Adds dynamic audio rate control option

When DRC is on, game syncs to frame rate instead of audio buffer
capacity. Audio is resampled to generate more samples when buffer is
low and less when buffer is high, to keep buffer 40%-60% full. Buffer
size doubled to keep same avg. audio latency value.

Audio can distort when buffer is out of range, not often during
gameplay. Better resampler could improve but would be slower. When
buffer is always out of range (heavy frameskip), it is better to leave
off, DRC doesn't help anyway then.

Idea from RetroArch audio_driver.c and
https://near.sh/articles/audio/dynamic-rate-control

1 files changed, 63 insertions, 8 deletions

diff --git a/plat_sdl.c b/plat_sdl.c
index 0beb924..817a47e 100644
--- a/plat_sdl.c
+++ b/plat_sdl.c
@@ -18,6 +18,8 @@ struct audio_state {
 	struct audio_frame *buf;
 	int in_sample_rate;
 	int out_sample_rate;
+	int sample_rate_adj;
+	int adj_out_sample_rate;
 };
 
 struct audio_state audio;
@@ -25,11 +27,16 @@ struct audio_state audio;
 static void plat_sound_select_resampler(void);
 void (*plat_sound_write)(const struct audio_frame *data, int frames);
 
+#define DRC_MAX_ADJUSTMENT 0.003
+#define DRC_ADJ_BELOW 0.4
+#define DRC_ADJ_ABOVE 0.6
+
 static char msg[HUD_LEN];
 static unsigned msg_priority = 0;
 static unsigned msg_expire = 0;
 
 static bool frame_dirty = false;
+static int frame_time = 1000000 / 60;
 
 static void video_expire_msg(void)
 {
@@ -65,16 +72,16 @@ static int audio_resample_nearest(struct audio_frame data) {
 	static int diff = 0;
 	int consumed = 0;
 
-	if (diff < audio.out_sample_rate) {
+	if (diff < audio.adj_out_sample_rate) {
 		audio.buf[audio.buf_w++] = data;
 		if (audio.buf_w >= audio.buf_len) audio.buf_w = 0;
 
 		diff += audio.in_sample_rate;
 	}
 
-	if (diff >= audio.out_sample_rate) {
+	if (diff >= audio.adj_out_sample_rate) {
 		consumed++;
-		diff -= audio.out_sample_rate;
+		diff -= audio.adj_out_sample_rate;
 	}
 
 	return consumed;
@@ -233,9 +240,25 @@ void plat_video_process(const void *data, unsigned width, unsigned height, size_
 
 void plat_video_flip(void)
 {
-	if (frame_dirty)
+	static unsigned int next_frame_time_us = 0;
+
+	if (frame_dirty) {
+		unsigned int time = plat_get_ticks_us();
+
+		if (limit_frames && enable_drc && time < next_frame_time_us) {
+			usleep(next_frame_time_us - time);
+		}
+
+		if (!next_frame_time_us)
+			next_frame_time_us = time;
+
 		fb_flip();
 
+		do {
+			next_frame_time_us += frame_time;
+		} while (next_frame_time_us < time);
+	}
+
 	frame_dirty = false;
 }
 
@@ -324,6 +347,9 @@ static int plat_sound_init(void)
 
 	audio.in_sample_rate = sample_rate;
 	audio.out_sample_rate = received.freq;
+	audio.sample_rate_adj = audio.out_sample_rate * DRC_MAX_ADJUSTMENT;
+	audio.adj_out_sample_rate = audio.out_sample_rate;
+
 	plat_sound_select_resampler();
 	plat_sound_resize_buffer();
 
@@ -347,12 +373,22 @@ float plat_sound_capacity(void)
 }
 
 #define BATCH_SIZE 100
-void plat_sound_write_resample(const struct audio_frame *data, int frames, int (*resample)(struct audio_frame data))
+void plat_sound_write_resample(const struct audio_frame *data, int frames, int (*resample)(struct audio_frame data), bool drc)
 {
 	int consumed = 0;
 	if (audio.buf_len == 0)
 		return;
 
+	if (drc) {
+		if (plat_sound_capacity() < DRC_ADJ_BELOW) {
+			audio.adj_out_sample_rate = audio.out_sample_rate - audio.sample_rate_adj;
+		} else if (plat_sound_capacity() > DRC_ADJ_ABOVE) {
+			audio.adj_out_sample_rate = audio.out_sample_rate + audio.sample_rate_adj;
+		} else {
+			audio.adj_out_sample_rate = audio.out_sample_rate;
+		}
+	}
+
 	SDL_LockAudio();
 
 	while (frames > 0) {
@@ -382,12 +418,17 @@ void plat_sound_write_resample(const struct audio_frame *data, int frames, int (
 
 void plat_sound_write_passthrough(const struct audio_frame *data, int frames)
 {
-	plat_sound_write_resample(data, frames, audio_resample_passthrough);
+	plat_sound_write_resample(data, frames, audio_resample_passthrough, false);
 }
 
 void plat_sound_write_nearest(const struct audio_frame *data, int frames)
 {
-	plat_sound_write_resample(data, frames, audio_resample_nearest);
+	plat_sound_write_resample(data, frames, audio_resample_nearest, false);
+}
+
+void plat_sound_write_drc(const struct audio_frame *data, int frames)
+{
+	plat_sound_write_resample(data, frames, audio_resample_nearest, true);
 }
 
 void plat_sound_resize_buffer(void) {
@@ -398,6 +439,10 @@ void plat_sound_resize_buffer(void) {
 		? current_audio_buffer_size * audio.in_sample_rate / frame_rate
 		: 0;
 
+		/* Dynamic adjustment keeps buffer 50% full, need double size */
+	if (enable_drc)
+		audio.buf_len *= 2;
+
 	if (audio.buf_len == 0) {
 		SDL_UnlockAudio();
 		return;
@@ -422,7 +467,10 @@ void plat_sound_resize_buffer(void) {
 
 static void plat_sound_select_resampler(void)
 {
-	if (audio.in_sample_rate == audio.out_sample_rate) {
+	if (enable_drc) {
+		PA_INFO("Using audio adjustment (in: %d, out: %d-%d)\n", audio.in_sample_rate, audio.out_sample_rate - audio.sample_rate_adj, audio.out_sample_rate + audio.sample_rate_adj);
+		plat_sound_write = plat_sound_write_drc;
+	} else if (audio.in_sample_rate == audio.out_sample_rate) {
 		PA_INFO("Using passthrough resampler (in: %d, out: %d)\n", audio.in_sample_rate, audio.out_sample_rate);
 		plat_sound_write = plat_sound_write_passthrough;
 	} else {
@@ -479,7 +527,14 @@ int plat_reinit(void)
 			PA_ERROR("SDL sound failed to init: %s\n", SDL_GetError());
 			return -1;
 		}
+	} else {
+		plat_sound_resize_buffer();
+		plat_sound_select_resampler();
 	}
+
+	if (frame_rate != 0)
+		frame_time = 1000000 / frame_rate;
+
 	scale_update_scaler();
 	return 0;
 }