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DSPHLE: Added an option for 48khz output and supporting upsampling code.

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Games that have higher frequency sounds and music should sound a bit better using 48k. 

I don't have any games that use DTKMusic so that upsampling code untested. If you get strange sounds only at 48k try toggling dtk music to see if that isolates the problem and let me know.

git-svn-id: https://dolphin-emu.googlecode.com/svn/trunk@6383 8ced0084-cf51-0410-be5f-012b33b47a6e
This commit is contained in:
mylek4
2010-11-12 03:39:07 +00:00
parent c6d95c9087
commit 67f850dd37
9 changed files with 149 additions and 109 deletions
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81
Source/Core/Core/Src/HW/AudioInterface.cpp
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@ -281,17 +281,11 @@ unsigned int Callback_GetStreaming(short* _pDestBuffer, unsigned int _numSamples
const int lvolume = g_AudioRegister.m_Volume.leftVolume;
const int rvolume = g_AudioRegister.m_Volume.rightVolume;
if (g_AISampleRate == 48000 && _sampleRate == 32000)
{
_dbg_assert_msg_(AUDIO_INTERFACE, !(_numSamples & 1), "Number of Samples: %i must be even!", _numSamples);
_numSamples = _numSamples * 3 / 2;
}
else if (g_AISampleRate == 32000 && _sampleRate == 48000)
{
// AyuanX: Up-sampling is not implemented yet
PanicAlert("AUDIO_INTERFACE: Up-sampling is not implemented yet!");
}
int pcm_l = 0, pcm_r = 0;
for (unsigned int i = 0; i < _numSamples; i++)
@ -299,45 +293,82 @@ unsigned int Callback_GetStreaming(short* _pDestBuffer, unsigned int _numSamples
if (pos == 0)
ReadStreamBlock(pcm);
if (g_AISampleRate == 48000 && _sampleRate == 32000)
if (g_AISampleRate == 48000 && _sampleRate == 32000) //downsample 48>32
{
if (i % 3)
{
pcm_l = (((pcm_l + (int)pcm[pos*2]) / 2 * lvolume) >> 8) + (int)(*_pDestBuffer);
if (pcm_l > 32767)
pcm_l = 32767;
else if (pcm_l < -32767)
pcm_l = -32767;
if (pcm_l > 32767) pcm_l = 32767;
else if (pcm_l < -32767) pcm_l = -32767;
*_pDestBuffer++ = pcm_l;
pcm_r = (((pcm_r + (int)pcm[pos*2+1]) / 2 * rvolume) >> 8) + (int)(*_pDestBuffer);
if (pcm_r > 32767)
pcm_r = 32767;
else if (pcm_r < -32767)
pcm_r = -32767;
if (pcm_r > 32767) pcm_r = 32767;
else if (pcm_r < -32767) pcm_r = -32767;
*_pDestBuffer++ = pcm_r;
}
pcm_l = pcm[pos*2];
pcm_r = pcm[pos*2+1];
pos++;
}
else
else if (g_AISampleRate == 32000 && _sampleRate == 48000) //upsample 32>48
{
//starts with one sample of 0
const u32 ratio = (u32)( 65536.0f * 32000.0f / (float)_sampleRate );
static u32 frac = 0;
static s16 l1 = 0;
static s16 l2 = 0;
static s16 r1 = 0;
static s16 r2 = 0;
if ( frac >= 0x10000 || frac == 0)
{
frac &= 0xffff;
l1 = l2; //current
l2 = pcm[pos * 2]; //next
r1 = r2; //current
r2 = pcm[pos * 2 + 1]; //next
}
pcm_l = (l1 << 16) + (l2 - l1) * (u16)frac >> 16;
pcm_r = (l1 << 16) + (l2 - l1) * (u16)frac >> 16;
pcm_l = (pcm_l * lvolume >> 8) + (int)(*_pDestBuffer);
if (pcm_l > 32767) pcm_l = 32767;
else if (pcm_l < -32767) pcm_l = -32767;
*_pDestBuffer++ = pcm_l;
pcm_r = (pcm_r * lvolume >> 8) + (int)(*_pDestBuffer);
if (pcm_r > 32767) pcm_r = 32767;
else if (pcm_r < -32767) pcm_r = -32767;
*_pDestBuffer++ = pcm_r;
frac += ratio;
pos += frac >> 16;
}
else //1:1 no resampling
{
pcm_l = (((int)pcm[pos*2] * lvolume) >> 8) + (int)(*_pDestBuffer);
if (pcm_l > 32767)
pcm_l = 32767;
else if (pcm_l < -32767)
pcm_l = -32767;
if (pcm_l > 32767) pcm_l = 32767;
else if (pcm_l < -32767) pcm_l = -32767;
*_pDestBuffer++ = pcm_l;
pcm_r = (((int)pcm[pos*2+1] * rvolume) >> 8) + (int)(*_pDestBuffer);
if (pcm_r > 32767)
pcm_r = 32767;
else if (pcm_r < -32767)
pcm_r = -32767;
if (pcm_r > 32767) pcm_r = 32767;
else if (pcm_r < -32767) pcm_r = -32767;
*_pDestBuffer++ = pcm_r;
pos++;
}
if (++pos == 28)
if (pos == 28)
pos = 0;
}
}