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author | Eugene Sandulenko | 2015-11-09 16:39:17 +0100 |
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committer | Eugene Sandulenko | 2015-11-09 16:39:17 +0100 |
commit | 82c98e98033eafa2ed04febe2607f09636e7e6a5 (patch) | |
tree | 450e3e543c839c6b89aedf1a1e329fd9072d51ee /audio/rate.cpp | |
parent | 30b6682130e5aefe1e019eb37c0cd25b5831d225 (diff) | |
parent | 9003ce517ff9906b0288f9f7c02197fd091d4554 (diff) | |
download | scummvm-rg350-82c98e98033eafa2ed04febe2607f09636e7e6a5.tar.gz scummvm-rg350-82c98e98033eafa2ed04febe2607f09636e7e6a5.tar.bz2 scummvm-rg350-82c98e98033eafa2ed04febe2607f09636e7e6a5.zip |
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.cpp | 32 |
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); |