bsnes/ruby/audio/pulseaudio.cpp

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#include <pulse/pulseaudio.h>
struct AudioPulseAudio : Audio {
~AudioPulseAudio() { term(); }
struct {
pa_mainloop* mainloop = nullptr;
pa_context* context = nullptr;
pa_stream* stream = nullptr;
pa_sample_spec spec;
pa_buffer_attr buffer_attr;
bool first;
} device;
struct {
uint32_t* data = nullptr;
size_t size;
unsigned offset;
} buffer;
struct {
bool synchronize = false;
Update to v098r01 release. byuu says: Changelog: - SFC: balanced profile removed - SFC: performance profile removed - SFC: code for handling non-threaded CPU, SMP, DSP, PPU removed - SFC: Coprocessor, Controller (and expansion port) shared Thread code merged to SFC::Cothread - Cothread here just means "Thread with CPU affinity" (couldn't think of a better name, sorry) - SFC: CPU now has vector<Thread*> coprocessors, peripherals; - this is the beginning of work to allow expansion port devices to be dynamically changed at run-time - ruby: all audio drivers default to 48000hz instead of 22050hz now if no frequency is assigned - note: the WASAPI driver can default to whatever the native frequency is; doesn't have to be 48000hz - tomoko: removed the ability to change the frequency from the UI (but it will display the frequency used) - tomoko: removed the timing settings panel - the goal is to work toward smooth video via adaptive sync - the model is broken by not being in control of the audio frequency anyway - it's further broken by PAL running at 50hz and WSC running at 75hz - it was always broken anyway by SNES interlace timing varying from progressive timing - higan: audio/ stub created (for now, it's just nall/dsp/ moved here and included as a header) - higan: video/ stub created - higan/GNUmakefile: now includes build rules for essential components (libco, emulator, audio, video) The audio changes are in preparation to merge wareya's awesome WASAPI work without the need for the nall/dsp resampler.
2016-04-09 03:40:12 +00:00
unsigned frequency = 48000;
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>()) {
settings.synchronize = value.get<bool>();
return true;
}
if(name == Audio::Frequency && value.is<unsigned>()) {
settings.frequency = value.get<unsigned>();
if(device.stream) {
pa_operation_unref(pa_stream_update_sample_rate(device.stream, settings.frequency, NULL, NULL));
}
return true;
}
if(name == Audio::Latency && value.is<unsigned>()) {
settings.latency = value.get<unsigned>();
if(device.stream) {
device.buffer_attr.tlength = pa_usec_to_bytes(settings.latency * PA_USEC_PER_MSEC, &device.spec);
pa_stream_set_buffer_attr(device.stream, &device.buffer_attr, NULL, NULL);
}
return true;
}
return false;
}
auto sample(int16_t left, int16_t right) -> void {
pa_stream_begin_write(device.stream, (void**)&buffer.data, &buffer.size);
buffer.data[buffer.offset++] = (uint16_t)left << 0 | (uint16_t)right << 16;
if((buffer.offset + 1) * pa_frame_size(&device.spec) <= buffer.size) return;
while(true) {
if(device.first) {
device.first = false;
pa_mainloop_iterate(device.mainloop, 0, NULL);
} else {
pa_mainloop_iterate(device.mainloop, 1, NULL);
}
unsigned length = pa_stream_writable_size(device.stream);
if(length >= buffer.offset * pa_frame_size(&device.spec)) break;
if(settings.synchronize == false) {
buffer.offset = 0;
return;
}
}
pa_stream_write(device.stream, (const void*)buffer.data, buffer.offset * pa_frame_size(&device.spec), NULL, 0LL, PA_SEEK_RELATIVE);
buffer.data = 0;
buffer.offset = 0;
}
auto clear() -> void {
}
auto init() -> bool {
device.mainloop = pa_mainloop_new();
device.context = pa_context_new(pa_mainloop_get_api(device.mainloop), "ruby::pulseaudio");
pa_context_connect(device.context, NULL, PA_CONTEXT_NOFLAGS, NULL);
pa_context_state_t cstate;
do {
pa_mainloop_iterate(device.mainloop, 1, NULL);
cstate = pa_context_get_state(device.context);
if(!PA_CONTEXT_IS_GOOD(cstate)) return false;
} while(cstate != PA_CONTEXT_READY);
device.spec.format = PA_SAMPLE_S16LE;
device.spec.channels = 2;
device.spec.rate = settings.frequency;
device.stream = pa_stream_new(device.context, "audio", &device.spec, NULL);
device.buffer_attr.maxlength = -1;
device.buffer_attr.tlength = pa_usec_to_bytes(settings.latency * PA_USEC_PER_MSEC, &device.spec);
device.buffer_attr.prebuf = -1;
device.buffer_attr.minreq = -1;
device.buffer_attr.fragsize = -1;
pa_stream_flags_t flags = (pa_stream_flags_t)(PA_STREAM_ADJUST_LATENCY | PA_STREAM_VARIABLE_RATE);
pa_stream_connect_playback(device.stream, NULL, &device.buffer_attr, flags, NULL, NULL);
pa_stream_state_t sstate;
do {
pa_mainloop_iterate(device.mainloop, 1, NULL);
sstate = pa_stream_get_state(device.stream);
if(!PA_STREAM_IS_GOOD(sstate)) return false;
} while(sstate != PA_STREAM_READY);
buffer.size = 960;
buffer.offset = 0;
device.first = true;
return true;
}
auto term() -> void {
if(buffer.data) {
pa_stream_cancel_write(device.stream);
buffer.data = nullptr;
}
if(device.stream) {
pa_stream_disconnect(device.stream);
pa_stream_unref(device.stream);
device.stream = nullptr;
}
if(device.context) {
pa_context_disconnect(device.context);
pa_context_unref(device.context);
device.context = nullptr;
}
if(device.mainloop) {
pa_mainloop_free(device.mainloop);
device.mainloop = nullptr;
}
}
};