signed int ReGBA_AudioUpdate()
{
u32 i, j;
s16* audio_buff;
s16 *dst_ptr, *dst_ptr1;
u32 n = ds2_checkAudiobuff();
int ret;
u8 WasInUnderrun = Stats.InSoundBufferUnderrun;
Stats.InSoundBufferUnderrun = n == 0 && ReGBA_GetAudioSamplesAvailable() < AUDIO_LEN * 2;
if (Stats.InSoundBufferUnderrun && !WasInUnderrun)
Stats.SoundBufferUnderrunCount++;
// On auto frameskip, sound buffers being full or empty determines
// whether we're late.
if (AUTO_SKIP)
{
if (n >= 2)
{
// We're in no hurry, because 2 buffers are still full.
// Minimum skip 1
if(SKIP_RATE > 1)
{
#if defined TRACE || defined TRACE_FRAMESKIP
ReGBA_Trace("I: Decreasing automatic frameskip: %u..%u", SKIP_RATE, SKIP_RATE - 1);
#endif
SKIP_RATE--;
}
}
else
{
// Maximum skip 9
if(SKIP_RATE < 8)
{
#if defined TRACE || defined TRACE_FRAMESKIP
ReGBA_Trace("I: Increasing automatic frameskip: %u..%u", SKIP_RATE, SKIP_RATE + 1);
#endif
SKIP_RATE++;
}
}
}
/* There must be AUDIO_LEN * 2 samples generated in order for the first
* AUDIO_LEN to be valid. Some sound is generated in the past from the
* future, and if the first AUDIO_LEN is grabbed before the core has had
* time to generate all of it (at AUDIO_LEN * 2), the end may still be
* silence, causing crackling. */
if (ReGBA_GetAudioSamplesAvailable() < AUDIO_LEN * 2)
{
// Generate more sound first, please!
return -1;
}
// We have enough sound. Complete this update.
if (game_fast_forward || temporary_fast_forward)
{
if (n >= AUDIO_BUFFER_COUNT)
{
// Drain the buffer down to a manageable size, then exit.
// This needs to be high to avoid audible crackling/bubbling,
// but not so high as to require all of the sound to be emitted.
// gpSP synchronises on the sound, after all. -Neb, 2013-03-23
ReGBA_DiscardAudioSamples(ReGBA_GetAudioSamplesAvailable() - AUDIO_LEN);
return 0;
}
}
else
{
// Wait for at least one buffer to be free for audio.
// Output assertion: The return value is between 0, inclusive,
// and AUDIO_BUFFER_COUNT, inclusive, but can also be
// 4294967295; that's (unsigned int) -1. (DSTWO SPECIFIC HACK)
unsigned int n2;
while ((n2 = ds2_checkAudiobuff()) >= AUDIO_BUFFER_COUNT && (int) n2 >= 0);
}
audio_buff = ds2_getAudiobuff();
if (audio_buff == NULL) {
#if defined TRACE || defined TRACE_SOUND
ReGBA_Trace("Recovered from the lack of a buffer");
#endif
return -1;
}
dst_ptr = audio_buff; // left (stereo)
dst_ptr1 = dst_ptr + (int) (AUDIO_LEN / OUTPUT_FREQUENCY_DIVISOR); // right (stereo)
for(i= 0; i < AUDIO_LEN; i += OUTPUT_FREQUENCY_DIVISOR)
{
s16 Left = 0, Right = 0, LeftPart, RightPart;
for (j = 0; j < OUTPUT_FREQUENCY_DIVISOR; j++) {
ReGBA_LoadNextAudioSample(&LeftPart, &RightPart);
if (LeftPart > 2047) LeftPart = 2047;
else if (LeftPart < -2048) LeftPart = -2048;
Left += LeftPart / OUTPUT_FREQUENCY_DIVISOR;
if (RightPart > 2047) RightPart = 2047;
else if (RightPart < -2048) RightPart = -2048;
Right += RightPart / OUTPUT_FREQUENCY_DIVISOR;
}
*dst_ptr++ = Left << 4;
*dst_ptr1++ = Right << 4;
}
if (game_fast_forward || temporary_fast_forward)
{
// Dampen the sound with the previous samples written
// (or unitialised data if we just started the emulator)
StartFastForwardedSound(audio_buff,
&audio_buff[(int) (AUDIO_LEN / OUTPUT_FREQUENCY_DIVISOR)]);
// Store the end for the next time
EndFastForwardedSound(&audio_buff[(int) (AUDIO_LEN / OUTPUT_FREQUENCY_DIVISOR) - DAMPEN_SAMPLE_COUNT],
&audio_buff[(int) (AUDIO_LEN / OUTPUT_FREQUENCY_DIVISOR) * 2 - DAMPEN_SAMPLE_COUNT]);
}
Stats.InSoundBufferUnderrun = 0;
ds2_updateAudio();
return 0;
}
int doPlay(FILE *fp, int samples_per_trans)
{
int len;
int m, n;
void *audiobuf;
unsigned int key;
int flag;
int vol;
unsigned short *audio_buffer_addr;
len = wave_BytePsample * samples_per_trans;
audiobuf = (void*)malloc(len);
if(NULL == audiobuf) return -1;
do {
audio_buffer_addr = (unsigned short*)ds2_getAudiobuff();
} while(NULL == audio_buffer_addr);
vol = 127;
flag = 1;
while(flag)
{
n = fread(audiobuf, 1, len, fp);
if(0 == n) break;
m = (n + len -1)/len;
m *= len;
if(n < m)
{
memset((char*)((unsigned int)audiobuf + n), 0, m -n);
}
//assume 16-bit and stereo audio
unsigned short *src;
unsigned short *dst0;
unsigned short *dst1;
src = (unsigned short*)audiobuf;
//the audio buffer's front part are left channel's data, behind ate right
// channel's data, different as the wave file, it's data not interleaved
dst0 = audio_buffer_addr;
dst1 = audio_buffer_addr + samples_per_trans;
n = 0;
while(n++ < samples_per_trans)
{
*dst0++ = *src++;
*dst1++ = *src++;
}
//ther are 4 audio buffers on the ds2sdk, we should not make it overflow
while(ds2_checkAudiobuff() > 3)
{
//128 samples (channles * Bytes Per channel) can play 2.9ms @ 44.1KHz
//mdelay((samples_per_trans/128)*2.9);
mdelay(1);
}
ds2_updateAudio();
while(ds2_checkAudiobuff() > 1)
{
key = getKey();
switch(key)
{
//incread volume
case KEY_UP:
vol += 1;
if(vol > 255) vol = 255;
ds2_setVolume(vol);
printf("volume: %d\n", vol);
break;
//decreade volume
case KEY_DOWN:
vol -= 1;
if(vol < 0) vol = 0;
ds2_setVolume(vol);
printf("volume: %d\n", vol);
break;
//exit
case KEY_B:
flag = 0;
break;
}
}
do {
audio_buffer_addr = (unsigned short*)ds2_getAudiobuff();
} while(NULL == audio_buffer_addr);
}
free(audiobuf);
return 0;
}
