#if defined(USE_SDL_AUDIO) #include #include "audiostream.h" #include "stdclass.h" static SDL_AudioDeviceID audiodev; static bool needs_resampling; static cResetEvent read_wait; static cMutex stream_mutex; static struct { uint32_t prevs; uint32_t sample_buffer[2048]; } audiobuf; static unsigned sample_count = 0; // To easily access samples. union Sample { int16_t s[2]; uint32_t l; }; static float InterpolateCatmull4pt3oX(float x0, float x1, float x2, float x3, float t) { return 0.45 * ((2 * x1) + t * ((-x0 + x2) + t * ((2 * x0 - 5 * x1 + 4 * x2 - x3) + t * (-x0 + 3 * x1 - 3 * x2 + x3)))); } static void sdl2_audiocb(void* userdata, Uint8* stream, int len) { stream_mutex.Lock(); // Wait until there's enough samples to feed the kraken unsigned oslen = len / sizeof(uint32_t); unsigned islen = needs_resampling ? oslen * 16 / 17 : oslen; unsigned minlen = needs_resampling ? islen + 2 : islen; // Resampler looks ahead by 2 samples. if (sample_count < minlen) { // No data, just output a bit of silence for the underrun memset(stream, 0, len); stream_mutex.Unlock(); read_wait.Set(); return; } if (!needs_resampling) { // Just copy bytes for this case. memcpy(stream, &audiobuf.sample_buffer[0], len); } else { // 44.1KHz to 48KHz (actually 46.86KHz) resampling uint32_t *outbuf = (uint32_t*)stream; const float ra = 1.0f / 17; Sample *sbuf = (Sample*)&audiobuf.sample_buffer[0]; // [-1] stores the previous iteration last sample output for (u32 i = 0; i < islen/16; i++) { *outbuf++ = sbuf[i*16+ 0].l; // First sample stays at the same location. for (int k = 1; k < 17; k++) { Sample r; // Note we access offset -1 on first iteration, as to access prevs r.s[0] = InterpolateCatmull4pt3oX(sbuf[i*16+k-2].s[0], sbuf[i*16+k-1].s[0], sbuf[i*16+k].s[0], sbuf[i*16+k+1].s[0], 1 - ra*k); r.s[1] = InterpolateCatmull4pt3oX(sbuf[i*16+k-2].s[1], sbuf[i*16+k-1].s[1], sbuf[i*16+k].s[1], sbuf[i*16+k+1].s[1], 1 - ra*k); *outbuf++ = r.l; } } audiobuf.prevs = audiobuf.sample_buffer[islen-1]; } // Move samples in the buffer and consume them memmove(&audiobuf.sample_buffer[0], &audiobuf.sample_buffer[islen], (sample_count-islen)*sizeof(uint32_t)); sample_count -= islen; stream_mutex.Unlock(); read_wait.Set(); } static void sdl2_audio_init() { if (!SDL_WasInit(SDL_INIT_AUDIO)) SDL_InitSubSystem(SDL_INIT_AUDIO); // Support 44.1KHz (native) but also upsampling to 48KHz SDL_AudioSpec wav_spec, out_spec; memset(&wav_spec, 0, sizeof(wav_spec)); wav_spec.freq = 44100; wav_spec.format = AUDIO_S16; wav_spec.channels = 2; wav_spec.samples = 1024; // Must be power of two wav_spec.callback = sdl2_audiocb; // Try 44.1KHz which should be faster since it's native. audiodev = SDL_OpenAudioDevice(NULL, 0, &wav_spec, &out_spec, 0); if (!audiodev) { needs_resampling = true; wav_spec.freq = 48000; audiodev = SDL_OpenAudioDevice(NULL, 0, &wav_spec, &out_spec, 0); if (!audiodev) ERROR_LOG(AUDIO, "SDL2: SDL_OpenAudioDevice failed"); else INFO_LOG(AUDIO, "SDL2: Using resampling to 48 KHz"); } } static u32 sdl2_audio_push(const void* frame, u32 samples, bool wait) { // Unpause the device shall it be paused. if (SDL_GetAudioDeviceStatus(audiodev) != SDL_AUDIO_PLAYING) SDL_PauseAudioDevice(audiodev, 0); // If wait, then wait for the buffer to be smaller than a certain size. stream_mutex.Lock(); if (wait) { while (sample_count + samples > sizeof(audiobuf.sample_buffer)/sizeof(audiobuf.sample_buffer[0])) { stream_mutex.Unlock(); read_wait.Wait(); read_wait.Reset(); stream_mutex.Lock(); } } // Copy as many samples as possible, drop any remaining (this should not happen usually) unsigned free_samples = sizeof(audiobuf.sample_buffer) / sizeof(audiobuf.sample_buffer[0]) - sample_count; unsigned tocopy = samples < free_samples ? samples : free_samples; memcpy(&audiobuf.sample_buffer[sample_count], frame, tocopy * sizeof(uint32_t)); sample_count += tocopy; stream_mutex.Unlock(); return 1; } static void sdl2_audio_term() { if (audiodev) { // Stop audio playback. SDL_PauseAudioDevice(audiodev, 1); read_wait.Set(); SDL_CloseAudioDevice(audiodev); audiodev = SDL_AudioDeviceID(); } } static audiobackend_t audiobackend_sdl2audio = { "sdl2", // Slug "Simple DirectMedia Layer 2 Audio", // Name &sdl2_audio_init, &sdl2_audio_push, &sdl2_audio_term, NULL }; static bool sdl2audiobe = RegisterAudioBackend(&audiobackend_sdl2audio); #endif