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ps2, kludge for handling audsrv shortcomings

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This commit is contained in:
kub 2024-02-15 23:08:24 +01:00
parent dedf7fa7d5
commit fd604aa720
1 changed files with 74 additions and 29 deletions
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103
platform/ps2/emu.c
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@ -79,9 +79,10 @@ static uint8_t vsync; /* 0 (Disabled), 1 (Enabled), 2 (Dynamic) */
/* sound stuff */
#define SOUND_BLOCK_COUNT 6
#define SOUND_BUFFER_SIZE (2*54000/50*SOUND_BLOCK_COUNT) // max.rate/min.frames
#define SOUND_BUFFER_CHUNK (2*54000/50) // max.rate/min.frames, stereo
static short __attribute__((aligned(4))) sndBuffer[SOUND_BUFFER_SIZE];
static short __attribute__((aligned(4))) sndBuffer[SOUND_BUFFER_CHUNK*SOUND_BLOCK_COUNT];
static short __attribute__((aligned(4))) nulBuffer[SOUND_BUFFER_CHUNK];
static short *snd_playptr = NULL, *sndBuffer_endptr = NULL;
static int samples_made = 0, samples_done = 0, samples_block = 0;
static int sound_thread_exit = 0;
@ -91,6 +92,23 @@ extern void *_gp;
static int mp3_init(void) { return 0; }
/* audsrv in ps2sdk has shortcomings:
* - it has a bug which prevents it from discerning "ringbuffer empty" from
* "ringbuffer full", which leads to audio not stopped if all queued samples
* have been played. Hence, it repeats the complete ringbuffer over and again.
* - on audsrv_set_format the ringbuffer is preset to be about 40% filled,
* regardless of the data in the buffer at that moment. Old data is played out
* if audio play is started.
* - stopping audio play is keeping any remaining samples in the buffer, which
* are played first after the next audio play. There's no method to clear the
* ringbuffer.
*
* To cope with this, audio samples are always pushed to audsrv to prevent the
* ringbuffer from emptying, even in the menu. This also avoids stopping audio.
* Since silence is played in the menu, the behaviour of set_format when leaving
* the menu is covered since the buffer is filled with silence at that time.
*/
static void writeSound(int len)
{
int ret, l;
@ -106,45 +124,72 @@ static void writeSound(int len)
}
if (sndBuffer_endptr < PicoIn.sndOut)
sndBuffer_endptr = PicoIn.sndOut;
samples_made += len / 2;
// signal the snd thread
samples_made += len / 2;
// lprintf("signal, %i/%i\n", samples_done, samples_made);
ret = SignalSema(sound_sem);
if (ret < 0) lprintf("snd signal ret %08x\n", ret);
// ret = SignalSema(sound_sem);
// if (ret < 0) lprintf("snd signal ret %08x\n", ret);
}
static int sound_thread(void *argp)
{
lprintf("sthr: start\n");
while (!sound_thread_exit)
{
int ret = 0;
if (samples_made - samples_done < samples_block) {
// wait for data (use at least 2 blocks)
// lprintf("sthr: wait... (%i)\n", samples_made - samples_done);
while (samples_made - samples_done < samples_block*2 && !sound_thread_exit)
ret = WaitSema(sound_sem);
if (ret < 0) lprintf("sthr: WaitSema: %i\n", ret);
continue;
// curb the sample queue to prevent it from filling
while (samples_made - samples_done > 4*samples_block) {
short *sndOut = PicoIn.sndOut, *sndEnd = sndBuffer_endptr;
int buflen = sndEnd - snd_playptr;
if (sndOut > snd_playptr)
buflen = sndOut - snd_playptr;
if (buflen > samples_made - samples_done - 4*samples_block)
buflen = samples_made - samples_done - 4*samples_block;
samples_done += buflen;
snd_playptr += buflen;
if (snd_playptr >= sndBuffer_endptr)
snd_playptr -= sndBuffer_endptr - sndBuffer;
}
// lprintf("sthr: got data: %i\n", samples_made - samples_done);
short *sndOut = PicoIn.sndOut, *sndEnd = sndBuffer_endptr;
int buflen = sndEnd - snd_playptr;
if (sndOut >= snd_playptr)
buflen = sndOut - snd_playptr;
if (buflen > samples_block)
buflen = samples_block;
ret = audsrv_play_audio((char *)snd_playptr, buflen*2);
if (ret != buflen*2 && ret >= 0) lprintf("sthr: play ret: %i, buflen: %i\n", ret, buflen*2);
if (ret < 0) lprintf("sthr: play: ret %08x; pos %i/%i\n", ret, samples_done, samples_made);
samples_done += buflen;
snd_playptr += buflen;
// queue samples to audsrv, minimum 2 frames
// if there aren't enough samlpes, queue silence
int queued = audsrv_queued()/2;
while (queued < 2*samples_block) {
short *sndOut = PicoIn.sndOut, *sndEnd = sndBuffer_endptr;
if (snd_playptr >= sndBuffer_endptr)
snd_playptr = sndBuffer;
// compute sample chunk size
int buflen = sndEnd - snd_playptr;
if (sndOut > snd_playptr)
buflen = sndOut - snd_playptr;
if (buflen > samples_made - samples_done)
buflen = samples_made - samples_done;
if (buflen > 4*samples_block - queued)
buflen = 4*samples_block - queued;
// play audio
if (buflen > 0) {
ret = audsrv_play_audio((char *)snd_playptr, buflen*2);
samples_done += buflen;
snd_playptr += buflen;
if (snd_playptr >= sndBuffer_endptr)
snd_playptr -= sndBuffer_endptr - sndBuffer;
} else {
buflen = 3*samples_block - queued;
ret = audsrv_play_audio((char *)nulBuffer, buflen*2);
}
if (ret != buflen*2 && ret >= 0) lprintf("sthr: play ret: %i, buflen: %i\n", ret, buflen*2);
if (ret < 0) lprintf("sthr: play: ret %08x; pos %i/%i\n", ret, samples_done, samples_made);
queued = audsrv_queued()/2;
}
ret = WaitSema(sound_sem);
if (ret < 0) lprintf("sthr: WaitSema failed (%d)\n", ret);
}
lprintf("sthr: exit\n");
@ -235,14 +280,13 @@ void pemu_sound_start(void) {
PicoOpt_old = PicoIn.opt;
pal_old = Pico.m.pal;
}
ret = audsrv_play_audio((char *)snd_playptr, 4);
audsrv_play_audio((char *)snd_playptr, 2*2);
}
void pemu_sound_stop(void)
{
samples_made = samples_done = 0;
plat_sleep_ms(200);
audsrv_stop_audio();
}
static void sound_deinit(void)
@ -259,6 +303,7 @@ static void sound_deinit(void)
static int vsync_handler(void)
{
iSignalSema(vsync_sema_id);
iSignalSema(sound_sem);
ExitHandler();
return 0;