Bill Cox
This file is part of the Sonic Library.
This file is licensed under the Apache 2.0 license.
public
class
Sonic {
private
static
final
int
SONIC_MIN_PITCH = 65
;
private
static
final
int
SONIC_MAX_PITCH = 400
;
//
This is used to down-sample some inputs to improve speed
private
static
final
int
SONIC_AMDF_FREQ = 4000
;
//
The number of points to use in the sinc FIR filter for resampling.
private
static
final
int
SINC_FILTER_POINTS = 12
;
private
static
final
int
SINC_TABLE_SIZE = 601
;
//
Lookup table for windowed sinc function of SINC_FILTER_POINTS points.
private
static
final
short
sincTable[] =
{
0, 0, 0, 0, 0, 0, 0, -1, -1, -2, -2, -3, -4, -6, -7, -9, -10, -12, -14
,
-17, -19, -21, -24, -26, -29, -32, -34, -37, -40, -42, -44, -47, -48, -50
,
-51, -52, -53, -53, -53, -52, -50, -48, -46, -43, -39, -34, -29, -22, -16
,
-8, 0, 9, 19, 29, 41, 53, 65, 79, 92, 107, 121, 137, 152, 168, 184, 200
,
215, 231, 247, 262, 276, 291, 304, 317, 328, 339, 348, 357, 363, 369, 372
,
374, 375, 373, 369, 363, 355, 345, 332, 318, 300, 281, 259, 234, 208, 178
,
147, 113, 77, 39, 0, -41, -85, -130, -177, -225, -274, -324, -375, -426
,
-478, -530, -581, -632, -682, -731, -779, -825, -870, -912, -951, -989
,
-1023, -1053, -1080, -1104, -1123, -1138, -1149, -1154, -1155, -1151
,
-1141, -1125, -1105, -1078, -1046, -1007, -963, -913, -857, -796, -728
,
-655, -576, -492, -403, -309, -210, -107, 0, 111, 225, 342, 462, 584, 708
,
833, 958, 1084, 1209, 1333, 1455, 1575, 1693, 1807, 1916, 2022, 2122, 2216
,
2304, 2384, 2457, 2522, 2579, 2625, 2663, 2689, 2706, 2711, 2705, 2687
,
2657, 2614, 2559, 2491, 2411, 2317, 2211, 2092, 1960, 1815, 1658, 1489
,
1308, 1115, 912, 698, 474, 241, 0, -249, -506, -769, -1037, -1310, -1586
,
-1864, -2144, -2424, -2703, -2980, -3254, -3523, -3787, -4043, -4291
,
-4529, -4757, -4972, -5174, -5360, -5531, -5685, -5819, -5935, -6029
,
-6101, -6150, -6175, -6175, -6149, -6096, -6015, -5905, -5767, -5599
,
-5401, -5172, -4912, -4621, -4298, -3944, -3558, -3141, -2693, -2214
,
-1705, -1166, -597, 0, 625, 1277, 1955, 2658, 3386, 4135, 4906, 5697, 6506
,
7332, 8173, 9027, 9893, 10769, 11654, 12544, 13439, 14335, 15232, 16128
,
17019, 17904, 18782, 19649, 20504, 21345, 22170, 22977, 23763, 24527
,
25268, 25982, 26669, 27327, 27953, 28547, 29107, 29632, 30119, 30569
,
30979, 31349, 31678, 31964, 32208, 32408, 32565, 32677, 32744, 32767
,
32744, 32677, 32565, 32408, 32208, 31964, 31678, 31349, 30979, 30569
,
30119, 29632, 29107, 28547, 27953, 27327, 26669, 25982, 25268, 24527
,
23763, 22977, 22170, 21345, 20504, 19649, 18782, 17904, 17019, 16128
,
15232, 14335, 13439, 12544, 11654, 10769, 9893, 9027, 8173, 7332, 6506
,
5697, 4906, 4135, 3386, 2658, 1955, 1277, 625, 0, -597, -1166, -1705
,
-2214, -2693, -3141, -3558, -3944, -4298, -4621, -4912, -5172, -5401
,
-5599, -5767, -5905, -6015, -6096, -6149, -6175, -6175, -6150, -6101
,
-6029, -5935, -5819, -5685, -5531, -5360, -5174, -4972, -4757, -4529
,
-4291, -4043, -3787, -3523, -3254, -2980, -2703, -2424, -2144, -1864
,
-1586, -1310, -1037, -769, -506, -249, 0, 241, 474, 698, 912, 1115, 1308
,
1489, 1658, 1815, 1960, 2092, 2211, 2317, 2411, 2491, 2559, 2614, 2657
,
2687, 2705, 2711, 2706, 2689, 2663, 2625, 2579, 2522, 2457, 2384, 2304
,
2216, 2122, 2022, 1916, 1807, 1693, 1575, 1455, 1333, 1209, 1084, 958, 833
,
708, 584, 462, 342, 225, 111, 0, -107, -210, -309, -403, -492, -576, -655
,
-728, -796, -857, -913, -963, -1007, -1046, -1078, -1105, -1125, -1141
,
-1151, -1155, -1154, -1149, -1138, -1123, -1104, -1080, -1053, -1023, -989
,
-951, -912, -870, -825, -779, -731, -682, -632, -581, -530, -478, -426
,
-375, -324, -274, -225, -177, -130, -85, -41, 0, 39, 77, 113, 147, 178
,
208, 234, 259, 281, 300, 318, 332, 345, 355, 363, 369, 373, 375, 374, 372
,
369, 363, 357, 348, 339, 328, 317, 304, 291, 276, 262, 247, 231, 215, 200
,
184, 168, 152, 137, 121, 107, 92, 79, 65, 53, 41, 29, 19, 9, 0, -8, -16
,
-22, -29, -34, -39, -43, -46, -48, -50, -52, -53, -53, -53, -52, -51, -50
,
-48, -47, -44, -42, -40, -37, -34, -32, -29, -26, -24, -21, -19, -17, -14
,
-12, -10, -9, -7, -6, -4, -3, -2, -2, -1, -1, 0, 0, 0, 0, 0, 0, 0
private
short
inputBuffer[];
private
short
outputBuffer[];
private
short
pitchBuffer[];
private
short
downSampleBuffer[];
private
float
speed;
private
float
volume;
private
float
pitch;
private
float
rate;
private
int
oldRatePosition;
private
int
newRatePosition;
private
boolean
useChordPitch;
private
int
quality;
private
int
numChannels;
private
int
inputBufferSize;
private
int
pitchBufferSize;
private
int
outputBufferSize;
private
int
numInputSamples;
private
int
numOutputSamples;
private
int
numPitchSamples;
private
int
minPeriod;
private
int
maxPeriod;
private
int
maxRequired;
private
int
remainingInputToCopy;
private
int
sampleRate;
private
int
prevPeriod;
private
int
prevMinDiff;
private
int
minDiff;
private
int
maxDiff;
//
Resize the array.
private
short
[] resize(
short
[] oldArray,
int
newLength)
newLength
*=
numChannels;
short
[] newArray =
new
short
[newLength];
int
length = oldArray.length <= newLength?
oldArray.length : newLength;
System.arraycopy(oldArray,
0, newArray, 0
, length);
return
newArray;
//
Move samples from one array to another. May move samples down within an array, but not up.
private
void
move(
short
dest[],
int
destPos,
short
source[],
int
sourcePos,
int
numSamples)
System.arraycopy(source, sourcePos
*numChannels, dest, destPos*numChannels, numSamples*
numChannels);
//
Scale the samples by the factor.
private
void
scaleSamples(
short
samples[],
int
position,
int
numSamples,
float
volume)
int
fixedPointVolume = (
int
)(volume*4096.0f
);
int
start = position*
numChannels;
int
stop = start + numSamples*
numChannels;
for
(
int
xSample = start; xSample < stop; xSample++
) {
int
value = (samples[xSample]*fixedPointVolume) >> 12
;
if
(value > 32767
) {
value
= 32767
;
}
else
if
(value < -32767
) {
value
= -32767
;
samples[xSample]
= (
short
)value;
//
Get the speed of the stream.
public
float
getSpeed()
return
speed;
//
Set the speed of the stream.
public
void
setSpeed(
float
speed)
this
.speed =
speed;
//
Get the pitch of the stream.
public
float
getPitch()
return
pitch;
//
Set the pitch of the stream.
public
void
setPitch(
float
pitch)
this
.pitch =
pitch;
//
Get the rate of the stream.
public
float
getRate()
return
rate;
//
Set the playback rate of the stream. This scales pitch and speed at the same time.
public
void
setRate(
float
rate)
this
.rate =
rate;
this
.oldRatePosition = 0
;
this
.newRatePosition = 0
;
//
Get the vocal chord pitch setting.
public
boolean
getChordPitch()
return
useChordPitch;
//
Set the vocal chord mode for pitch computation. Default is off.
public
void
setChordPitch(
boolean
useChordPitch)
this
.useChordPitch =
useChordPitch;
//
Get the quality setting.
public
int
getQuality()
return
quality;
//
Set the "quality". Default 0 is virtually as good as 1, but very much faster.
public
void
setQuality(
int
quality)
this
.quality =
quality;
//
Get the scaling factor of the stream.
public
float
getVolume()
return
volume;
//
Set the scaling factor of the stream.
public
void
setVolume(
float
volume)
this
.volume =
volume;
//
Allocate stream buffers.
private
void
allocateStreamBuffers(
int
sampleRate,
int
numChannels)
minPeriod
= sampleRate/
SONIC_MAX_PITCH;
maxPeriod
= sampleRate/
SONIC_MIN_PITCH;
maxRequired
= 2*
maxPeriod;
inputBufferSize
=
maxRequired;
inputBuffer
=
new
short
[maxRequired*
numChannels];
outputBufferSize
=
maxRequired;
outputBuffer
=
new
short
[maxRequired*
numChannels];
pitchBufferSize
=
maxRequired;
pitchBuffer
=
new
short
[maxRequired*
numChannels];
downSampleBuffer
=
new
short
[maxRequired];
this
.sampleRate =
sampleRate;
this
.numChannels =
numChannels;
oldRatePosition
= 0
;
newRatePosition
= 0
;
prevPeriod
= 0
;
//
Create a sonic stream.
public
Sonic(
int
sampleRate,
int
numChannels)
allocateStreamBuffers(sampleRate, numChannels);
speed
= 1.0f
;
pitch
= 1.0f
;
volume
= 1.0f
;
rate
= 1.0f
;
oldRatePosition
= 0
;
newRatePosition
= 0
;
useChordPitch
=
false
;
quality
= 0
;
//
Get the sample rate of the stream.
public
int
getSampleRate()
return
sampleRate;
//
Set the sample rate of the stream. This will cause samples buffered in the stream to be lost.
public
void
setSampleRate(
int
sampleRate)
allocateStreamBuffers(sampleRate, numChannels);
//
Get the number of channels.
public
int
getNumChannels()
return
numChannels;
//
Set the num channels of the stream. This will cause samples buffered in the stream to be lost.
public
void
setNumChannels(
int
numChannels)
allocateStreamBuffers(sampleRate, numChannels);
//
Enlarge the output buffer if needed.
private
void
enlargeOutputBufferIfNeeded(
int
numSamples)
if
(numOutputSamples + numSamples >
outputBufferSize) {
outputBufferSize
+= (outputBufferSize >> 1) +
numSamples;
outputBuffer
=
resize(outputBuffer, outputBufferSize);
//
Enlarge the input buffer if needed.
private
void
enlargeInputBufferIfNeeded(
int
numSamples)
if
(numInputSamples + numSamples >
inputBufferSize) {
inputBufferSize
+= (inputBufferSize >> 1) +
numSamples;
inputBuffer
=
resize(inputBuffer, inputBufferSize);
//
Add the input samples to the input buffer.
private
void
addFloatSamplesToInputBuffer(
float
samples[],
int
numSamples)
if
(numSamples == 0
) {
return
;
enlargeInputBufferIfNeeded(numSamples);
int
xBuffer = numInputSamples*
numChannels;
for
(
int
xSample = 0; xSample < numSamples*numChannels; xSample++
) {
inputBuffer[xBuffer
++] = (
short
)(samples[xSample]*32767.0f
);
numInputSamples
+=
numSamples;
//
Add the input samples to the input buffer.
private
void
addShortSamplesToInputBuffer(
short
samples[],
int
numSamples)
if
(numSamples == 0
) {
return
;
enlargeInputBufferIfNeeded(numSamples);
move(inputBuffer, numInputSamples, samples,
0
, numSamples);
numInputSamples
+=
numSamples;
//
Add the input samples to the input buffer.
private
void
addUnsignedByteSamplesToInputBuffer(
byte
samples[],
int
numSamples)
short
sample;
enlargeInputBufferIfNeeded(numSamples);
int
xBuffer = numInputSamples*
numChannels;
for
(
int
xSample = 0; xSample < numSamples*numChannels; xSample++
) {
sample
= (
short
)((samples[xSample] & 0xff) - 128);
//
Convert from unsigned to signed
inputBuffer[xBuffer++] = (
short
) (sample << 8
);
numInputSamples
+=
numSamples;
//
Add the input samples to the input buffer. They must be 16-bit little-endian encoded in a byte array.
private
void
addBytesToInputBuffer(
byte
inBuffer[],
int
numBytes)
int
numSamples = numBytes/(2*
numChannels);
short
sample;
enlargeInputBufferIfNeeded(numSamples);
int
xBuffer = numInputSamples*
numChannels;
for
(
int
xByte = 0; xByte + 1 < numBytes; xByte += 2
) {
sample
= (
short
)((inBuffer[xByte] & 0xff) | (inBuffer[xByte + 1] << 8
));
inputBuffer[xBuffer
++] =
sample;
numInputSamples
+=
numSamples;
//
Remove input samples that we have already processed.
private
void
removeInputSamples(
int
position)
int
remainingSamples = numInputSamples -
position;
move(inputBuffer,
0
, inputBuffer, position, remainingSamples);
numInputSamples
=
remainingSamples;
//
Just copy from the array to the output buffer
private
void
copyToOutput(
short
samples[],
int
position,
int
numSamples)
enlargeOutputBufferIfNeeded(numSamples);
move(outputBuffer, numOutputSamples, samples, position, numSamples);
numOutputSamples
+=
numSamples;
//
Just copy from the input buffer to the output buffer. Return num samples copied.
private
int
copyInputToOutput(
int
position)
int
numSamples =
remainingInputToCopy;
if
(numSamples >
maxRequired) {
numSamples
=
maxRequired;
copyToOutput(inputBuffer, position, numSamples);
remainingInputToCopy
-=
numSamples;
return
numSamples;
//
Read data out of the stream. Sometimes no data will be available, and zero
//
is returned, which is not an error condition.
public
int
readFloatFromStream(
float
samples[],
int
maxSamples)
int
numSamples =
numOutputSamples;
int
remainingSamples = 0
;
if
(numSamples == 0
) {
return
0
;
if
(numSamples >
maxSamples) {
remainingSamples
= numSamples -
maxSamples;
numSamples
=
maxSamples;
for
(
int
xSample = 0; xSample < numSamples*numChannels; xSample++
) {
samples[xSample]
= (outputBuffer[xSample])/32767.0f
;
move(outputBuffer,
0
, outputBuffer, numSamples, remainingSamples);
numOutputSamples
=
remainingSamples;
return
numSamples;
//
Read short data out of the stream. Sometimes no data will be available, and zero
//
is returned, which is not an error condition.
public
int
readShortFromStream(
short
samples[],
int
maxSamples)
int
numSamples =
numOutputSamples;
int
remainingSamples = 0
;
if
(numSamples == 0
) {
return
0
;
if
(numSamples >
maxSamples) {
remainingSamples
= numSamples -
maxSamples;
numSamples
=
maxSamples;
move(samples,
0, outputBuffer, 0
, numSamples);
move(outputBuffer,
0
, outputBuffer, numSamples, remainingSamples);
numOutputSamples
=
remainingSamples;
return
numSamples;
//
Read unsigned byte data out of the stream. Sometimes no data will be available, and zero
//
is returned, which is not an error condition.
public
int
readUnsignedByteFromStream(
byte
samples[],
int
maxSamples)
int
numSamples =
numOutputSamples;
int
remainingSamples = 0
;
if
(numSamples == 0
) {
return
0
;
if
(numSamples >
maxSamples) {
remainingSamples
= numSamples -
maxSamples;
numSamples
=
maxSamples;
for
(
int
xSample = 0; xSample < numSamples*numChannels; xSample++
) {
samples[xSample]
= (
byte
)((outputBuffer[xSample] >> 8) + 128
);
move(outputBuffer,
0
, outputBuffer, numSamples, remainingSamples);
numOutputSamples
=
remainingSamples;
return
numSamples;
//
Read unsigned byte data out of the stream. Sometimes no data will be available, and zero
//
is returned, which is not an error condition.
public
int
readBytesFromStream(
byte
outBuffer[],
int
maxBytes)
int
maxSamples = maxBytes/(2*
numChannels);
int
numSamples =
numOutputSamples;
int
remainingSamples = 0
;
if
(numSamples == 0 || maxSamples == 0
) {
return
0
;
if
(numSamples >
maxSamples) {
remainingSamples
= numSamples -
maxSamples;
numSamples
=
maxSamples;
for
(
int
xSample = 0; xSample < numSamples*numChannels; xSample++
) {
short
sample =
outputBuffer[xSample];
outBuffer[xSample
<< 1] = (
byte
)(sample & 0xff
);
outBuffer[(xSample
<< 1) + 1] = (
byte
)(sample >> 8
);
move(outputBuffer,
0
, outputBuffer, numSamples, remainingSamples);
numOutputSamples
=
remainingSamples;
return
2*numSamples*
numChannels;
//
Force the sonic stream to generate output using whatever data it currently
//
has. No extra delay will be added to the output, but flushing in the middle of
//
words could introduce distortion.
public
void
flushStream()
int
remainingSamples =
numInputSamples;
float
s = speed/
pitch;
float
r = rate*
pitch;
int
expectedOutputSamples = numOutputSamples + (
int
)((remainingSamples/s + numPitchSamples)/r + 0.5f
);
//
Add enough silence to flush both input and pitch buffers.
enlargeInputBufferIfNeeded(remainingSamples + 2*
maxRequired);
for
(
int
xSample = 0; xSample < 2*maxRequired*numChannels; xSample++
) {
inputBuffer[remainingSamples
*numChannels + xSample] = 0
;
numInputSamples
+= 2*
maxRequired;
writeShortToStream(
null
, 0
);
//
Throw away any extra samples we generated due to the silence we added.
if
(numOutputSamples >
expectedOutputSamples) {
numOutputSamples
=
expectedOutputSamples;
//
Empty input and pitch buffers.
numInputSamples = 0
;
remainingInputToCopy
= 0
;
numPitchSamples
= 0
;
//
Return the number of samples in the output buffer
public
int
samplesAvailable()
return
numOutputSamples;
//
If skip is greater than one, average skip samples together and write them to
//
the down-sample buffer. If numChannels is greater than one, mix the channels
//
together as we down sample.
private
void
downSampleInput(
short
samples[],
int
position,
int
skip)
int
numSamples = maxRequired/
skip;
int
samplesPerValue = numChannels*
skip;
int
value;
position
*=
numChannels;
for
(
int
i = 0; i < numSamples; i++
) {
value
= 0
;
for
(
int
j = 0; j < samplesPerValue; j++
) {
value
+= samples[position + i*samplesPerValue +
j];
value
/=
samplesPerValue;
downSampleBuffer[i]
= (
short
)value;
//
Find the best frequency match in the range, and given a sample skip multiple.
//
For now, just find the pitch of the first channel.
private
int
findPitchPeriodInRange(
short
samples[],
int
position,
int
minPeriod,
int
maxPeriod)
int
bestPeriod = 0, worstPeriod = 255
;
int
minDiff = 1, maxDiff = 0
;
position
*=
numChannels;
for
(
int
period = minPeriod; period <= maxPeriod; period++
) {
int
diff = 0
;
for
(
int
i = 0; i < period; i++
) {
short
sVal = samples[position +
i];
short
pVal = samples[position + period +
i];
diff
+= sVal >= pVal? sVal - pVal : pVal -
sVal;
/*
Note that the highest number of samples we add into diff will be less
than 256, since we skip samples. Thus, diff is a 24 bit number, and
we can safely multiply by numSamples without overflow
*/
if
(diff*bestPeriod < minDiff*
period) {
minDiff
=
diff;
bestPeriod
=
period;
if
(diff*worstPeriod > maxDiff*
period) {
maxDiff
=
diff;
worstPeriod
=
period;
this
.minDiff = minDiff/
bestPeriod;
this
.maxDiff = maxDiff/
worstPeriod;
return
bestPeriod;
//
At abrupt ends of voiced words, we can have pitch periods that are better
//
approximated by the previous pitch period estimate. Try to detect this case.
private
boolean
prevPeriodBetter(
int
minDiff,
int
maxDiff,
boolean
preferNewPeriod)
if
(minDiff == 0 || prevPeriod == 0
) {
return
false
;
if
(preferNewPeriod) {
if
(maxDiff > minDiff*3
) {
//
Got a reasonable match this period
return
false
;
if
(minDiff*2 <= prevMinDiff*3
) {
//
Mismatch is not that much greater this period
return
false
;
}
else
{
if
(minDiff <=
prevMinDiff) {
return
false
;
return
true
;
//
Find the pitch period. This is a critical step, and we may have to try
//
multiple ways to get a good answer. This version uses AMDF. To improve
//
speed, we down sample by an integer factor get in the 11KHz range, and then
//
do it again with a narrower frequency range without down sampling
private
int
findPitchPeriod(
short
samples[],
int
position,
boolean
preferNewPeriod)
int
period, retPeriod;
int
skip = 1
;
if
(sampleRate > SONIC_AMDF_FREQ && quality == 0
) {
skip
= sampleRate/
SONIC_AMDF_FREQ;
if
(numChannels == 1 && skip == 1
) {
period
=
findPitchPeriodInRange(samples, position, minPeriod, maxPeriod);
}
else
{
downSampleInput(samples, position, skip);
period
= findPitchPeriodInRange(downSampleBuffer, 0, minPeriod/
skip,
maxPeriod
/
skip);
if
(skip != 1
) {
period
*=
skip;
int
minP = period - (skip << 2
);
int
maxP = period + (skip << 2
);
if
(minP <
minPeriod) {
minP
=
minPeriod;
if
(maxP >
maxPeriod) {
maxP
=
maxPeriod;
if
(numChannels == 1
) {
period
=
findPitchPeriodInRange(samples, position, minP, maxP);
}
else
{
downSampleInput(samples, position,
1
);
period
= findPitchPeriodInRange(downSampleBuffer, 0
, minP, maxP);
if
(prevPeriodBetter(minDiff, maxDiff, preferNewPeriod)) {
retPeriod
=
prevPeriod;
}
else
{
retPeriod
=
period;
prevMinDiff
=
minDiff;
prevPeriod
=
period;
return
retPeriod;
//
Overlap two sound segments, ramp the volume of one down, while ramping the
//
other one from zero up, and add them, storing the result at the output.
private
void
overlapAdd(
int
numSamples,
int
numChannels,
short
out[],
int
outPos,
short
rampDown[],
int
rampDownPos,
short
rampUp[],
int
rampUpPos)
for
(
int
i = 0; i < numChannels; i++
) {
int
o = outPos*numChannels +
i;
int
u = rampUpPos*numChannels +
i;
int
d = rampDownPos*numChannels +
i;
for
(
int
t = 0; t < numSamples; t++
) {
out[o]
= (
short
)((rampDown[d]*(numSamples - t) + rampUp[u]*t)/
numSamples);
o
+=
numChannels;
d
+=
numChannels;
u
+=
numChannels;
//
Overlap two sound segments, ramp the volume of one down, while ramping the
//
other one from zero up, and add them, storing the result at the output.
private
void
overlapAddWithSeparation(
int
numSamples,
int
numChannels,
int
separation,
short
out[],
int
outPos,
short
rampDown[],
int
rampDownPos,
short
rampUp[],
int
rampUpPos)
for
(
int
i = 0; i < numChannels; i++
) {
int
o = outPos*numChannels +
i;
int
u = rampUpPos*numChannels +
i;
int
d = rampDownPos*numChannels +
i;
for
(
int
t = 0; t < numSamples + separation; t++
) {
if
(t <
separation) {
out[o]
= (
short
)(rampDown[d]*(numSamples - t)/
numSamples);
d
+=
numChannels;
}
else
if
(t <
numSamples) {
out[o]
= (
short
)((rampDown[d]*(numSamples - t) + rampUp[u]*(t - separation))/
numSamples);
d
+=
numChannels;
u
+=
numChannels;
}
else
{
out[o]
= (
short
)(rampUp[u]*(t - separation)/
numSamples);
u
+=
numChannels;
o
+=
numChannels;
//
Just move the new samples in the output buffer to the pitch buffer
private
void
moveNewSamplesToPitchBuffer(
int
originalNumOutputSamples)
int
numSamples = numOutputSamples -
originalNumOutputSamples;
if
(numPitchSamples + numSamples >
pitchBufferSize) {
pitchBufferSize
+= (pitchBufferSize >> 1) +
numSamples;
pitchBuffer
=
resize(pitchBuffer, pitchBufferSize);
move(pitchBuffer, numPitchSamples, outputBuffer, originalNumOutputSamples, numSamples);
numOutputSamples
=
originalNumOutputSamples;
numPitchSamples
+=
numSamples;
//
Remove processed samples from the pitch buffer.
private
void
removePitchSamples(
int
numSamples)
if
(numSamples == 0
) {
return
;
move(pitchBuffer,
0, pitchBuffer, numSamples, numPitchSamples -
numSamples);
numPitchSamples
-=
numSamples;
//
Change the pitch. The latency this introduces could be reduced by looking at
//
past samples to determine pitch, rather than future.
private
void
adjustPitch(
int
originalNumOutputSamples)
int
period, newPeriod, separation;
int
position = 0
;
if
(numOutputSamples ==
originalNumOutputSamples) {
return
;
moveNewSamplesToPitchBuffer(originalNumOutputSamples);
while
(numPitchSamples - position >=
maxRequired) {
period
= findPitchPeriod(pitchBuffer, position,
false
);
newPeriod
= (
int
)(period/
pitch);
enlargeOutputBufferIfNeeded(newPeriod);
if
(pitch >= 1.0f
) {
overlapAdd(newPeriod, numChannels, outputBuffer, numOutputSamples, pitchBuffer,
position, pitchBuffer, position
+ period -
newPeriod);
}
else
{
separation
= newPeriod -
period;
overlapAddWithSeparation(period, numChannels, separation, outputBuffer, numOutputSamples,
pitchBuffer, position, pitchBuffer, position);
numOutputSamples
+=
newPeriod;
position
+=
period;
removePitchSamples(position);
//
Aproximate the sinc function times a Hann window from the sinc table.
private
int
findSincCoefficient(
int
i,
int
ratio,
int
width) {
int
lobePoints = (SINC_TABLE_SIZE-1)/
SINC_FILTER_POINTS;
int
left = i*lobePoints + (ratio*lobePoints)/
width;
int
right = left + 1
;
int
position = i*lobePoints*width + ratio*lobePoints - left*
width;
int
leftVal =
sincTable[left];
int
rightVal =
sincTable[right];
return
((leftVal*(width - position) + rightVal*position) << 1)/
width;
//
Return 1 if value >= 0, else -1. This represents the sign of value.
private
int
getSign(
int
value) {
return
value >= 0? 1 : -1
;
//
Interpolate the new output sample.
private
short
interpolate(
short
in[],
int
inPos,
//
Index to first sample which already includes channel offset.
int
oldSampleRate,
int
newSampleRate)
//
Compute N-point sinc FIR-filter here. Clip rather than overflow.
int
i;
int
total = 0
;
int
position = newRatePosition*
oldSampleRate;
int
leftPosition = oldRatePosition*
newSampleRate;
int
rightPosition = (oldRatePosition + 1)*
newSampleRate;
int
ratio = rightPosition - position - 1
;
int
width = rightPosition -
leftPosition;
int
weight, value;
int
oldSign;
int
overflowCount = 0
;
for
(i = 0; i < SINC_FILTER_POINTS; i++
) {
weight
=
findSincCoefficient(i, ratio, width);
/*
printf("%u %f\n", i, weight);
*/
value
= in[inPos + i*numChannels]*
weight;
oldSign
=
getSign(total);
total
+=
value;
if
(oldSign != getSign(total) && getSign(value) ==
oldSign) {
/*
We must have overflowed. This can happen with a sinc filter.
*/
overflowCount
+=
oldSign;
/*
It is better to clip than to wrap if there was a overflow.
*/
if
(overflowCount > 0
) {
return
Short.MAX_VALUE;
}
else
if
(overflowCount < 0
) {
return
Short.MIN_VALUE;
return
(
short
)(total >> 16
);
//
Change the rate.
private
void
adjustRate(
float
rate,
int
originalNumOutputSamples)
int
newSampleRate = (
int
)(sampleRate/
rate);
int
oldSampleRate =
sampleRate;
int
position;
//
Set these values to help with the integer math
while
(newSampleRate > (1 << 14) || oldSampleRate > (1 << 14
)) {
newSampleRate
>>= 1
;
oldSampleRate
>>= 1
;
if
(numOutputSamples ==
originalNumOutputSamples) {
return
;
moveNewSamplesToPitchBuffer(originalNumOutputSamples);
//
Leave at least one pitch sample in the buffer
for
(position = 0; position < numPitchSamples - 1; position++
) {
while
((oldRatePosition + 1)*newSampleRate > newRatePosition*
oldSampleRate) {
enlargeOutputBufferIfNeeded(
1
);
for
(
int
i = 0; i < numChannels; i++
) {
outputBuffer[numOutputSamples
*numChannels + i] =
interpolate(pitchBuffer,
position
*numChannels +
i, oldSampleRate, newSampleRate);
newRatePosition
++
;
numOutputSamples
++
;
oldRatePosition
++
;
if
(oldRatePosition ==
oldSampleRate) {
oldRatePosition
= 0
;
if
(newRatePosition !=
newSampleRate) {
System.out.printf(
"Assertion failed: newRatePosition != newSampleRate\n"
);
assert
false
;
newRatePosition
= 0
;
removePitchSamples(position);
//
Skip over a pitch period, and copy period/speed samples to the output
private
int
skipPitchPeriod(
short
samples[],
int
position,
float
speed,
int
period)
int
newSamples;
if
(speed >= 2.0f
) {
newSamples
= (
int
)(period/(speed - 1.0f
));
}
else
{
newSamples
=
period;
remainingInputToCopy
= (
int
)(period*(2.0f - speed)/(speed - 1.0f
));
enlargeOutputBufferIfNeeded(newSamples);
overlapAdd(newSamples, numChannels, outputBuffer, numOutputSamples, samples, position,
samples, position
+
period);
numOutputSamples
+=
newSamples;
return
newSamples;
//
Insert a pitch period, and determine how much input to copy directly.
private
int
insertPitchPeriod(
short
samples[],
int
position,
float
speed,
int
period)
int
newSamples;
if
(speed < 0.5f
) {
newSamples
= (
int
)(period*speed/(1.0f -
speed));
}
else
{
newSamples
=
period;
remainingInputToCopy
= (
int
)(period*(2.0f*speed - 1.0f)/(1.0f -
speed));
enlargeOutputBufferIfNeeded(period
+
newSamples);
move(outputBuffer, numOutputSamples, samples, position, period);
overlapAdd(newSamples, numChannels, outputBuffer, numOutputSamples
+
period, samples,
position
+
period, samples, position);
numOutputSamples
+= period +
newSamples;
return
newSamples;
//
Resample as many pitch periods as we have buffered on the input. Return 0 if
//
we fail to resize an input or output buffer. Also scale the output by the volume.
private
void
changeSpeed(
float
speed)
int
numSamples =
numInputSamples;
int
position = 0
, period, newSamples;
if
(numInputSamples <
maxRequired) {
return
;
if
(remainingInputToCopy > 0
) {
newSamples
=
copyInputToOutput(position);
position
+=
newSamples;
}
else
{
period
= findPitchPeriod(inputBuffer, position,
true
);
if
(speed > 1.0
) {
newSamples
=
skipPitchPeriod(inputBuffer, position, speed, period);
position
+= period +
newSamples;
}
else
{
newSamples
=
insertPitchPeriod(inputBuffer, position, speed, period);
position
+=
newSamples;
}
while
(position + maxRequired <=
numSamples);
removeInputSamples(position);
//
Resample as many pitch periods as we have buffered on the input. Scale the output by the volume.
private
void
processStreamInput()
int
originalNumOutputSamples =
numOutputSamples;
float
s = speed/
pitch;
float
r =
rate;
if
(!
useChordPitch) {
r
*=
pitch;
if
(s > 1.00001 || s < 0.99999
) {
changeSpeed(s);
}
else
{
copyToOutput(inputBuffer,
0
, numInputSamples);
numInputSamples
= 0
;
if
(useChordPitch) {
if
(pitch != 1.0f
) {
adjustPitch(originalNumOutputSamples);
}
else
if
(r != 1.0f
) {
adjustRate(r, originalNumOutputSamples);
if
(volume != 1.0f
) {
//
Adjust output volume.
scaleSamples(outputBuffer, originalNumOutputSamples, numOutputSamples -
originalNumOutputSamples,
volume);
//
Write floating point data to the input buffer and process it.
public
void
writeFloatToStream(
float
samples[],
int
numSamples)
addFloatSamplesToInputBuffer(samples, numSamples);
processStreamInput();
//
Write the data to the input stream, and process it.
public
void
writeShortToStream(
short
samples[],
int
numSamples)
addShortSamplesToInputBuffer(samples, numSamples);
processStreamInput();
//
Simple wrapper around sonicWriteFloatToStream that does the unsigned byte to short
//
conversion for you.
public
void
writeUnsignedByteToStream(
byte
samples[],
int
numSamples)
addUnsignedByteSamplesToInputBuffer(samples, numSamples);
processStreamInput();
//
Simple wrapper around sonicWriteBytesToStream that does the byte to 16-bit LE conversion.
public
void
writeBytesToStream(
byte
inBuffer[],
int
numBytes)
addBytesToInputBuffer(inBuffer, numBytes);
processStreamInput();
//
This is a non-stream oriented interface to just change the speed of a sound sample
public
static
int
changeFloatSpeed(
float
samples[],
int
numSamples,
float
speed,
float
pitch,
float
rate,
float
volume,
boolean
useChordPitch,
int
sampleRate,
int
numChannels)
Sonic stream
=
new
Sonic(sampleRate, numChannels);
stream.setSpeed(speed);
stream.setPitch(pitch);
stream.setRate(rate);
stream.setVolume(volume);
stream.setChordPitch(useChordPitch);
stream.writeFloatToStream(samples, numSamples);
stream.flushStream();
numSamples
=
stream.samplesAvailable();
stream.readFloatFromStream(samples, numSamples);
return
numSamples;
/*
This is a non-stream oriented interface to just change the speed of a sound sample
*/
public
int
sonicChangeShortSpeed(
short
samples[],
int
numSamples,
float
speed,
float
pitch,
float
rate,
float
volume,
boolean
useChordPitch,
int
sampleRate,
int
numChannels)
Sonic stream
=
new
Sonic(sampleRate, numChannels);
stream.setSpeed(speed);
stream.setPitch(pitch);
stream.setRate(rate);
stream.setVolume(volume);
stream.setChordPitch(useChordPitch);
stream.writeShortToStream(samples, numSamples);
stream.flushStream();
numSamples
=
stream.samplesAvailable();
stream.readShortFromStream(samples, numSamples);
return
numSamples;
1、变速不变音
/* This file was written by Bill Cox in 2011, and is licensed under the Apache 2.0 license. */ import com.google.common.io.ByteArrayDataOutput; import com.google.common.io.ByteStreams; import com.xxx.Sonic; import com.sun.media.sound.WaveFileWriter; import javax.sound.sampled.AudioFileFormat; import javax.sound.sampled.AudioFormat; import javax.sound.sampled.AudioInputStream; import javax.sound.sampled.AudioSystem; import javax.sound.sampled.DataLine; import javax.sound.sampled.LineUnavailableException; import javax.sound.sampled.SourceDataLine; import javax.sound.sampled.UnsupportedAudioFileException; import java.io.ByteArrayInputStream; import java.io.File; import java.io.FileOutputStream; import java.io.IOException; import java.io.InputStream; import java.io.OutputStream; public class TestSonic { static ByteArrayDataOutput out = ByteStreams.newDataOutput(); // Run sonic. private static void runSonic( AudioInputStream audioStream, SourceDataLine line, float speed, float pitch, float rate, float volume, boolean emulateChordPitch, int quality, int sampleRate, int numChannels) throws IOException Sonic sonic = new Sonic(sampleRate, numChannels); int bufferSize = line.getBufferSize(); byte inBuffer[] = new byte[bufferSize]; byte outBuffer[] = new byte[bufferSize]; int numRead, numWritten; sonic.setSpeed(speed); sonic.setPitch(pitch); sonic.setRate(rate); sonic.setVolume(volume); sonic.setChordPitch(emulateChordPitch); sonic.setQuality(quality); numRead = audioStream.read(inBuffer, 0, bufferSize); if(numRead <= 0) { sonic.flushStream(); } else { sonic.writeBytesToStream(inBuffer, numRead); numWritten = sonic.readBytesFromStream(outBuffer, bufferSize); if(numWritten > 0) { line.write(outBuffer, 0, numWritten); byte[] target = new byte[numWritten]; System.arraycopy(outBuffer, 0, target, 0, numWritten); out.write(target); } while(numWritten > 0); } while(numRead > 0); saveFile(out.toByteArray()); public static void saveFile(byte[] bytes) String fileName = "c:/"+ System.currentTimeMillis() + ".wav"; OutputStream outStream = null; try { outStream = new FileOutputStream(new File(fileName)); WaveFileWriter writer = new WaveFileWriter(); AudioFormat frmt = new AudioFormat(16000, 16, 1, true, false); ByteArrayInputStream bi = new ByteArrayInputStream(bytes); InputStream in = new AudioInputStream(bi, frmt,bytes.length); writer.write((AudioInputStream) in, AudioFileFormat.Type.WAVE, outStream); outStream.close(); } catch (Exception e) { e.printStackTrace(); public static void main( String[] argv) throws UnsupportedAudioFileException, IOException, LineUnavailableException float speed = 1.5f;//1.5倍速度播放 float pitch = 1.0f; float rate = 1.0f; float volume = 1.0f; boolean emulateChordPitch = false; int quality = 0; AudioInputStream stream = AudioSystem.getAudioInputStream(new File("C:\\Users\\Administrator\\AppData\\Local\\Temp\\aefee1de-f856-4532-8a1c-79a00c2517dd\\generated.wav")); AudioFormat format = stream.getFormat(); int sampleRate = (int)format.getSampleRate(); int numChannels = format.getChannels(); SourceDataLine.Info info = new DataLine.Info(SourceDataLine.class, format, ((int)stream.getFrameLength()*format.getFrameSize())); SourceDataLine line = (SourceDataLine)AudioSystem.getLine(info); line.open(stream.getFormat()); line.start(); runSonic(stream, line, speed, pitch, rate, volume, emulateChordPitch, quality, sampleRate, numChannels); line.drain(); line.stop();2、调整音量
package com.xxx; /* This file was written by Bill Cox in 2011, and is licensed under the Apache 2.0 license. */ import com.google.common.io.ByteArrayDataOutput; import com.google.common.io.ByteStreams; import com.xxx.web.open.util.Sonic; import com.sun.media.sound.WaveFileWriter; import javax.sound.sampled.AudioFileFormat; import javax.sound.sampled.AudioFormat; import javax.sound.sampled.AudioInputStream; import javax.sound.sampled.AudioSystem; import javax.sound.sampled.DataLine; import javax.sound.sampled.LineUnavailableException; import javax.sound.sampled.SourceDataLine; import javax.sound.sampled.UnsupportedAudioFileException; import java.io.ByteArrayInputStream; import java.io.File; import java.io.FileOutputStream; import java.io.IOException; import java.io.InputStream; import java.io.OutputStream; public class TestSonic { static ByteArrayDataOutput out = ByteStreams.newDataOutput(); // Run sonic. private static void runSonic( AudioInputStream audioStream, SourceDataLine line, float speed, float pitch, float rate, float volume, boolean emulateChordPitch, int quality, int sampleRate, int numChannels) throws IOException Sonic sonic = new Sonic(sampleRate, numChannels); int bufferSize = line.getBufferSize(); byte inBuffer[] = new byte[bufferSize]; byte outBuffer[] = new byte[bufferSize]; int numRead, numWritten; sonic.setSpeed(speed); sonic.setPitch(pitch); sonic.setRate(rate); sonic.setVolume(volume); sonic.setChordPitch(emulateChordPitch); sonic.setQuality(quality); numRead = audioStream.read(inBuffer, 0, bufferSize); if(numRead <= 0) { sonic.flushStream(); } else { sonic.writeBytesToStream(inBuffer, numRead); numWritten = sonic.readBytesFromStream(outBuffer, bufferSize); if(numWritten > 0) { line.write(outBuffer, 0, numWritten); byte[] target = new byte[numWritten]; System.arraycopy(outBuffer, 0, target, 0, numWritten); out.write(target); } while(numWritten > 0); } while(numRead > 0); //saveFile(out.toByteArray()); public static void saveFile(byte[] bytes) String fileName = "c:/"+ System.currentTimeMillis() + ".wav"; OutputStream outStream = null; try { outStream = new FileOutputStream(new File(fileName)); WaveFileWriter writer = new WaveFileWriter(); AudioFormat frmt = new AudioFormat(16000, 16, 1, true, false); ByteArrayInputStream bi = new ByteArrayInputStream(bytes); InputStream in = new AudioInputStream(bi, frmt,bytes.length); writer.write((AudioInputStream) in, AudioFileFormat.Type.WAVE, outStream); outStream.close(); } catch (Exception e) { e.printStackTrace(); public static void main( String[] argv) throws UnsupportedAudioFileException, IOException, LineUnavailableException float speed = 1.0f; float pitch = 1.0f; float rate = 1.0f; float volume = 0.5f; //调整音量 boolean emulateChordPitch = false; int quality = 0; AudioInputStream stream = AudioSystem.getAudioInputStream(new File("d:\\generated.wav")); AudioFormat format = stream.getFormat(); int sampleRate = (int)format.getSampleRate(); int numChannels = format.getChannels(); SourceDataLine.Info info = new DataLine.Info(SourceDataLine.class, format, ((int)stream.getFrameLength()*format.getFrameSize())); SourceDataLine line = (SourceDataLine)AudioSystem.getLine(info); line.open(stream.getFormat()); line.start(); runSonic(stream, line, speed, pitch, rate, volume, emulateChordPitch, quality, sampleRate, numChannels); line.drain(); line.stop();package com.xxx; /* This file was written by Bill Cox in 2011, and is licensed under the Apache 2.0 license. */ import com.google.common.io.ByteArrayDataOutput; import com.google.common.io.ByteStreams; import com.xxx.web.open.util.Sonic; import com.sun.media.sound.WaveFileWriter; import javax.sound.sampled.AudioFileFormat; import javax.sound.sampled.AudioFormat; import javax.sound.sampled.AudioInputStream; import javax.sound.sampled.AudioSystem; import javax.sound.sampled.DataLine; import javax.sound.sampled.LineUnavailableException; import javax.sound.sampled.SourceDataLine; import javax.sound.sampled.UnsupportedAudioFileException; import java.io.ByteArrayInputStream; import java.io.File; import java.io.FileOutputStream; import java.io.IOException; import java.io.InputStream; import java.io.OutputStream; public class TestSonic { static ByteArrayDataOutput out = ByteStreams.newDataOutput(); // Run sonic. private static void runSonic( AudioInputStream audioStream, SourceDataLine line, float speed, float pitch, float rate, float volume, boolean emulateChordPitch, int quality, int sampleRate, int numChannels) throws IOException Sonic sonic = new Sonic(sampleRate, numChannels); int bufferSize = line.getBufferSize(); byte inBuffer[] = new byte[bufferSize]; byte outBuffer[] = new byte[bufferSize]; int numRead, numWritten; sonic.setSpeed(speed); sonic.setPitch(pitch); sonic.setRate(rate); sonic.setVolume(volume); sonic.setChordPitch(emulateChordPitch); sonic.setQuality(quality); numRead = audioStream.read(inBuffer, 0, bufferSize); if(numRead <= 0) { sonic.flushStream(); } else { sonic.writeBytesToStream(inBuffer, numRead); numWritten = sonic.readBytesFromStream(outBuffer, bufferSize); if(numWritten > 0) { line.write(outBuffer, 0, numWritten); byte[] target = new byte[numWritten]; System.arraycopy(outBuffer, 0, target, 0, numWritten); out.write(target); } while(numWritten > 0); } while(numRead > 0); //saveFile(out.toByteArray()); public static void saveFile(byte[] bytes) String fileName = "c:/"+ System.currentTimeMillis() + ".wav"; OutputStream outStream = null; try { outStream = new FileOutputStream(new File(fileName)); WaveFileWriter writer = new WaveFileWriter(); AudioFormat frmt = new AudioFormat(16000, 16, 1, true, false); ByteArrayInputStream bi = new ByteArrayInputStream(bytes); InputStream in = new AudioInputStream(bi, frmt,bytes.length); writer.write((AudioInputStream) in, AudioFileFormat.Type.WAVE, outStream); outStream.close(); } catch (Exception e) { e.printStackTrace(); public static void main( String[] argv) throws UnsupportedAudioFileException, IOException, LineUnavailableException float speed = 1.0f; float pitch = 1.500f; //大于1音调变高，小于1音调变低 float rate = 1.0f; float volume = 0.5f; boolean emulateChordPitch = false; int quality = 0; AudioInputStream stream = AudioSystem.getAudioInputStream(new File("d:\\generated.wav")); AudioFormat format = stream.getFormat(); int sampleRate = (int)format.getSampleRate(); int numChannels = format.getChannels(); SourceDataLine.Info info = new DataLine.Info(SourceDataLine.class, format, ((int)stream.getFrameLength()*format.getFrameSize())); SourceDataLine line = (SourceDataLine)AudioSystem.getLine(info); line.open(stream.getFormat()); line.start(); runSonic(stream, line, speed, pitch, rate, volume, emulateChordPitch, quality, sampleRate, numChannels); line.drain(); line.stop();当然你也可以同时设置 volume,speed,pitch
本博客文章绝大多数为原创，少量为转载，代码经过测试验证，如果有疑问直接留言或者私信我。
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