Add options for buffer size and headroom in Frontend options

workspace/all/common/api.c

@@ -2275,6 +2275,9 @@ SDL_Color GFX_mapColor(uint32_t c)
 #ifndef SAMPLES
 #define SAMPLES 512 // default
 #endif
+#ifndef BUFFER_MULTIPLIER
+#define BUFFER_MULTIPLIER 8 // default
+#endif
 
 #define ms SDL_GetTicks
 
@@ -2343,6 +2346,33 @@ void SND_setQuality(int quality)
 	soundQuality = qualityLevels[quality];
 	resetSrcState = 1;
 }
+
+static int audio_buffer_samples = SAMPLES;
+static int audio_buffer_multiplier = BUFFER_MULTIPLIER;
+
+void SND_setBufferSize(int samples) {
+	if (samples != audio_buffer_samples) {
+		audio_buffer_samples = samples;
+		LOG_info("Audio buffer size set to %d samples\n", samples);
+		if (snd.initialized) {
+			SND_resetAudio(snd.sample_rate_in, snd.frame_rate);
+		}
+	}
+}
+
+void SND_setBufferMultiplier(int multiplier) {
+	if (multiplier != audio_buffer_multiplier) {
+		audio_buffer_multiplier = multiplier;
+		if (snd.initialized && snd.sample_rate_out > 0) {
+			snd.frame_count = ((float)snd.sample_rate_out / SCREEN_FPS) * audio_buffer_multiplier;
+			perf.buffer_size = snd.frame_count;
+			SND_resizeBuffer();
+			SND_pauseAudio(true);
+			LOG_info("Audio buffer multiplier set to %d, new frame_count: %d\n", multiplier, snd.frame_count);
+		}
+	}
+}
+
 ResampledFrames resample_audio(const SND_Frame *input_frames,
 							   int input_frame_count, int input_sample_rate,
 							   int output_sample_rate, double ratio)
@@ -2796,7 +2826,7 @@ void SND_init(double sample_rate, double frame_rate)
 	spec_in.freq = PLAT_pickSampleRate(sample_rate, MAX_SAMPLE_RATE);
 	spec_in.format = AUDIO_S16;
 	spec_in.channels = 2;
-	spec_in.samples = SAMPLES;
+	spec_in.samples = audio_buffer_samples;
 	spec_in.callback = SND_audioCallback;
 
 #if defined(USE_SDL2)
@@ -2821,7 +2851,7 @@ void SND_init(double sample_rate, double frame_rate)
 
 	LOG_info("We now have audio device #%d\n", snd.device_id);
 
-	snd.frame_count = ((float)spec_out.freq / SCREEN_FPS) * 8; // buffer size based on sample rate out (times 12 samples headroom)
+	snd.frame_count = ((float)spec_out.freq / SCREEN_FPS) * audio_buffer_multiplier;
 	perf.buffer_size = snd.frame_count;
 	snd.sample_rate_in = sample_rate;
 	snd.sample_rate_out = spec_out.freq;
@@ -2832,7 +2862,7 @@ void SND_init(double sample_rate, double frame_rate)
 
 	// start with audiodevice paused so buffer can fill a little, snd_batchsamples will unpause it
 	SND_pauseAudio(true);
-	LOG_info("sample rate: %i (req) %i (rec) [samples %i]\n", snd.sample_rate_in, snd.sample_rate_out, SAMPLES);
+	LOG_info("sample rate: %i (req) %i (rec) [samples %i]\n", snd.sample_rate_in, snd.sample_rate_out, audio_buffer_samples);
 	snd.initialized = 1;
 
 }
@@ -4388,4 +4418,4 @@ FALLBACK_IMPLEMENTATION void PLAT_bluetoothStreamBegin(int buffersize) {}
 FALLBACK_IMPLEMENTATION void PLAT_bluetoothStreamEnd() {}
 FALLBACK_IMPLEMENTATION void PLAT_bluetoothStreamQuit() {}
 FALLBACK_IMPLEMENTATION int PLAT_bluetoothVolume() { return 100; }
-FALLBACK_IMPLEMENTATION void PLAT_bluetoothSetVolume(int vol) {}
+FALLBACK_IMPLEMENTATION void PLAT_bluetoothSetVolume(int vol) {}

workspace/all/common/api.h

@@ -413,6 +413,8 @@ void SND_quit(void);
 void SND_resetAudio(double sample_rate, double frame_rate);
 void SND_pauseAudio(bool paused);
 void SND_setQuality(int quality);
+void SND_setBufferSize(int samples);
+void SND_setBufferMultiplier(int multiplier);
 
 // watch audio device changes
 typedef enum {

workspace/all/minarch/minarch.c

@@ -2027,6 +2027,14 @@ static char* resample_labels[] = {
 	"Max",
 	NULL
 };
+static char* buffer_size_labels[] = {
+	"128", "256", "512", "1024",
+	NULL
+};
+static char* buffer_multiplier_labels[] = {
+	"4", "6", "8", "10",
+	NULL
+};
 static char* rewind_enable_labels[] = {
 	"Off",
 	"On",
@@ -2319,6 +2327,8 @@ enum {
 	FE_OPT_REWIND_COMPRESSION,
 	FE_OPT_REWIND_COMPRESSION_ACCEL,
 	FE_OPT_REWIND_AUDIO,
+	FE_OPT_AUDIO_BUFFER,
+	FE_OPT_AUDIO_LATENCY,
 	FE_OPT_COUNT,
 };
 
@@ -2737,6 +2747,26 @@ static struct Config {
 				.values = onoff_labels,
 				.labels = onoff_labels,
 			},
+			[FE_OPT_AUDIO_BUFFER] = {
+				.key	= "minarch__audio_buffer",
+				.name	= "Audio Buffer Size",
+				.desc	= "Size of audio chunks sent to the device.\nSmaller values lower latency but use more CPU.",
+				.default_value = 2,
+				.value = 2,
+				.count = 4,
+				.values = buffer_size_labels,
+				.labels = buffer_size_labels,
+			},
+			[FE_OPT_AUDIO_LATENCY] = {
+				.key	= "minarch__audio_latency",
+				.name	= "Audio Buffer Headroom",
+				.desc	= "How much audio is buffered ahead of time.\nSmaller values lower latency but may crackle.",
+				.default_value = 2,
+				.value = 2,
+				.count = 4,
+				.values = buffer_multiplier_labels,
+				.labels = buffer_multiplier_labels,
+			},
 			[FE_OPT_COUNT] = {NULL}
 		}
 	},
@@ -3109,6 +3139,16 @@ static void Config_syncFrontend(char* key, int value) {
 	else if (exactMatch(key,config.frontend.options[FE_OPT_REWIND_COMPRESSION_ACCEL].key)) {
 		i = FE_OPT_REWIND_COMPRESSION_ACCEL;
 	}
+	else if (exactMatch(key, config.frontend.options[FE_OPT_AUDIO_BUFFER].key)) {
+		int buf_size = atoi(config.frontend.options[FE_OPT_AUDIO_BUFFER].values[value]);
+		SND_setBufferSize(buf_size);
+		i = FE_OPT_AUDIO_BUFFER;
+	}
+	else if (exactMatch(key, config.frontend.options[FE_OPT_AUDIO_LATENCY].key)) {
+		int multiplier = atoi(config.frontend.options[FE_OPT_AUDIO_LATENCY].values[value]);
+		SND_setBufferMultiplier(multiplier);
+		i = FE_OPT_AUDIO_LATENCY;
+	}
 	if (i==-1) return;
 	Option* option = &config.frontend.options[i];
 	option->value = value;