Another std::min, more auto & decltype, remove more casts

src/Audacity.h

@@ -162,7 +162,7 @@ void QuitAudacity();
 #endif
 
 // These macros are used widely, so declared here.
-#define QUANTIZED_TIME(time, rate) (((double)((sampleCount)floor(((double)(time) * (rate)) + 0.5))) / (rate))
+#define QUANTIZED_TIME(time, rate) (floor(((double)(time) * (rate)) + 0.5) / (rate))
 // dB - linear amplitude convesions
 #define DB_TO_LINEAR(x) (pow(10.0, (x) / 20.0))
 #define LINEAR_TO_DB(x) (20.0 * log10(x))

src/Sequence.cpp

@@ -1488,7 +1488,7 @@ bool Sequence::Append(samplePtr buffer, sampleFormat format,
        (length =
         (pLastBlock = &mBlock.back())->f->GetLength()) < mMinSamples) {
       SeqBlock &lastBlock = *pLastBlock;
-      const sampleCount addLen = std::min(mMaxSamples - length, len);
+      const auto addLen = std::min(mMaxSamples - length, len);
 
       Read(buffer2.ptr(), mSampleFormat, lastBlock, 0, length);
 

src/effects/Effect.cpp

@@ -2441,7 +2441,7 @@ sampleCount Effect::RealtimeProcess(int group,
       len = 0;
       for (decltype(numSamples) block = 0; block < numSamples; block += mBlockSize)
       {
-         auto cnt = (block + mBlockSize > numSamples ? numSamples - block : mBlockSize);
+         auto cnt = std::min(numSamples - block, mBlockSize);
          len += RealtimeProcess(processor, clientIn, clientOut, cnt);
 
          for (int i = 0 ; i < mNumAudioIn; i++)

src/ondemand/ODDecodeFFmpegTask.cpp

@@ -387,7 +387,7 @@ int ODFFmpegDecoder::Decode(SampleBuffer & data, sampleFormat & format, sampleCo
             //if we've skipped over some samples, fill the gap with silence.  This could happen often in the beginning of the file.
          if(actualDecodeStart>start && firstpass) {
             // find the number of samples for the leading silence
-            int amt = actualDecodeStart - start;
+            auto amt = actualDecodeStart - start;
             auto cache = make_movable<FFMpegDecodeCache>();
 
             //printf("skipping/zeroing %i samples. - now:%llu (%f), last:%llu, lastlen:%llu, start %llu, len %llu\n",amt,actualDecodeStart, actualDecodeStartdouble, mCurrentPos, mCurrentLen, start, len);
@@ -503,21 +503,20 @@ int ODFFmpegDecoder::FillDataFromCache(samplePtr & data, sampleFormat outFormat,
          if(start<mDecodeCache[i]->start && start+len  > mDecodeCache[i]->start+mDecodeCache[i]->len)
             continue;
 
-         int samplesHit;
-         int hitStartInCache;
-         int hitStartInRequest;
          int nChannels = mDecodeCache[i]->numChannels;
-         samplesHit = FFMIN(start+len,mDecodeCache[i]->start+mDecodeCache[i]->len)
+         auto samplesHit = (
-                        - FFMAX(mDecodeCache[i]->start,start);
+            FFMIN(start+len,mDecodeCache[i]->start+mDecodeCache[i]->len)
+               - FFMAX(mDecodeCache[i]->start,start)
+         );
          //find the start of the hit relative to the cache buffer start.
-         hitStartInCache   = FFMAX(0,start-mDecodeCache[i]->start);
+         const auto hitStartInCache   = FFMAX(0,start-mDecodeCache[i]->start);
          //we also need to find out which end was hit - if it is the tail only we need to update from a later index.
-         hitStartInRequest = start <mDecodeCache[i]->start?len - samplesHit: 0;
+         const auto hitStartInRequest = start < mDecodeCache[i]->start
-         sampleCount outIndex,inIndex;
+            ? len - samplesHit : 0;
-         for(int j=0;j<samplesHit;j++)
+         for(decltype(samplesHit) j = 0; j < samplesHit; j++)
          {
-            outIndex = hitStartInRequest + j;
+            const auto outIndex = hitStartInRequest + j;
-            inIndex = (hitStartInCache + j) * nChannels + channel;
+            const auto inIndex = (hitStartInCache + j) * nChannels + channel;
             switch (mDecodeCache[i]->samplefmt)
             {
                case AV_SAMPLE_FMT_U8:

src/widgets/NumericTextCtrl.cpp

@@ -767,8 +767,7 @@ void NumericConverter::ValueToControls(double rawValue, bool nearest /* = true *
    //rawValue = 4.9995f; Only for testing!
    if (mType == TIME)
       rawValue =
-      (double)((sampleCount)floor(rawValue * mSampleRate +
+         floor(rawValue * mSampleRate + (nearest ? 0.5f : 0.0f))
-                                  (nearest ? 0.5f : 0.0f)))
             / mSampleRate; // put on a sample
    double theValue =
       rawValue * mScalingFactor + .000001; // what's this .000001 for? // well, no log of 0