231 lines
6.6 KiB
C++
231 lines
6.6 KiB
C++
#if defined(USE_SDL_AUDIO)
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#include <SDL.h>
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#include "audiostream.h"
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#include "stdclass.h"
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#include <mutex>
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#include <atomic>
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static SDL_AudioDeviceID audiodev;
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static bool needs_resampling;
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static cResetEvent read_wait;
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static std::mutex stream_mutex;
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static struct {
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uint32_t prevs;
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uint32_t *sample_buffer;
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} audiobuf;
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static unsigned sample_count = 0;
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static SDL_AudioDeviceID recorddev;
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u8 recordbuf[480 * 4];
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std::atomic<size_t> rec_read;
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std::atomic<size_t> rec_write;
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// To easily access samples.
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union Sample { int16_t s[2]; uint32_t l; };
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static float InterpolateCatmull4pt3oX(float x0, float x1, float x2, float x3, float t) {
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return 0.45 * ((2 * x1) + t * ((-x0 + x2) + t * ((2 * x0 - 5 * x1 + 4 * x2 - x3) + t * (-x0 + 3 * x1 - 3 * x2 + x3))));
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}
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static void sdl2_audiocb(void* userdata, Uint8* stream, int len) {
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stream_mutex.lock();
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// Wait until there's enough samples to feed the kraken
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unsigned oslen = len / sizeof(uint32_t);
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unsigned islen = needs_resampling ? oslen * 16 / 17 : oslen;
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unsigned minlen = needs_resampling ? islen + 2 : islen; // Resampler looks ahead by 2 samples.
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if (sample_count < minlen) {
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// No data, just output a bit of silence for the underrun
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memset(stream, 0, len);
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stream_mutex.unlock();
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read_wait.Set();
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return;
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}
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if (!needs_resampling) {
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// Just copy bytes for this case.
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memcpy(stream, &audiobuf.sample_buffer[0], len);
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}
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else {
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// 44.1KHz to 48KHz (actually 46.86KHz) resampling
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uint32_t *outbuf = (uint32_t*)stream;
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const float ra = 1.0f / 17;
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Sample *sbuf = (Sample*)&audiobuf.sample_buffer[0]; // [-1] stores the previous iteration last sample output
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for (u32 i = 0; i < islen/16; i++) {
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*outbuf++ = sbuf[i*16+ 0].l; // First sample stays at the same location.
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for (int k = 1; k < 17; k++) {
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Sample r;
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// Note we access offset -1 on first iteration, as to access prevs
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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);
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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);
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*outbuf++ = r.l;
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}
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}
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audiobuf.prevs = audiobuf.sample_buffer[islen-1];
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}
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// Move samples in the buffer and consume them
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memmove(&audiobuf.sample_buffer[0], &audiobuf.sample_buffer[islen], (sample_count-islen)*sizeof(uint32_t));
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sample_count -= islen;
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stream_mutex.unlock();
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read_wait.Set();
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}
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static void sdl2_audio_init() {
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if (!SDL_WasInit(SDL_INIT_AUDIO))
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{
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if (SDL_InitSubSystem(SDL_INIT_AUDIO))
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ERROR_LOG(AUDIO, "SDL2 error initializing audio subsystem: %s", SDL_GetError());
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}
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audiobuf.sample_buffer = new uint32_t[config::AudioBufferSize]();
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// Support 44.1KHz (native) but also upsampling to 48KHz
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SDL_AudioSpec wav_spec, out_spec;
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memset(&wav_spec, 0, sizeof(wav_spec));
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wav_spec.freq = 44100;
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wav_spec.format = AUDIO_S16;
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wav_spec.channels = 2;
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wav_spec.samples = SAMPLE_COUNT * 2; // Must be power of two
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wav_spec.callback = sdl2_audiocb;
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// Try 44.1KHz which should be faster since it's native.
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audiodev = SDL_OpenAudioDevice(NULL, 0, &wav_spec, &out_spec, 0);
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if (!audiodev)
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{
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WARN_LOG(AUDIO, "SDL2: SDL_OpenAudioDevice failed: %s", SDL_GetError());
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needs_resampling = true;
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wav_spec.freq = 48000;
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audiodev = SDL_OpenAudioDevice(NULL, 0, &wav_spec, &out_spec, 0);
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if (!audiodev)
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ERROR_LOG(AUDIO, "SDL2: SDL_OpenAudioDevice failed: %s", SDL_GetError());
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else
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INFO_LOG(AUDIO, "SDL2: Using resampling to 48 KHz");
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}
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}
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static u32 sdl2_audio_push(const void* frame, u32 samples, bool wait) {
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// Unpause the device shall it be paused.
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if (SDL_GetAudioDeviceStatus(audiodev) != SDL_AUDIO_PLAYING)
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SDL_PauseAudioDevice(audiodev, 0);
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// If wait, then wait for the buffer to be smaller than a certain size.
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stream_mutex.lock();
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if (wait) {
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while (sample_count + samples > (u32)config::AudioBufferSize) {
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stream_mutex.unlock();
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read_wait.Wait();
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stream_mutex.lock();
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}
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}
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// Copy as many samples as possible, drop any remaining (this should not happen usually)
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unsigned free_samples = config::AudioBufferSize - sample_count;
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unsigned tocopy = samples < free_samples ? samples : free_samples;
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memcpy(&audiobuf.sample_buffer[sample_count], frame, tocopy * sizeof(uint32_t));
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sample_count += tocopy;
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stream_mutex.unlock();
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return 1;
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}
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static void sdl2_audio_term() {
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if (audiodev)
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{
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// Stop audio playback.
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SDL_PauseAudioDevice(audiodev, 1);
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read_wait.Set();
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SDL_CloseAudioDevice(audiodev);
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audiodev = SDL_AudioDeviceID();
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}
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delete [] audiobuf.sample_buffer;
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audiobuf.sample_buffer = nullptr;
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}
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void sdl2_record_cb(void *userdata, u8 *stream, int len)
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{
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DEBUG_LOG(AUDIO, "SDL2: sdl2_record_cb len %d write %zd read %zd", len, (size_t)rec_write, (size_t)rec_read);
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while (len > 0)
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{
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size_t plen = std::min((size_t)len, sizeof(recordbuf) - rec_write);
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memcpy(&recordbuf[rec_write], stream, plen);
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len -= plen;
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rec_write = (rec_write + plen) % sizeof(recordbuf);
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stream += plen;
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}
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}
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static bool sdl2_record_init(u32 sampling_freq)
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{
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rec_write = 0;
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rec_read = 0;
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SDL_AudioSpec wav_spec, out_spec;
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memset(&wav_spec, 0, sizeof(wav_spec));
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wav_spec.freq = sampling_freq;
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wav_spec.format = AUDIO_S16;
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wav_spec.channels = 1;
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wav_spec.samples = 256; // Must be power of two
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wav_spec.callback = sdl2_record_cb;
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recorddev = SDL_OpenAudioDevice(NULL, 1, &wav_spec, &out_spec, 0);
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if (recorddev == 0)
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{
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INFO_LOG(AUDIO, "SDL2: Cannot open audio capture device: %s", SDL_GetError());
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return false;
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}
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SDL_PauseAudioDevice(recorddev, 0);
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INFO_LOG(AUDIO, "SDL2: opened audio capture device");
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return true;
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}
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static void sdl2_record_term()
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{
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if (recorddev != 0)
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{
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SDL_PauseAudioDevice(recorddev, 1);
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SDL_CloseAudioDevice(recorddev);
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recorddev = 0;
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}
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}
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static u32 sdl2_record(void* frame, u32 samples)
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{
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u32 count = 0;
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samples *= 2;
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while (samples > 0)
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{
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u32 avail = std::min(rec_write - rec_read, sizeof(recordbuf) - rec_read);
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if (avail == 0)
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break;
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avail = std::min(avail, samples);
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memcpy((u8 *)frame + count, &recordbuf[rec_read], avail);
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rec_read = (rec_read + avail) % sizeof(recordbuf);
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samples -= avail;
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count += avail;
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}
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DEBUG_LOG(AUDIO, "SDL2: sdl2_record len %d ret %d write %zd read %zd", samples * 2, count, (size_t)rec_write, (size_t)rec_read);
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return count / 2;
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}
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static audiobackend_t audiobackend_sdl2audio = {
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"sdl2", // Slug
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"Simple DirectMedia Layer 2 Audio", // Name
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&sdl2_audio_init,
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&sdl2_audio_push,
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&sdl2_audio_term,
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NULL,
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&sdl2_record_init,
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&sdl2_record,
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&sdl2_record_term
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};
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static bool sdl2audiobe = RegisterAudioBackend(&audiobackend_sdl2audio);
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#endif
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