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authorEugene Sandulenko2015-11-09 16:39:17 +0100
committerEugene Sandulenko2015-11-09 16:39:17 +0100
commit82c98e98033eafa2ed04febe2607f09636e7e6a5 (patch)
tree450e3e543c839c6b89aedf1a1e329fd9072d51ee /audio/rate.cpp
parent30b6682130e5aefe1e019eb37c0cd25b5831d225 (diff)
parent9003ce517ff9906b0288f9f7c02197fd091d4554 (diff)
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Merge pull request #625 from digitall/rate-hack
AUDIO: Add support for sample rates >65kHz.
Diffstat (limited to 'audio/rate.cpp')
-rw-r--r--audio/rate.cpp32
1 files changed, 21 insertions, 11 deletions
diff --git a/audio/rate.cpp b/audio/rate.cpp
index 19d9c8c61e..6264465e19 100644
--- a/audio/rate.cpp
+++ b/audio/rate.cpp
@@ -46,6 +46,16 @@ namespace Audio {
*/
#define INTERMEDIATE_BUFFER_SIZE 512
+/**
+ * The default fractional type in frac.h (with 16 fractional bits) limits
+ * the rate conversion code to 65536Hz audio: we need to able to handle
+ * 96kHz audio, so we use fewer fractional bits in this code.
+ */
+enum {
+ FRAC_BITS_LOW = 15,
+ FRAC_ONE_LOW = (1L << FRAC_BITS_LOW),
+ FRAC_HALF_LOW = (1L << (FRAC_BITS_LOW-1))
+};
/**
* Audio rate converter based on simple resampling. Used when no
@@ -187,18 +197,18 @@ public:
*/
template<bool stereo, bool reverseStereo>
LinearRateConverter<stereo, reverseStereo>::LinearRateConverter(st_rate_t inrate, st_rate_t outrate) {
- if (inrate >= 65536 || outrate >= 65536) {
- error("rate effect can only handle rates < 65536");
+ if (inrate >= 131072 || outrate >= 131072) {
+ error("rate effect can only handle rates < 131072");
}
- opos = FRAC_ONE;
+ opos = FRAC_ONE_LOW;
// Compute the linear interpolation increment.
- // This will overflow if inrate >= 2^16, and underflow if outrate >= 2^16.
+ // This will overflow if inrate >= 2^17, and underflow if outrate >= 2^17.
// Also, if the quotient of the two rate becomes too small / too big, that
// would cause problems, but since we rarely scale from 1 to 65536 Hz or vice
// versa, I think we can live with that limitation ;-).
- opos_inc = (inrate << FRAC_BITS) / outrate;
+ opos_inc = (inrate << FRAC_BITS_LOW) / outrate;
ilast0 = ilast1 = 0;
icur0 = icur1 = 0;
@@ -220,7 +230,7 @@ int LinearRateConverter<stereo, reverseStereo>::flow(AudioStream &input, st_samp
while (obuf < oend) {
// read enough input samples so that opos < 0
- while ((frac_t)FRAC_ONE <= opos) {
+ while ((frac_t)FRAC_ONE_LOW <= opos) {
// Check if we have to refill the buffer
if (inLen == 0) {
inPtr = inBuf;
@@ -235,17 +245,17 @@ int LinearRateConverter<stereo, reverseStereo>::flow(AudioStream &input, st_samp
ilast1 = icur1;
icur1 = *inPtr++;
}
- opos -= FRAC_ONE;
+ opos -= FRAC_ONE_LOW;
}
// Loop as long as the outpos trails behind, and as long as there is
// still space in the output buffer.
- while (opos < (frac_t)FRAC_ONE && obuf < oend) {
+ while (opos < (frac_t)FRAC_ONE_LOW && obuf < oend) {
// interpolate
st_sample_t out0, out1;
- out0 = (st_sample_t)(ilast0 + (((icur0 - ilast0) * opos + FRAC_HALF) >> FRAC_BITS));
+ out0 = (st_sample_t)(ilast0 + (((icur0 - ilast0) * opos + FRAC_HALF_LOW) >> FRAC_BITS_LOW));
out1 = (stereo ?
- (st_sample_t)(ilast1 + (((icur1 - ilast1) * opos + FRAC_HALF) >> FRAC_BITS)) :
+ (st_sample_t)(ilast1 + (((icur1 - ilast1) * opos + FRAC_HALF_LOW) >> FRAC_BITS_LOW)) :
out0);
// output left channel
@@ -333,7 +343,7 @@ public:
template<bool stereo, bool reverseStereo>
RateConverter *makeRateConverter(st_rate_t inrate, st_rate_t outrate) {
if (inrate != outrate) {
- if ((inrate % outrate) == 0) {
+ if ((inrate % outrate) == 0 && (inrate < 65536)) {
return new SimpleRateConverter<stereo, reverseStereo>(inrate, outrate);
} else {
return new LinearRateConverter<stereo, reverseStereo>(inrate, outrate);