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-rw-r--r--sound/mixer.cpp15
-rw-r--r--sound/rate.cpp42
-rw-r--r--sound/rate.h20
3 files changed, 36 insertions, 41 deletions
diff --git a/sound/mixer.cpp b/sound/mixer.cpp
index 69d4e0d5af..dbc87f01f7 100644
--- a/sound/mixer.cpp
+++ b/sound/mixer.cpp
@@ -716,8 +716,7 @@ void ChannelRaw::mix(int16 *data, uint len) {
}
const int volume = _mixer->getVolume();
- st_size_t tmpLen = len;
- _converter->flow(*_input, data, &tmpLen, volume);
+ _converter->flow(*_input, data, len, volume);
#else
byte *s, *end;
@@ -837,8 +836,7 @@ void ChannelStream::mix(int16 *data, uint len) {
}
const int volume = _mixer->getVolume();
- st_size_t tmpLen = len;
- _converter->flow(*_input, data, &tmpLen, volume);
+ _converter->flow(*_input, data, len, volume);
#else
if (_pos == _endOfData) {
// Normally, the stream stays around even if all its data is used up.
@@ -960,8 +958,7 @@ void ChannelMP3::mix(int16 *data, uint len) {
}
const int volume = _mixer->getVolume();
- st_size_t tmpLen = len;
- _converter->flow(*_input, data, &tmpLen, volume);
+ _converter->flow(*_input, data, len, volume);
#else
const int volume = _mixer->getVolume();
@@ -1044,8 +1041,7 @@ void ChannelMP3CDMusic::mix(int16 *data, uint len) {
}
const int volume = _mixer->getMusicVolume();
- st_size_t tmpLen = len;
- _converter->flow(*_input, data, &tmpLen, volume);
+ _converter->flow(*_input, data, len, volume);
#else
mad_timer_t frame_duration;
const int volume = _mixer->getMusicVolume();
@@ -1198,8 +1194,7 @@ void ChannelVorbis::mix(int16 *data, uint len) {
}
const int volume = isMusicChannel() ? _mixer->getMusicVolume() : _mixer->getVolume();
- st_size_t tmpLen = len;
- _converter->flow(*_input, data, &tmpLen, volume);
+ _converter->flow(*_input, data, len, volume);
#else
if (_end_pos > 0 && ov_pcm_tell(_ov_file) >= _end_pos) {
destroy();
diff --git a/sound/rate.cpp b/sound/rate.cpp
index 4f05dd326e..5498f48202 100644
--- a/sound/rate.cpp
+++ b/sound/rate.cpp
@@ -66,8 +66,8 @@ protected:
public:
LinearRateConverter(st_rate_t inrate, st_rate_t outrate);
- int flow(AudioInputStream &input, st_sample_t *obuf, st_size_t *osamp, st_volume_t vol);
- int drain(st_sample_t *obuf, st_size_t *osamp, st_volume_t vol) {
+ int flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol);
+ int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol) {
return (ST_SUCCESS);
}
};
@@ -108,14 +108,13 @@ LinearRateConverter<stereo, reverseStereo>::LinearRateConverter(st_rate_t inrate
* Return number of samples processed.
*/
template<bool stereo, bool reverseStereo>
-int LinearRateConverter<stereo, reverseStereo>::flow(AudioInputStream &input, st_sample_t *obuf, st_size_t *osamp, st_volume_t vol)
+int LinearRateConverter<stereo, reverseStereo>::flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol)
{
st_sample_t *ostart, *oend;
- st_sample_t out;
- unsigned long tmp;
+ st_sample_t out[2], tmpOut;
ostart = obuf;
- oend = obuf + *osamp * 2;
+ oend = obuf + osamp * 2;
while (obuf < oend) {
@@ -140,25 +139,25 @@ int LinearRateConverter<stereo, reverseStereo>::flow(AudioInputStream &input, st
while (ipos > opos + 1) {
// interpolate
- out = (st_sample_t) (ilast[reverseStereo ? 1 : 0] + (((icur[reverseStereo ? 1 : 0] - ilast[reverseStereo ? 1 : 0]) * opos_frac + (1UL << (FRAC_BITS-1))) >> FRAC_BITS));
+ tmpOut = (st_sample_t)(ilast[0] + (((icur[0] - ilast[0]) * opos_frac + (1UL << (FRAC_BITS-1))) >> FRAC_BITS));
// adjust volume
- out = out * vol / 256;
-
- // output left channel sample
- clampedAdd(*obuf++, out);
+ out[0] = out[1] = (st_sample_t)((tmpOut * vol) >> 8);
if (stereo) {
// interpolate
- out = (st_sample_t) (ilast[reverseStereo ? 0 : 1] + (((icur[reverseStereo ? 0 : 1] - ilast[reverseStereo ? 0 : 1]) * opos_frac + (1UL << (FRAC_BITS-1))) >> FRAC_BITS));
+ tmpOut = (st_sample_t)(ilast[1] + (((icur[1] - ilast[1]) * opos_frac + (1UL << (FRAC_BITS-1))) >> FRAC_BITS));
// adjust volume
- out = out * vol / 256;
+ out[reverseStereo ? 0 : 1] = (st_sample_t)((tmpOut * vol) >> 8);
}
- // output right channel sample
- clampedAdd(*obuf++, out);
+ // output left channel
+ clampedAdd(*obuf++, out[0]);
+
+ // output right channel
+ clampedAdd(*obuf++, out[1]);
// Increment output position
- tmp = opos_frac + opos_inc_frac;
+ unsigned long tmp = opos_frac + opos_inc_frac;
opos += opos_inc + (tmp >> FRAC_BITS);
opos_frac = tmp & ((1UL << FRAC_BITS) - 1);
@@ -169,7 +168,6 @@ int LinearRateConverter<stereo, reverseStereo>::flow(AudioInputStream &input, st
}
the_end:
- *osamp = (obuf - ostart) / 2;
return (ST_SUCCESS);
}
@@ -183,20 +181,20 @@ the_end:
template<bool stereo, bool reverseStereo>
class CopyRateConverter : public RateConverter {
public:
- virtual int flow(AudioInputStream &input, st_sample_t *obuf, st_size_t *osamp, st_volume_t vol) {
+ virtual int flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol) {
int16 tmp[2];
- st_size_t len = *osamp;
+ st_size_t len = osamp;
assert(input.isStereo() == stereo);
while (!input.eof() && len--) {
- tmp[0] = tmp[1] = input.read() * vol / 256;
+ tmp[0] = tmp[1] = (input.read() * vol) >> 8;
if (stereo)
- tmp[reverseStereo ? 0 : 1] = input.read() * vol / 256;
+ tmp[reverseStereo ? 0 : 1] = (input.read() * vol) >> 8;
clampedAdd(*obuf++, tmp[0]);
clampedAdd(*obuf++, tmp[1]);
}
return (ST_SUCCESS);
}
- virtual int drain(st_sample_t *obuf, st_size_t *osamp, st_volume_t vol) {
+ virtual int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol) {
return (ST_SUCCESS);
}
};
diff --git a/sound/rate.h b/sound/rate.h
index b55897caad..694dbe0b16 100644
--- a/sound/rate.h
+++ b/sound/rate.h
@@ -48,14 +48,14 @@ typedef eff_struct *eff_t;
#define ST_SUCCESS (0)
static inline void clampedAdd(int16& a, int b) {
- int val = a + b;
+ register int val = a + b;
if (val > ST_SAMPLE_MAX)
- a = ST_SAMPLE_MAX;
+ val = ST_SAMPLE_MAX;
else if (val < ST_SAMPLE_MIN)
- a = ST_SAMPLE_MIN;
- else
- a = val;
+ val = ST_SAMPLE_MIN;
+
+ a = val;
}
// Q&D hack to get this SOX stuff to work
@@ -68,19 +68,21 @@ class RateConverter {
public:
RateConverter() {}
virtual ~RateConverter() {}
- virtual int flow(AudioInputStream &input, st_sample_t *obuf, st_size_t *osamp, st_volume_t vol) = 0;
- virtual int drain(st_sample_t *obuf, st_size_t *osamp, st_volume_t vol) = 0;
+ virtual int flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol) = 0;
+ virtual int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol) = 0;
};
+/*
class ResampleRateConverter : public RateConverter {
protected:
eff_struct effp;
public:
ResampleRateConverter(st_rate_t inrate, st_rate_t outrate, int quality);
~ResampleRateConverter();
- virtual int flow(AudioInputStream &input, st_sample_t *obuf, st_size_t *osamp, st_volume_t vol);
- virtual int drain(st_sample_t *obuf, st_size_t *osamp, st_volume_t vol);
+ virtual int flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol);
+ virtual int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol);
};
+*/
RateConverter *makeRateConverter(st_rate_t inrate, st_rate_t outrate, bool stereo, bool reverseStereo = false);
generated by cgit v1.2.3 (git 2.25.1) at 2024-08-02 12:42:06 +0000