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Audio: Write to device's buffer directly without temporary buffers. #90013

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Audio: Write to device's buffer directly without temporary buffers. #90013

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20 changes: 13 additions & 7 deletions drivers/SCsub
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Original file line number Diff line number Diff line change
Expand Up @@ -9,15 +9,19 @@ SConscript("unix/SCsub")
SConscript("windows/SCsub")

# Sounds drivers
SConscript("alsa/SCsub")
SConscript("coreaudio/SCsub")
SConscript("pulseaudio/SCsub")
if env["platform"] in ["ios", "macos"]:
SConscript("coreaudio/SCsub")

if env["platform"] == "linuxbsd":
if env["alsa"]:
SConscript("alsa/SCsub")
if env["pulseaudio"]:
SConscript("pulseaudio/SCsub")

if env["platform"] == "windows":
if env["xaudio2"]:
SConscript("xaudio2/SCsub")
SConscript("wasapi/SCsub")
Comment on lines +12 to 24
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@kus04e4ek kus04e4ek Apr 29, 2024 •
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I think it's better this way, as no empty files will be compiled. Maybe it would be better to add checks like this to other drivers? Also should I delete checks like #ifdef ALSA_ENABLED in files?

if not env.msvc:
SConscript("backtrace/SCsub")
if env["xaudio2"]:
SConscript("xaudio2/SCsub")

# Midi drivers
SConscript("alsamidi/SCsub")
Expand All @@ -36,6 +40,8 @@ if env["opengl3"]:

# Core dependencies
SConscript("png/SCsub")
if env["platform"] == "windows" and not env.msvc:
SConscript("backtrace/SCsub")
Comment on lines +43 to +44
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Moved out of Sounds drivers section


env.add_source_files(env.drivers_sources, "*.cpp")

Expand Down
5 changes: 2 additions & 3 deletions drivers/alsa/SCsub
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Expand Up @@ -2,8 +2,7 @@

Import("env")

if "alsa" in env and env["alsa"]:
if env["use_sowrap"]:
env.add_source_files(env.drivers_sources, "asound-so_wrap.c")
if env["use_sowrap"]:
env.add_source_files(env.drivers_sources, "asound-so_wrap.c")

env.add_source_files(env.drivers_sources, "*.cpp")
79 changes: 30 additions & 49 deletions drivers/alsa/audio_driver_alsa.cpp
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Expand Up @@ -46,10 +46,11 @@ extern int initialize_pulse(int verbose);
Error AudioDriverALSA::init_output_device() {
mix_rate = _get_configured_mix_rate();

speaker_mode = SPEAKER_MODE_STEREO;
// TODO: `channels` and `buffer_format` are hardcoded.
channels = 2;
buffer_format = BUFFER_FORMAT_INTEGER_16;

// If there is a specified output device check that it is really present
// If there is a specified output device check that it is really present.
if (output_device_name != "Default") {
PackedStringArray list = get_output_device_list();
if (list.find(output_device_name) == -1) {
Expand All @@ -62,20 +63,15 @@ Error AudioDriverALSA::init_output_device() {
snd_pcm_hw_params_t *hwparams;
snd_pcm_sw_params_t *swparams;

#define CHECK_FAIL(m_cond) \
if (m_cond) { \
fprintf(stderr, "ALSA ERR: %s\n", snd_strerror(status)); \
if (pcm_handle) { \
snd_pcm_close(pcm_handle); \
pcm_handle = nullptr; \
} \
ERR_FAIL_COND_V(m_cond, ERR_CANT_OPEN); \
#define CHECK_FAIL(m_cond) \
if (m_cond) { \
if (pcm_handle) { \
snd_pcm_close(pcm_handle); \
pcm_handle = nullptr; \
} \
ERR_FAIL_V_MSG(ERR_CANT_OPEN, vformat("ALSA: %s", snd_strerror(status))); \
}

//todo, add
//6 chans - "plug:surround51"
//4 chans - "plug:surround40";

if (output_device_name == "Default") {
status = snd_pcm_open(&pcm_handle, "default", SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
} else {
Expand All @@ -97,48 +93,43 @@ Error AudioDriverALSA::init_output_device() {
status = snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
CHECK_FAIL(status < 0);

//not interested in anything else
status = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE);
CHECK_FAIL(status < 0);

//todo: support 4 and 6
status = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, 2);
status = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, channels);
CHECK_FAIL(status < 0);

status = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &mix_rate, nullptr);
CHECK_FAIL(status < 0);

// In ALSA the period size seems to be the one that will determine the actual latency
// Ref: https://www.alsa-project.org/main/index.php/FramesPeriods
// In ALSA the period size seems to be the one that will determine the actual latency.
// Ref: https://www.alsa-project.org/main/index.php/FramesPeriods.
unsigned int periods = 2;
int latency = Engine::get_singleton()->get_audio_output_latency();
buffer_frames = closest_power_of_2(latency * mix_rate / 1000);
buffer_size = buffer_frames * periods;
period_size = buffer_frames;

// set buffer size from project settings
// Set buffer size from project settings.
snd_pcm_uframes_t buffer_size = buffer_frames * periods;
status = snd_pcm_hw_params_set_buffer_size_near(pcm_handle, hwparams, &buffer_size);
CHECK_FAIL(status < 0);

status = snd_pcm_hw_params_set_period_size_near(pcm_handle, hwparams, &period_size, nullptr);
status = snd_pcm_hw_params_set_period_size_near(pcm_handle, hwparams, &buffer_frames, nullptr);
CHECK_FAIL(status < 0);

print_verbose("Audio buffer frames: " + itos(period_size) + " calculated latency: " + itos(period_size * 1000 / mix_rate) + "ms");
print_verbose("Audio buffer frames: " + itos(buffer_frames) + " calculated latency: " + itos(buffer_frames * 1000 / mix_rate) + "ms");

status = snd_pcm_hw_params_set_periods_near(pcm_handle, hwparams, &periods, nullptr);
CHECK_FAIL(status < 0);

status = snd_pcm_hw_params(pcm_handle, hwparams);
CHECK_FAIL(status < 0);

//snd_pcm_hw_params_free(&hwparams);

snd_pcm_sw_params_alloca(&swparams);

status = snd_pcm_sw_params_current(pcm_handle, swparams);
CHECK_FAIL(status < 0);

status = snd_pcm_sw_params_set_avail_min(pcm_handle, swparams, period_size);
status = snd_pcm_sw_params_set_avail_min(pcm_handle, swparams, buffer_frames);
CHECK_FAIL(status < 0);

status = snd_pcm_sw_params_set_start_threshold(pcm_handle, swparams, 1);
Expand All @@ -147,9 +138,7 @@ Error AudioDriverALSA::init_output_device() {
status = snd_pcm_sw_params(pcm_handle, swparams);
CHECK_FAIL(status < 0);

samples_in.resize(period_size * channels);
samples_out.resize(period_size * channels);

samples_out.resize(buffer_frames * channels * get_size_of_sample(buffer_format));
return OK;
}

Expand All @@ -162,7 +151,7 @@ Error AudioDriverALSA::init() {
#endif
#ifdef PULSEAUDIO_ENABLED
// On pulse enabled systems Alsa will silently use pulse.
// It doesn't matter if this fails as that likely means there is no pulse
// It doesn't matter if this fails as that likely means there is no pulse.
initialize_pulse(dylibloader_verbose);
#endif

Expand Down Expand Up @@ -200,28 +189,16 @@ void AudioDriverALSA::thread_func(void *p_udata) {
ad->lock();
ad->start_counting_ticks();

if (!ad->active.is_set()) {
for (uint64_t i = 0; i < ad->period_size * ad->channels; i++) {
ad->samples_out.write[i] = 0;
}

} else {
ad->audio_server_process(ad->period_size, ad->samples_in.ptrw());
ad->audio_server_process(ad->buffer_frames, ad->samples_out.ptr(), ad->active.is_set());

for (uint64_t i = 0; i < ad->period_size * ad->channels; i++) {
ad->samples_out.write[i] = ad->samples_in[i] >> 16;
}
}

int todo = ad->period_size;
int todo = ad->buffer_frames;
int total = 0;

while (todo && !ad->exit_thread.is_set()) {
int16_t *src = (int16_t *)ad->samples_out.ptr();
int wrote = snd_pcm_writei(ad->pcm_handle, (void *)(src + (total * ad->channels)), todo);
int wrote = snd_pcm_writei(ad->pcm_handle, ad->samples_out.ptr() + total, todo);

if (wrote > 0) {
total += wrote;
total += wrote * ad->channels * AudioDriver::get_size_of_sample(ad->buffer_format);
todo -= wrote;
} else if (wrote == -EAGAIN) {
ad->stop_counting_ticks();
Expand Down Expand Up @@ -273,8 +250,12 @@ int AudioDriverALSA::get_mix_rate() const {
return mix_rate;
}

AudioDriver::SpeakerMode AudioDriverALSA::get_speaker_mode() const {
return speaker_mode;
int AudioDriverALSA::get_output_channels() const {
return channels;
}

AudioDriver::BufferFormat AudioDriverALSA::get_output_buffer_format() const {
return buffer_format;
}

PackedStringArray AudioDriverALSA::get_output_device_list() {
Expand Down
27 changes: 12 additions & 15 deletions drivers/alsa/audio_driver_alsa.h
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Expand Up @@ -53,25 +53,23 @@ class AudioDriverALSA : public AudioDriver {
String output_device_name = "Default";
String new_output_device = "Default";

Vector<int32_t> samples_in;
Vector<int16_t> samples_out;

Error init_output_device();
void finish_output_device();

static void thread_func(void *p_udata);
LocalVector<int8_t> samples_out;

unsigned int mix_rate = 0;
SpeakerMode speaker_mode;
int channels = 0;

BufferFormat buffer_format = NO_BUFFER;

snd_pcm_uframes_t buffer_frames;
snd_pcm_uframes_t buffer_size;
snd_pcm_uframes_t period_size;
int channels = 0;

SafeFlag active;
SafeFlag exit_thread;

Error init_output_device();
void finish_output_device();

static void thread_func(void *p_udata);

public:
virtual const char *get_name() const override {
return "ALSA";
Expand All @@ -80,18 +78,17 @@ class AudioDriverALSA : public AudioDriver {
virtual Error init() override;
virtual void start() override;
virtual int get_mix_rate() const override;
virtual SpeakerMode get_speaker_mode() const override;

virtual void lock() override;
virtual void unlock() override;
virtual void finish() override;

virtual int get_output_channels() const override;
virtual BufferFormat get_output_buffer_format() const override;

virtual PackedStringArray get_output_device_list() override;
virtual String get_output_device() override;
virtual void set_output_device(const String &p_name) override;

AudioDriverALSA() {}
~AudioDriverALSA() {}
};

#endif // ALSA_ENABLED
Expand Down
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