diff options
| -rw-r--r-- | genesis.c | 3 | ||||
| -rwxr-xr-x | render_sdl.c | 35 |
2 files changed, 24 insertions, 14 deletions
@@ -1072,7 +1072,6 @@ static void start_genesis(system_header *system, char *statefile) { genesis_context *gen = (genesis_context *)system; set_keybindings(&gen->io); - render_set_video_standard((gen->version_reg & HZ50) ? VID_PAL : VID_NTSC); if (statefile) { //first try loading as a native format savestate deserialize_buffer state; @@ -1237,6 +1236,8 @@ genesis_context *alloc_init_genesis(rom_info *rom, void *main_rom, void *lock_on gen->int_latency_prev1 = MCLKS_PER_68K * 32; gen->int_latency_prev2 = MCLKS_PER_68K * 16; + render_set_video_standard((gen->version_reg & HZ50) ? VID_PAL : VID_NTSC); + gen->ym = malloc(sizeof(ym2612_context)); ym_init(gen->ym, gen->master_clock, MCLKS_PER_YM, system_opts); diff --git a/render_sdl.c b/render_sdl.c index c8bb814..1b94ff1 100755 --- a/render_sdl.c +++ b/render_sdl.c @@ -182,41 +182,48 @@ static int32_t last_buffered = NO_LAST_BUFFERED; static uint8_t need_adjust, need_pause; static float adjust_ratio, average_change; #define BUFFER_FRAMES_THRESHOLD 6 -#define MAX_ADJUST 0.000625 +#define BASE_MAX_ADJUST 0.05 +static float max_adjust; static void audio_callback_drc(void *userData, uint8_t *byte_stream, int len) { //TODO: update progress tracking so we can adjust resample rate memset(byte_stream, 0, len); - int32_t min_buffered = 0x7FFFFFFF; + int32_t cur_min_buffered = 0x7FFFFFFF; uint32_t min_remaining_buffer = 0xFFFFFFFF; for (uint8_t i = 0; i < num_audio_sources; i++) { + printf("buffered at start: %d, ", ((audio_sources[i]->read_end - audio_sources[i]->read_start) & audio_sources[i]->mask) / audio_sources[i]->num_channels); int32_t buffered = mix(audio_sources[i], byte_stream, len); - min_buffered = buffered < min_buffered ? buffered : min_buffered; + printf("buffered after mix: %d\n", buffered); + cur_min_buffered = buffered < cur_min_buffered ? buffered : cur_min_buffered; uint32_t remaining = (audio_sources[i]->mask + 1)/audio_sources[i]->num_channels - buffered; min_remaining_buffer = remaining < min_remaining_buffer ? remaining : min_remaining_buffer; } if (last_buffered > NO_LAST_BUFFERED) { average_change *= 0.8f; - average_change += ((int32_t)min_buffered - last_buffered) * 0.2f; + average_change += ((int32_t)cur_min_buffered - last_buffered) * 0.2f; } - last_buffered = min_buffered; + last_buffered = cur_min_buffered; float frames_to_problem; if (average_change < 0) { - frames_to_problem = (float)min_buffered / -average_change; + frames_to_problem = (float)cur_min_buffered / -average_change; } else { frames_to_problem = (float)min_remaining_buffer / average_change; } - if (frames_to_problem < BUFFER_FRAMES_THRESHOLD || min_buffered < 0) { + if ( + frames_to_problem < BUFFER_FRAMES_THRESHOLD || cur_min_buffered < 0 + || (cur_min_buffered < min_buffered/2 && average_change < 0) + || (cur_min_buffered > (2*min_buffered) && average_change > 0) + ) { need_adjust = num_audio_sources; - if (min_buffered < 0) { - adjust_ratio = MAX_ADJUST; + if (cur_min_buffered < 0) { + adjust_ratio = max_adjust; need_pause = 1; } else { adjust_ratio = -1.5 * average_change / buffer_samples; - if (fabsf(adjust_ratio) > MAX_ADJUST) { - adjust_ratio = adjust_ratio > 0 ? MAX_ADJUST : -MAX_ADJUST; + if (fabsf(adjust_ratio) > max_adjust) { + adjust_ratio = adjust_ratio > 0 ? max_adjust : -max_adjust; } } printf("frames_to_problem: %f, avg_change: %f, adjust_ratio: %f\n", frames_to_problem, average_change, adjust_ratio); @@ -227,7 +234,7 @@ static void audio_callback_drc(void *userData, uint8_t *byte_stream, int len) audio_sources[i]->adjusted = 0; } } else { - printf("no adjust - frames_to_problem: %f, avg_change: %f, min_buffered: %d, min_remaining_buffer: %d\n", frames_to_problem, average_change, min_buffered, min_remaining_buffer); + printf("no adjust - frames_to_problem: %f, avg_change: %f, cur_min_buffered: %d, min_remaining_buffer: %d\n", frames_to_problem, average_change, cur_min_buffered, min_remaining_buffer); } } @@ -365,7 +372,7 @@ static void do_audio_ready(audio_source *src) uint8_t local_need_pause; SDL_LockAudio(); src->read_end = src->buffer_pos; - num_buffered = (src->read_end - src->read_start) & src->mask; + num_buffered = ((src->read_end - src->read_start) & src->mask) / src->num_channels; if (need_adjust && !src->adjusted) { src->adjusted = 1; need_adjust--; @@ -843,6 +850,7 @@ void render_init(int width, int height, char * title, uint8_t fullscreen) } buffer_samples = actual.samples; sample_rate = actual.freq; + max_adjust = BASE_MAX_ADJUST * (float)buffer_samples / (float)sample_rate; printf("Initialized audio at frequency %d with a %d sample buffer, ", actual.freq, actual.samples); if (actual.format == AUDIO_S16SYS) { puts("signed 16-bit int format"); @@ -911,6 +919,7 @@ void render_set_video_standard(vid_std std) float mult = max_repeat > 1 ? 2.5 : 1.5; min_buffered = (((float)max_repeat * mult * (float)sample_rate/(float)source_hz) / (float)buffer_samples) + 0.9999; min_buffered *= buffer_samples; + printf("Min samples buffered before audio start: %d\n", min_buffered); } void render_update_caption(char *title) |