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#include <alsa/asoundlib.h>
struct AudioALSA : Audio {
~AudioALSA() { term(); }
struct {
snd_pcm_t* handle = nullptr;
snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE;
snd_pcm_uframes_t buffer_size;
snd_pcm_uframes_t period_size;
int channels = 2;
const char* name = "default";
} device;
struct {
uint32_t* data = nullptr;
unsigned length = 0;
} buffer;
struct {
bool synchronize = false;
unsigned frequency = 22050;
unsigned latency = 60;
} settings;
auto cap(const string& name) -> bool {
if(name == Audio::Synchronize) return true;
if(name == Audio::Frequency) return true;
if(name == Audio::Latency) return true;
return false;
}
auto get(const string& name) -> any {
if(name == Audio::Synchronize) return settings.synchronize;
if(name == Audio::Frequency) return settings.frequency;
if(name == Audio::Latency) return settings.latency;
return {};
}
auto set(const string& name, const any& value) -> bool {
if(name == Audio::Synchronize && value.is<bool>()) {
if(settings.synchronize != value.get<bool>()) {
settings.synchronize = value.get<bool>();
if(device.handle) init();
}
return true;
}
if(name == Audio::Frequency && value.is<unsigned>()) {
if(settings.frequency != value.get<unsigned>()) {
settings.frequency = value.get<unsigned>();
if(device.handle) init();
}
return true;
}
if(name == Audio::Latency && value.is<unsigned>()) {
if(settings.latency != value.get<unsigned>()) {
settings.latency = value.get<unsigned>();
if(device.handle) init();
}
return true;
}
return false;
}
auto sample(uint16_t left, uint16_t right) -> void {
if(!device.handle) return;
buffer.data[buffer.length++] = left + (right << 16);
if(buffer.length < device.period_size) return;
snd_pcm_sframes_t avail;
do {
avail = snd_pcm_avail_update(device.handle);
if(avail < 0) snd_pcm_recover(device.handle, avail, 1);
if(avail < buffer.length) {
if(settings.synchronize == false) {
buffer.length = 0;
return;
}
int error = snd_pcm_wait(device.handle, -1);
if(error < 0) snd_pcm_recover(device.handle, error, 1);
}
} while(avail < buffer.length);
//below code has issues with PulseAudio sound server
#if 0
if(settings.synchronize == false) {
snd_pcm_sframes_t avail = snd_pcm_avail_update(device.handle);
if(avail < device.period_size) {
buffer.length = 0;
return;
}
}
#endif
uint32_t* buffer_ptr = buffer.data;
int i = 4;
while((buffer.length > 0) && i--) {
snd_pcm_sframes_t written = snd_pcm_writei(device.handle, buffer_ptr, buffer.length);
if(written < 0) {
//no samples written
snd_pcm_recover(device.handle, written, 1);
} else if(written <= buffer.length) {
buffer.length -= written;
buffer_ptr += written;
}
}
if(i < 0) {
if(buffer.data == buffer_ptr) {
buffer.length--;
buffer_ptr++;
}
memmove(buffer.data, buffer_ptr, buffer.length * sizeof(uint32_t));
}
}
auto clear() -> void {
}
auto init() -> bool {
if(snd_pcm_open(&device.handle, device.name, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK) < 0) {
term();
return false;
}
//below code will not work with 24khz frequency rate (ALSA library bug)
#if 0
if(snd_pcm_set_params(device.handle, device.format, SND_PCM_ACCESS_RW_INTERLEAVED,
device.channels, settings.frequency, 1, settings.latency * 1000) < 0) {
//failed to set device parameters
term();
return false;
}
if(snd_pcm_get_params(device.handle, &device.buffer_size, &device.period_size) < 0) {
device.period_size = settings.latency * 1000 * 1e-6 * settings.frequency / 4;
}
#endif
snd_pcm_hw_params_t* hwparams;
snd_pcm_sw_params_t* swparams;
unsigned rate = settings.frequency;
unsigned buffer_time = settings.latency * 1000;
unsigned period_time = settings.latency * 1000 / 4;
snd_pcm_hw_params_alloca(&hwparams);
if(snd_pcm_hw_params_any(device.handle, hwparams) < 0) {
term();
return false;
}
if(snd_pcm_hw_params_set_access(device.handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED) < 0
|| snd_pcm_hw_params_set_format(device.handle, hwparams, device.format) < 0
|| snd_pcm_hw_params_set_channels(device.handle, hwparams, device.channels) < 0
|| snd_pcm_hw_params_set_rate_near(device.handle, hwparams, &rate, 0) < 0
|| snd_pcm_hw_params_set_period_time_near(device.handle, hwparams, &period_time, 0) < 0
|| snd_pcm_hw_params_set_buffer_time_near(device.handle, hwparams, &buffer_time, 0) < 0
) {
term();
return false;
}
if(snd_pcm_hw_params(device.handle, hwparams) < 0) {
term();
return false;
}
if(snd_pcm_get_params(device.handle, &device.buffer_size, &device.period_size) < 0) {
term();
return false;
}
snd_pcm_sw_params_alloca(&swparams);
if(snd_pcm_sw_params_current(device.handle, swparams) < 0) {
term();
return false;
}
if(snd_pcm_sw_params_set_start_threshold(device.handle, swparams,
(device.buffer_size / device.period_size) * device.period_size) < 0
) {
term();
return false;
}
if(snd_pcm_sw_params(device.handle, swparams) < 0) {
term();
return false;
}
buffer.data = new uint32_t[device.period_size];
return true;
}
auto term() -> void {
if(device.handle) {
//snd_pcm_drain(device.handle); //prevents popping noise; but causes multi-second lag
snd_pcm_close(device.handle);
device.handle = 0;
}
if(buffer.data) {
delete[] buffer.data;
buffer.data = 0;
}
}
};
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