Type agnosticism for some other variables that were not sampleCount...
... in some cases, this fixes narrowings.
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@ -335,7 +335,7 @@ void BenchmarkDialog::OnRun( wxCommandEvent & WXUNUSED(event))
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gPrefs->Flush();
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// Rememebr the old blocksize, so that we can restore it later.
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int oldBlockSize = Sequence::GetMaxDiskBlockSize();
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auto oldBlockSize = Sequence::GetMaxDiskBlockSize();
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Sequence::SetMaxDiskBlockSize(blockSize * 1024);
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wxBusyCursor busy;
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@ -194,7 +194,7 @@ static void RemoveDependencies(AudacityProject *project,
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continue;
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// Convert it from an aliased file to an actual file in the project.
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unsigned int len = aliasBlockFile->GetLength();
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auto len = aliasBlockFile->GetLength();
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BlockFilePtr newBlockFile;
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{
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SampleBuffer buffer(len, format);
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@ -150,7 +150,7 @@ void MixAndRender(TrackList *tracks, TrackFactory *trackFactory,
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int maxBlockLen = mixLeft->GetIdealBlockSize();
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auto maxBlockLen = mixLeft->GetIdealBlockSize();
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// If the caller didn't specify a time range, use the whole range in which
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// any input track had clips in it.
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@ -338,9 +338,7 @@ Mixer::Mixer(const WaveTrackConstArray &inputTracks,
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mQueueLen[i] = 0;
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}
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int envLen = mInterleavedBufferSize;
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if (mQueueMaxLen > envLen)
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envLen = mQueueMaxLen;
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const auto envLen = std::max(mQueueMaxLen, mInterleavedBufferSize);
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mEnvValues = new double[envLen];
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}
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@ -557,7 +555,7 @@ sampleCount Mixer::MixSameRate(int *channelFlags, WaveTrackCache &cache,
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sampleCount *pos)
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{
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const WaveTrack *const track = cache.GetTrack();
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int slen = mMaxOut;
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auto slen = mMaxOut;
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int c;
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const double t = *pos / track->GetRate();
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const double trackEndTime = track->GetEndTime();
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@ -1556,11 +1556,10 @@ void Sequence::Blockify(BlockArray &list, sampleCount start, samplePtr buffer, s
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{
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if (len <= 0)
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return;
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const int num = (len + (mMaxSamples - 1)) / mMaxSamples;
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auto num = (len + (mMaxSamples - 1)) / mMaxSamples;
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list.reserve(list.size() + num);
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for (int i = 0; i < num; i++) {
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for (decltype(num) i = 0; i < num; i++) {
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SeqBlock b;
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const auto offset = i * len / num;
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@ -608,7 +608,7 @@ bool WaveClip::GetWaveDisplay(WaveDisplay &display, double t0,
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/* handle values in the append buffer */
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int numSamples = mSequence->GetNumSamples();
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auto numSamples = mSequence->GetNumSamples();
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int a;
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// Not all of the required columns might be in the sequence.
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@ -2106,7 +2106,7 @@ void WaveTrack::GetEnvelopeValues(double *buffer, size_t bufferLen,
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// be set twice. Unfortunately, there is no easy way around this since the clips are not
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// stored in increasing time order. If they were, we could just track the time as the
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// buffer is filled.
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for (int i = 0; i < bufferLen; i++)
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for (decltype(bufferLen) i = 0; i < bufferLen; i++)
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{
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buffer[i] = 1.0;
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}
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@ -98,15 +98,15 @@ bool CompareAudioCommand::Apply(CommandExecutionContext context)
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double errorThreshold = GetDouble(wxT("Threshold"));
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// Initialize buffers for track data to be analyzed
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int buffSize = min(mTrack0->GetMaxBlockSize(), mTrack1->GetMaxBlockSize());
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auto buffSize = std::min(mTrack0->GetMaxBlockSize(), mTrack1->GetMaxBlockSize());
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float *buff0 = new float[buffSize];
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float *buff1 = new float[buffSize];
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// Compare tracks block by block
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long s0 = mTrack0->TimeToLongSamples(mT0);
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long s1 = mTrack0->TimeToLongSamples(mT1);
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long position = s0;
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long length = s1 - s0;
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auto s0 = mTrack0->TimeToLongSamples(mT0);
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auto s1 = mTrack0->TimeToLongSamples(mT1);
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auto position = s0;
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auto length = s1 - s0;
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while (position < s1)
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{
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// Get a block of data into the buffers
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@ -517,12 +517,12 @@ bool EffectAutoDuck::ApplyDuckFade(int trackNumber, WaveTrack* t,
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float *buf = new float[kBufSize];
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auto pos = start;
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int fadeDownSamples = t->TimeToLongSamples(
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auto fadeDownSamples = t->TimeToLongSamples(
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mOuterFadeDownLen + mInnerFadeDownLen);
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if (fadeDownSamples < 1)
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fadeDownSamples = 1;
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int fadeUpSamples = t->TimeToLongSamples(
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auto fadeUpSamples = t->TimeToLongSamples(
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mOuterFadeUpLen + mInnerFadeUpLen);
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if (fadeUpSamples < 1)
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fadeUpSamples = 1;
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@ -361,7 +361,7 @@ sampleCount EffectBassTreble::InstanceProcess(EffectBassTrebleState & data,
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data.a0Treble, data.a1Treble, data.a2Treble,
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data.b0Treble, data.b1Treble, data.b2Treble);
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for (sampleCount i = 0; i < blockLen; i++) {
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for (decltype(blockLen) i = 0; i < blockLen; i++) {
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obuf[i] = DoFilter(data, ibuf[i]) * data.gain;
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}
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@ -524,7 +524,7 @@ sampleCount EffectDistortion::InstanceProcess(EffectDistortionState& data, float
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data.param1 = mParams.mParam1;
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data.repeats = mParams.mRepeats;
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for (sampleCount i = 0; i < blockLen; i++) {
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for (decltype(blockLen) i = 0; i < blockLen; i++) {
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if (update && ((data.skipcount++) % skipsamples == 0)) {
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MakeTable();
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}
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@ -1617,7 +1617,7 @@ bool Effect::ProcessTrack(int count,
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auto cnt = mBlockSize - curBlockSize;
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for (int i = 0; i < mNumChannels; i++)
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{
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for (int j = 0 ; j < cnt; j++)
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for (decltype(cnt) j = 0 ; j < cnt; j++)
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{
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mInBufPos[i][j + curBlockSize] = 0.0;
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}
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@ -420,7 +420,7 @@ bool EffectEqualization48x::DeltaTrack(WaveTrack * t, WaveTrack * t2, sampleCoun
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auto curretLength = (trackBlockSize > len) ? len : trackBlockSize;
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t->Get((samplePtr)buffer1, floatSample, currentSample, curretLength);
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t2->Get((samplePtr)buffer2, floatSample, currentSample, curretLength);
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for(int i=0;i<curretLength;i++)
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for(decltype(curretLength) i=0;i<curretLength;i++)
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buffer1[i]-=buffer2[i];
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output->Append((samplePtr)buffer1, floatSample, curretLength);
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currentSample+=curretLength;
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@ -164,7 +164,7 @@ bool EffectFindClipping::ProcessOne(LabelTrack * lt,
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float *ptr = buffer;
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decltype(len) s = 0, startrun = 0, stoprun = 0, samps = 0;
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size_t block = 0;
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decltype(blockSize) block = 0;
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double startTime = -1.0;
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while (s < len) {
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@ -355,7 +355,7 @@ sampleCount EffectWahwah::InstanceProcess(EffectWahwahState & data, float **inBl
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data.phase = mPhase * M_PI / 180.0;
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data.outgain = DB_TO_LINEAR(mOutGain);
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for (int i = 0; i < blockLen; i++)
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for (decltype(blockLen) i = 0; i < blockLen; i++)
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{
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in = (double) ibuf[i];
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@ -311,7 +311,7 @@ int ExportFLAC::Export(AudacityProject *project,
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numChannels, SAMPLES_PER_RUN, false,
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rate, format, true, mixerSpec);
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int i, j;
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int i;
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FLAC__int32 **tmpsmplbuf = new FLAC__int32*[numChannels];
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for (i = 0; i < numChannels; i++) {
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tmpsmplbuf[i] = (FLAC__int32 *) calloc(SAMPLES_PER_RUN, sizeof(FLAC__int32));
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@ -332,12 +332,12 @@ int ExportFLAC::Export(AudacityProject *project,
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for (i = 0; i < numChannels; i++) {
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samplePtr mixed = mixer->GetBuffer(i);
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if (format == int24Sample) {
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for (j = 0; j < samplesThisRun; j++) {
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for (decltype(samplesThisRun) j = 0; j < samplesThisRun; j++) {
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tmpsmplbuf[i][j] = ((int *)mixed)[j];
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}
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}
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else {
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for (j = 0; j < samplesThisRun; j++) {
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for (decltype(samplesThisRun) j = 0; j < samplesThisRun; j++) {
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tmpsmplbuf[i][j] = ((short *)mixed)[j];
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}
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}
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@ -448,7 +448,7 @@ int PCMImportFileHandle::Import(TrackFactory *trackFactory,
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SampleBuffer buffer(maxBlock, mFormat);
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unsigned long framescompleted = 0;
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decltype(fileTotalFrames) framescompleted = 0;
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long block;
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do {
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