Split generic part of audio code into a separate file so it can be used in other targets besides SDL

1 files changed, 375 insertions, 0 deletions

diff --git a/render_audio.c b/render_audio.c
new file mode 100644
index 0000000..39a9ccd
--- /dev/null
+++ b/render_audio.c
@@ -0,0 +1,375 @@
+#include <limits.h>
+#include <string.h>
+#include <stdlib.h>
+#include <math.h>
+#include "render_audio.h"
+#include "util.h"
+#include "config.h"
+#include "blastem.h"
+
+static uint8_t output_channels;
+static uint32_t buffer_samples, sample_rate;
+static uint32_t missing_count;
+
+static audio_source *audio_sources[8];
+static audio_source *inactive_audio_sources[8];
+static uint8_t num_audio_sources;
+static uint8_t num_inactive_audio_sources;
+
+static float overall_gain_mult, *mix_buf;
+static int sample_size;
+
+typedef void (*conv_func)(float *samples, void *vstream, int sample_count);
+
+static void convert_null(float *samples, void *vstream, int sample_count)
+{
+	memset(vstream, 0, sample_count * sample_size);
+}
+
+static void convert_s16(float *samples, void *vstream, int sample_count)
+{
+	int16_t *stream = vstream;
+	for (int16_t *end = stream + sample_count; stream < end; stream++, samples++)
+	{
+		float sample = *samples;
+		int16_t out_sample;
+		if (sample >= 1.0f) {
+			out_sample = 0x7FFF;
+		} else if (sample <= -1.0f) {
+			out_sample = -0x8000;
+		} else {
+			out_sample = sample * 0x7FFF;
+		}
+		*stream = out_sample;
+	}
+}
+
+static void clamp_f32(float *samples, void *vstream, int sample_count)
+{
+	for (; sample_count > 0; sample_count--, samples++)
+	{
+		float sample = *samples;
+		if (sample > 1.0f) {
+			sample = 1.0f;
+		} else if (sample < -1.0f) {
+			sample = -1.0f;
+		}
+		*samples = sample;
+	}
+}
+
+static int32_t mix_f32(audio_source *audio, float *stream, int samples)
+{
+	float *end = stream + samples;
+	int16_t *src = audio->front;
+	uint32_t i = audio->read_start;
+	uint32_t i_end = audio->read_end;
+	float *cur = stream;
+	float gain_mult = audio->gain_mult * overall_gain_mult;
+	size_t first_add = output_channels > 1 ? 1 : 0, second_add = output_channels > 1 ? output_channels - 1 : 1;
+	if (audio->num_channels == 1) {
+		while (cur < end && i != i_end)
+		{
+			*cur += gain_mult * ((float)src[i]) / 0x7FFF;
+			cur += first_add;
+			*cur += gain_mult * ((float)src[i++]) / 0x7FFF;
+			cur += second_add;
+			i &= audio->mask;
+		}
+	} else {
+		while(cur < end && i != i_end)
+		{
+			*cur += gain_mult * ((float)src[i++]) / 0x7FFF;
+			cur += first_add;
+			*cur += gain_mult * ((float)src[i++]) / 0x7FFF;
+			cur += second_add;
+			i &= audio->mask;
+		}
+	}
+	if (!render_is_audio_sync()) {
+		audio->read_start = i;
+	}
+	if (cur != end) {
+		debug_message("Underflow of %d samples, read_start: %d, read_end: %d, mask: %X\n", (int)(end-cur)/2, audio->read_start, audio->read_end, audio->mask);
+		return (cur-end)/2;
+	} else {
+		return ((i_end - i) & audio->mask) / audio->num_channels;
+	}
+}
+
+static conv_func convert;
+
+
+int mix_and_convert(unsigned char *byte_stream, int len, int *min_remaining_out)
+{
+	int samples = len / sample_size;
+	float *mix_dest = mix_buf ? mix_buf : (float *)byte_stream;
+	memset(mix_dest, 0, samples * sizeof(float));
+	int min_buffered = INT_MAX;
+	int min_remaining_buffer = INT_MAX;
+	for (uint8_t i = 0; i < num_audio_sources; i++)
+	{
+		int buffered = mix_f32(audio_sources[i], mix_dest, samples);
+		int remaining = (audio_sources[i]->mask + 1) / audio_sources[i]->num_channels - buffered;
+		min_buffered = buffered < min_buffered ? buffered : min_buffered;
+		min_remaining_buffer = remaining < min_remaining_buffer ? remaining : min_remaining_buffer;
+		audio_sources[i]->front_populated = 0;
+		render_buffer_consumed(audio_sources[i]);
+	}
+	convert(mix_dest, byte_stream, samples);
+	if (min_remaining_out) {
+		*min_remaining_out = min_remaining_buffer;
+	}
+	return min_buffered;
+}
+
+uint8_t all_sources_ready(void)
+{
+	uint8_t num_populated = 0;
+	num_populated = 0;
+	for (uint8_t i = 0; i < num_audio_sources; i++)
+	{
+		if (audio_sources[i]->front_populated) {
+			num_populated++;
+		}
+	}
+	return num_populated == num_audio_sources;
+}
+
+#define BUFFER_INC_RES 0x40000000UL
+
+void render_audio_adjust_clock(audio_source *src, uint64_t master_clock, uint64_t sample_divider)
+{
+	src->buffer_inc = ((BUFFER_INC_RES * (uint64_t)sample_rate) / master_clock) * sample_divider;
+}
+
+void render_audio_adjust_speed(float adjust_ratio)
+{
+	for (uint8_t i = 0; i < num_audio_sources; i++)
+	{
+		audio_sources[i]->buffer_inc = ((double)audio_sources[i]->buffer_inc) + ((double)audio_sources[i]->buffer_inc) * adjust_ratio + 0.5;
+	}
+}
+
+audio_source *render_audio_source(uint64_t master_clock, uint64_t sample_divider, uint8_t channels)
+{
+	audio_source *ret = NULL;
+	uint32_t alloc_size = render_is_audio_sync() ? channels * buffer_samples : nearest_pow2(render_min_buffered() * 4 * channels);
+	render_lock_audio();
+		if (num_audio_sources < 8) {
+			ret = calloc(1, sizeof(audio_source));
+			ret->back = malloc(alloc_size * sizeof(int16_t));
+			ret->front = render_is_audio_sync() ? malloc(alloc_size * sizeof(int16_t)) : ret->back;
+			ret->front_populated = 0;
+			ret->opaque = render_new_audio_opaque();
+			ret->num_channels = channels;
+			audio_sources[num_audio_sources++] = ret;
+		}
+	render_unlock_audio();
+	if (!ret) {
+		fatal_error("Too many audio sources!");
+	} else {
+		render_audio_adjust_clock(ret, master_clock, sample_divider);
+		double lowpass_cutoff = get_lowpass_cutoff(config);
+		double rc = (1.0 / lowpass_cutoff) / (2.0 * M_PI);
+		ret->dt = 1.0 / ((double)master_clock / (double)(sample_divider));
+		double alpha = ret->dt / (ret->dt + rc);
+		ret->lowpass_alpha = (int32_t)(((double)0x10000) * alpha);
+		ret->buffer_pos = 0;
+		ret->buffer_fraction = 0;
+		ret->last_left = ret->last_right = 0;
+		ret->read_start = 0;
+		ret->read_end = render_is_audio_sync() ? buffer_samples * channels : 0;
+		ret->mask = render_is_audio_sync() ? 0xFFFFFFFF : alloc_size-1;
+		ret->gain_mult = 1.0f;
+	}
+	render_audio_created(ret);
+	
+	return ret;
+}
+
+
+static float db_to_mult(float gain)
+{
+	return powf(10.0f, gain/20.0f);
+}
+
+void render_audio_source_gaindb(audio_source *src, float gain)
+{
+	src->gain_mult = db_to_mult(gain);
+}
+
+void render_pause_source(audio_source *src)
+{
+	uint8_t found = 0, remaining_sources;
+	render_lock_audio();
+		for (uint8_t i = 0; i < num_audio_sources; i++)
+		{
+			if (audio_sources[i] == src) {
+				audio_sources[i] = audio_sources[--num_audio_sources];
+				found = 1;
+				remaining_sources = num_audio_sources;
+				break;
+			}
+		}
+		
+	render_unlock_audio();
+	if (found) {
+		render_source_paused(src, remaining_sources);
+	}
+	inactive_audio_sources[num_inactive_audio_sources++] = src;
+}
+
+void render_resume_source(audio_source *src)
+{
+	render_lock_audio();
+		if (num_audio_sources < 8) {
+			audio_sources[num_audio_sources++] = src;
+		}
+	render_unlock_audio();
+	for (uint8_t i = 0; i < num_inactive_audio_sources; i++)
+	{
+		if (inactive_audio_sources[i] == src) {
+			inactive_audio_sources[i] = inactive_audio_sources[--num_inactive_audio_sources];
+		}
+	}
+	render_source_resumed(src);
+}
+
+void render_free_source(audio_source *src)
+{
+	render_pause_source(src);
+	
+	free(src->front);
+	if (render_is_audio_sync()) {
+		free(src->back);
+		render_free_audio_opaque(src->opaque);
+	}
+	free(src);
+	num_inactive_audio_sources--;
+}
+
+static int16_t lowpass_sample(audio_source *src, int16_t last, int16_t current)
+{
+	int32_t tmp = current * src->lowpass_alpha + last * (0x10000 - src->lowpass_alpha);
+	current = tmp >> 16;
+	return current;
+}
+
+static void interp_sample(audio_source *src, int16_t last, int16_t current)
+{
+	int64_t tmp = last * ((src->buffer_fraction << 16) / src->buffer_inc);
+	tmp += current * (0x10000 - ((src->buffer_fraction << 16) / src->buffer_inc));
+	src->back[src->buffer_pos++] = tmp >> 16;
+}
+
+static uint32_t sync_samples;
+void render_put_mono_sample(audio_source *src, int16_t value)
+{
+	value = lowpass_sample(src, src->last_left, value);
+	src->buffer_fraction += src->buffer_inc;
+	uint32_t base = render_is_audio_sync() ? 0 : src->read_end;
+	while (src->buffer_fraction > BUFFER_INC_RES)
+	{
+		src->buffer_fraction -= BUFFER_INC_RES;
+		interp_sample(src, src->last_left, value);
+		
+		if (((src->buffer_pos - base) & src->mask) >= sync_samples) {
+			render_do_audio_ready(src);
+		}
+		src->buffer_pos &= src->mask;
+	}
+	src->last_left = value;
+}
+
+void render_put_stereo_sample(audio_source *src, int16_t left, int16_t right)
+{
+	left = lowpass_sample(src, src->last_left, left);
+	right = lowpass_sample(src, src->last_right, right);
+	src->buffer_fraction += src->buffer_inc;
+	uint32_t base = render_is_audio_sync() ? 0 : src->read_end;
+	while (src->buffer_fraction > BUFFER_INC_RES)
+	{
+		src->buffer_fraction -= BUFFER_INC_RES;
+		
+		interp_sample(src, src->last_left, left);
+		interp_sample(src, src->last_right, right);
+		
+		if (((src->buffer_pos - base) & src->mask)/2 >= sync_samples) {
+			render_do_audio_ready(src);
+		}
+		src->buffer_pos &= src->mask;
+	}
+	src->last_left = left;
+	src->last_right = right;
+}
+
+static void update_source(audio_source *src, double rc, uint8_t sync_changed)
+{
+	double alpha = src->dt / (src->dt + rc);
+	int32_t lowpass_alpha = (int32_t)(((double)0x10000) * alpha);
+	src->lowpass_alpha = lowpass_alpha;
+	if (sync_changed) {
+		uint32_t alloc_size = render_is_audio_sync() ? src->num_channels * buffer_samples : nearest_pow2(render_min_buffered() * 4 * src->num_channels);
+		src->back = realloc(src->back, alloc_size * sizeof(int16_t));
+		if (render_is_audio_sync()) {
+			src->front = malloc(alloc_size * sizeof(int16_t));
+		} else {
+			free(src->front);
+			src->front = src->back;
+		}
+		src->mask = render_is_audio_sync() ? 0xFFFFFFFF : alloc_size-1;
+		src->read_start = 0;
+		src->read_end = render_is_audio_sync() ? buffer_samples * src->num_channels : 0;
+		src->buffer_pos = 0;
+	}
+}
+
+uint8_t old_audio_sync;
+void render_audio_initialized(render_audio_format format, uint32_t rate, uint8_t channels, uint32_t buffer_size, int sample_size_in)
+{
+	sample_rate = rate;
+	output_channels = channels;
+	buffer_samples = buffer_size;
+	sample_size = sample_size_in;
+	if (mix_buf) {
+		free(mix_buf);
+		mix_buf = NULL;
+	}
+	switch(format)
+	{
+	case RENDER_AUDIO_S16:
+		convert = convert_s16;
+		mix_buf = calloc(output_channels * buffer_samples, sizeof(float));
+		break;
+	case RENDER_AUDIO_FLOAT:
+		convert = clamp_f32;
+		break;
+	case RENDER_AUDIO_UNKNOWN:
+		convert = convert_null;
+		mix_buf = calloc(output_channels * buffer_samples, sizeof(float));
+		break;
+	}
+	uint32_t syncs = render_audio_syncs_per_sec();
+	if (syncs) {
+		sync_samples = rate / syncs;
+	} else {
+		sync_samples = buffer_samples;
+	}
+	char * gain_str = tern_find_path(config, "audio\0gain\0", TVAL_PTR).ptrval;
+	overall_gain_mult = db_to_mult(gain_str ? atof(gain_str) : 0.0f);
+	uint8_t sync_changed = old_audio_sync != render_is_audio_sync();
+	old_audio_sync = render_is_audio_sync();
+	double lowpass_cutoff = get_lowpass_cutoff(config);
+	double rc = (1.0 / lowpass_cutoff) / (2.0 * M_PI);
+	render_lock_audio();
+		for (uint8_t i = 0; i < num_audio_sources; i++)
+		{
+			update_source(audio_sources[i], rc, sync_changed);
+		}
+	render_unlock_audio();
+	for (uint8_t i = 0; i < num_inactive_audio_sources; i++)
+	{
+		update_source(inactive_audio_sources[i], rc, sync_changed);
+	}
+}
\ No newline at end of file