void end_audio_frame() {
if (frame_offset == 0)
// No audio added; blip_end_frame() dislikes being called with an
// offset of 0
return;
assert(!(is_backwards_frame && frame_offset != get_frame_len()));
// Bring the signal level at the end of the frame to zero as outlined in
// set_audio_signal_level()
set_audio_signal_level(0);
blip_end_frame(blip, frame_offset);
if (playback_started) {
// Fudge playback rate by an amount proportional to the difference
// between the desired and current buffer fill levels to try to steer
// towards it
double const fudge_factor = 1.0 + 2*max_adjust*(0.5 - fill_level());
blip_set_rates(blip, cpu_clock_rate, sample_rate*fudge_factor);
}
else {
if (fill_level() >= 0.5) {
start_audio_playback();
playback_started = true;
}
}
int const n_samples = blip_read_samples(blip, blip_samples, ARRAY_LEN(blip_samples), 0);
// We expect to read all samples from blip_buf. If something goes wrong and
// we don't, clear the buffer to prevent data piling up in blip_buf's
// buffer (which lacks bounds checking).
int const avail = blip_samples_avail(blip);
if (avail != 0) {
printf("Warning: didn't read all samples from blip_buf (%d samples remain) - dropping samples\n",
avail);
blip_clear(blip);
}
#ifdef RECORD_MOVIE
add_movie_audio_frame(blip_samples, n_samples);
#endif
// Save the samples to the audio ring buffer
lock_audio();
write_samples(blip_samples, n_samples);
unlock_audio();
}
ECL_EXPORT void FrameAdvance(MyFrameInfo &f)
{
if (sgb)
{
sgb_set_controller_data((uint8_t *)&f.Keys);
}
else
{
GB_set_key_state(&GB, f.Keys & 0xff);
}
sound_start_clock = GB_epoch(&GB);
CurrentFramebuffer = f.VideoBuffer;
GB_set_lagged(&GB, true);
GB.frontend_rtc_time = f.Time;
uint32_t target = 35112 - FrameOverflow;
f.Cycles = GB_run_cycles(&GB, target);
FrameOverflow = f.Cycles - target;
if (sgb)
{
f.Width = 256;
f.Height = 224;
sgb_render_audio(GB_epoch(&GB), SgbSampleCallback);
}
else
{
f.Width = 160;
f.Height = 144;
}
blip_end_frame(leftblip, f.Cycles);
blip_end_frame(rightblip, f.Cycles);
f.Samples = blip_read_samples(leftblip, f.SoundBuffer, 2048, 1);
blip_read_samples(rightblip, f.SoundBuffer + 1, 2048, 1);
CurrentFramebuffer = NULL;
f.Lagged = GB_get_lagged(&GB);
}
int sound_update(unsigned int cycles)
{
int delta, preamp, time, l, r, *ptr;
/* Run PSG & FM chips until end of frame */
SN76489_Update(cycles);
fm_update(cycles);
/* FM output pre-amplification */
preamp = config.fm_preamp;
/* FM frame initial timestamp */
time = fm_cycles_start;
/* Restore last FM outputs from previous frame */
l = fm_last[0];
r = fm_last[1];
/* FM buffer start pointer */
ptr = fm_buffer;
/* flush FM samples */
if (config.hq_fm)
{
/* high-quality Band-Limited synthesis */
do
{
/* left channel */
delta = ((*ptr++ * preamp) / 100) - l;
l += delta;
blip_add_delta(snd.blips[0][0], time, delta);
/* right channel */
delta = ((*ptr++ * preamp) / 100) - r;
r += delta;
blip_add_delta(snd.blips[0][1], time, delta);
/* increment time counter */
time += fm_cycles_ratio;
}
while (time < cycles);
}
else
{
/* faster Linear Interpolation */
do
{
/* left channel */
delta = ((*ptr++ * preamp) / 100) - l;
l += delta;
blip_add_delta_fast(snd.blips[0][0], time, delta);
/* right channel */
delta = ((*ptr++ * preamp) / 100) - r;
r += delta;
blip_add_delta_fast(snd.blips[0][1], time, delta);
/* increment time counter */
time += fm_cycles_ratio;
}
while (time < cycles);
}
/* reset FM buffer pointer */
fm_ptr = fm_buffer;
/* save last FM output for next frame */
fm_last[0] = l;
fm_last[1] = r;
/* adjust FM cycle counters for next frame */
fm_cycles_count = fm_cycles_start = time - cycles;
/* end of blip buffers time frame */
blip_end_frame(snd.blips[0][0], cycles);
blip_end_frame(snd.blips[0][1], cycles);
/* return number of available samples */
return blip_samples_avail(snd.blips[0][0]);
}