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FS #7286. Do correct rounding of final 16 bit samples before sending to DAC, for you golden-eared people.

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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@14514 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Thom Johansen 2007-08-29 14:32:52 +00:00
parent 8f5f690760
commit c668de3085
3 changed files with 42 additions and 19 deletions
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13
apps/plugins/test_codec.c
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@ -221,7 +221,8 @@ static bool pcmbuf_insert_wav(const void *ch1, const void *ch2, int count)
const int32_t* data1_32;
const int32_t* data2_32;
unsigned char* p = wavbuffer;
int scale = wavinfo.sampledepth - 15;
const int scale = wavinfo.sampledepth - 15;
const int dc_bias = 1 << (scale - 1);
/* Prevent idle poweroff */
rb->reset_poweroff_timer();
@ -266,18 +267,18 @@ static bool pcmbuf_insert_wav(const void *ch1, const void *ch2, int count)
{
case STEREO_INTERLEAVED:
while (count--) {
int2le16(p, clip_sample((*data1_32++) >> scale));
int2le16(p, clip_sample((*data1_32++ + dc_bias) >> scale));
p += 2;
int2le16(p, clip_sample((*data1_32++) >> scale));
int2le16(p, clip_sample((*data1_32++ + dc_bias) >> scale));
p += 2;
}
break;
case STEREO_NONINTERLEAVED:
while (count--) {
int2le16(p, clip_sample((*data1_32++) >> scale));
int2le16(p, clip_sample((*data1_32++ + dc_bias) >> scale));
p += 2;
int2le16(p, clip_sample((*data2_32++) >> scale));
int2le16(p, clip_sample((*data2_32++ + dc_bias) >> scale));
p += 2;
}
@ -285,7 +286,7 @@ static bool pcmbuf_insert_wav(const void *ch1, const void *ch2, int count)
case STEREO_MONO:
while (count--) {
int2le16(p, clip_sample((*data1_32++) >> scale));
int2le16(p, clip_sample((*data1_32++ + dc_bias) >> scale));
p += 2;
}
break;