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author | Max Horn | 2003-08-05 00:29:37 +0000 |
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committer | Max Horn | 2003-08-05 00:29:37 +0000 |
commit | e1ff91ea7d64a656767770a1c52d6d5eaebe398f (patch) | |
tree | 3089875abc43bb53fe93b38b1526e99532b24b17 | |
parent | 8850f858bc8ceb86b84f75602241aad886e8173d (diff) | |
download | scummvm-rg350-e1ff91ea7d64a656767770a1c52d6d5eaebe398f.tar.gz scummvm-rg350-e1ff91ea7d64a656767770a1c52d6d5eaebe398f.tar.bz2 scummvm-rg350-e1ff91ea7d64a656767770a1c52d6d5eaebe398f.zip |
get rid of calls to the obsolete AudioInputStream::size method
svn-id: r9471
-rw-r--r-- | sound/rate.h | 12 | ||||
-rw-r--r-- | sound/resample.cpp | 46 |
2 files changed, 27 insertions, 31 deletions
diff --git a/sound/rate.h b/sound/rate.h index 694dbe0b16..9b8978b423 100644 --- a/sound/rate.h +++ b/sound/rate.h @@ -72,18 +72,6 @@ public: 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); -}; -*/ - RateConverter *makeRateConverter(st_rate_t inrate, st_rate_t outrate, bool stereo, bool reverseStereo = false); #endif diff --git a/sound/resample.cpp b/sound/resample.cpp index 8796989967..424508e88e 100644 --- a/sound/resample.cpp +++ b/sound/resample.cpp @@ -313,16 +313,15 @@ with them alone st_fail("resample: Can not handle this sample rate change. Nx not positive: %d", Nx); return (ST_EOF); } - - // Nx is the number of bytes we'd like to read, but of course that is limited - // by the number of bytes actually available... - if (Nx > (long)input.size()) - Nx = (long)input.size(); - fprintf(stderr,"Nx %d\n",Nx); - // Read in Nx bytes - for (i = r->Xread; i < Nx + r->Xread ; i++) + // Read in up to Nx bytes + for (i = r->Xread; i < Nx + r->Xread && !input.eos(); i++) { r->X[i] = (Float)input.read(); + } + Nx = i - r->Xread; // Compute how many samples we actually read + + fprintf(stderr,"Nx %d\n",Nx); + last = Nx + r->Xread; // 'last' is the idx after the last valid byte in X (i.e. number of bytes are in buffer X right now) @@ -409,20 +408,19 @@ printf("osamp = %ld, Nout = %ld\n", obufSize, Nout); */ int st_resample_drain(eff_t effp, st_sample_t *obuf, st_size_t *osamp, st_volume_t vol) { resample_t r = (resample_t) effp->priv; - long isamp_res, osamp_res; + long osamp_res; st_sample_t *Obuf; int rc; /*fprintf(stderr,"Xoff %d, Xt %d <--- DRAIN\n",r->Xoff, r->Xt);*/ /* stuff end with Xoff zeros */ - isamp_res = r->Xoff; + ZeroInputStream zero(r->Xoff); osamp_res = *osamp; Obuf = obuf; - while (isamp_res > 0 && osamp_res > 0) { + while (!zero.eos() && osamp_res > 0) { st_sample_t Osamp; Osamp = osamp_res; - ZeroInputStream zero(isamp_res); rc = st_resample_flow(effp, zero, Obuf, (st_size_t *) & Osamp, vol); if (rc) return rc; @@ -430,12 +428,11 @@ int st_resample_drain(eff_t effp, st_sample_t *obuf, st_size_t *osamp, st_volume isamp_res,osamp_res,Isamp,Osamp);*/ Obuf += Osamp; osamp_res -= Osamp; - isamp_res = zero.size(); } *osamp -= osamp_res; fprintf(stderr,"DRAIN osamp %d\n", *osamp); - if (isamp_res) - st_warn("drain overran obuf by %d\n", isamp_res); + if (!zero.eos()) + st_warn("drain overran obuf\n"); fflush(stderr); return (ST_SUCCESS); } @@ -738,6 +735,17 @@ static void LpFilter(double *c, long N, double frq, double Beta, long Num) { #pragma mark - +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); +}; + + ResampleRateConverter::ResampleRateConverter(st_rate_t inrate, st_rate_t outrate, int quality) { // FIXME: quality is for now a nasty hack. // Valid values are 0,1,2,3 (everything else is treated like 0 for now) @@ -755,11 +763,11 @@ ResampleRateConverter::~ResampleRateConverter() { st_resample_stop(&effp); } -int ResampleRateConverter::flow(AudioInputStream &input, st_sample_t *obuf, st_size_t *osamp, st_volume_t vol) { - return st_resample_flow(&effp, input, obuf, osamp, vol); +int ResampleRateConverter::flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol) { + return st_resample_flow(&effp, input, obuf, &osamp, vol); } -int ResampleRateConverter::drain(st_sample_t *obuf, st_size_t *osamp, st_volume_t vol) { - return st_resample_drain(&effp, obuf, osamp, vol); +int ResampleRateConverter::drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol) { + return st_resample_drain(&effp, obuf, &osamp, vol); } |