moved volume and pan control into flow func

svn-id: r9976

3 files changed, 19 insertions, 19 deletions

diff --git a/sound/mixer.cpp b/sound/mixer.cpp
index 557d9d37ce..cf11e62848 100644
--- a/sound/mixer.cpp
+++ b/sound/mixer.cpp
@@ -487,17 +487,6 @@ void Channel::destroy() {
 	delete this;
 }
 
-static void changeVolumeAndPan(int16 *data, uint len, byte volume, int8 pan) {
-	byte volumeLeft = abs(pan - 128);
-	byte volumeRight = abs(pan + 128);
-	for (uint32 i = 0; i < len; i++) {
-		int16 sampleL = data[i * 2 + 0];
-		int16 sampleR = data[i * 2 + 1];
-		data[i * 2 + 0] = (sampleL * volume * volumeLeft) / (256 * 256);
-		data[i * 2 + 1] = (sampleR * volume * volumeRight) / (256 * 256);
-	}
-}
-
 /* len indicates the number of sample *pairs*. So a value of
    10 means that the buffer contains twice 10 sample, each
    16 bits, for a total of 40 bytes.
@@ -509,8 +498,7 @@ void Channel::mix(int16 *data, uint len) {
 		destroy();
 	} else {
 		assert(_converter);
-		_converter->flow(*_input, data, len, getVolume());
-		changeVolumeAndPan(data, len, _volume, _pan);
+		_converter->flow(*_input, data, len, getVolume(), _volume, _pan);
 	}
 }
 
@@ -586,8 +574,7 @@ void ChannelStream::mix(int16 *data, uint len) {
 	}
 
 	assert(_converter);
-	_converter->flow(*_input, data, len, getVolume());
-	changeVolumeAndPan(data, len, _volume, _pan);
+	_converter->flow(*_input, data, len, getVolume(), _volume, _pan);
 }
 
 #ifdef USE_MAD
diff --git a/sound/rate.cpp b/sound/rate.cpp
index a99794c6ef..e4f9fb6b19 100644
--- a/sound/rate.cpp
+++ b/sound/rate.cpp
@@ -80,7 +80,7 @@ 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 flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol, byte vol_p, int8 pan);
 	int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol) {
 		return (ST_SUCCESS);
 	}
@@ -124,7 +124,7 @@ 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, byte vol_p, int8 pan)
 {
 	st_sample_t *ostart, *oend;
 	st_sample_t out[2], tmpOut;
@@ -170,6 +170,13 @@ int LinearRateConverter<stereo, reverseStereo>::flow(AudioInputStream &input, st
 				out[reverseStereo ? 0 : 1] = (st_sample_t)((tmpOut * vol) >> 8);
 			}
 	
+			byte pan_l = abs(pan - 128);
+			byte pan_r = abs(pan + 128);
+			out[0] = (st_sample_t)((out[0] * vol_p) >> 8);
+			out[1] = (st_sample_t)((out[1] * vol_p) >> 8);
+			out[0] = (st_sample_t)((out[0] * pan_l) >> 8);
+			out[1] = (st_sample_t)((out[1] * pan_r) >> 8);
+
 			// output left channel
 			clampedAdd(*obuf++, out[0]);
 	
@@ -201,7 +208,7 @@ 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, byte vol_p, int8 pan) {
 		int16 tmp[2];
 		st_size_t len = osamp;
 		assert(input.isStereo() == stereo);
@@ -209,6 +216,12 @@ public:
 			tmp[0] = tmp[1] = (input.read() * vol) >> 8;
 			if (stereo)
 				tmp[reverseStereo ? 0 : 1] = (input.read() * vol) >> 8;
+			byte pan_l = abs(pan - 128);
+			byte pan_r = abs(pan + 128);
+			tmp[0] = ((tmp[0] * vol_p) >> 8);
+			tmp[1] = ((tmp[1] * vol_p) >> 8);
+			tmp[0] = ((tmp[0] * pan_l) >> 8);
+			tmp[1] = ((tmp[1] * pan_r) >> 8);
 			clampedAdd(*obuf++, tmp[0]);
 			clampedAdd(*obuf++, tmp[1]);
 		}
diff --git a/sound/rate.h b/sound/rate.h
index 67224b2377..c693a165bd 100644
--- a/sound/rate.h
+++ b/sound/rate.h
@@ -62,7 +62,7 @@ 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 flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol, byte vol_p, int8 pan) = 0;
 	virtual int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol) = 0;
 };