
||||
impl->PlayDevice = QSA_PlayDevice;
|
||||
impl->GetDeviceBuf = QSA_GetDeviceBuf;
|
||||
impl->CloseDevice = QSA_CloseDevice;
|
||||
impl->Deinitialize = QSA_Deinitialize;
|
||||
|
||||
// !!! FIXME: most of this code has support for capture devices, but there's no CaptureFromDevice, etc functions. Fill them in!
|
||||
//impl->HasCaptureSupport = SDL_TRUE;
|
||||
|
||||
return SDL_TRUE;
|
||||
}
|
||||
|
||||
AudioBootStrap QSAAUDIO_bootstrap = {
|
||||
"qsa", "QNX QSA Audio", QSA_Init, 0
|
||||
};
|
||||
|
||||
#endif // SDL_AUDIO_DRIVER_QNX
|
||||
|
||||
40
vendor/sdl-3.0.0/src/audio/qnx/SDL_qsa_audio.h
vendored
Normal file
40
vendor/sdl-3.0.0/src/audio/qnx/SDL_qsa_audio.h
vendored
Normal file
|
|
@ -0,0 +1,40 @@
|
|||
/*
|
||||
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"
|
||||
|
||||
#ifndef __SDL_QSA_AUDIO_H__
|
||||
#define __SDL_QSA_AUDIO_H__
|
||||
|
||||
#include <sys/asoundlib.h>
|
||||
|
||||
#include "../SDL_sysaudio.h"
|
||||
|
||||
struct SDL_PrivateAudioData
|
||||
{
|
||||
snd_pcm_t *audio_handle; // The audio device handle
|
||||
int audio_fd; // The audio file descriptor, for selecting on
|
||||
SDL_bool timeout_on_wait; // Select timeout status
|
||||
Uint8 *pcm_buf; // Raw mixing buffer
|
||||
};
|
||||
|
||||
#endif // __SDL_QSA_AUDIO_H__
|
||||
|
||||
Loading…
Add table
Add a link
Reference in a new issue