mirror of
https://github.com/cookiengineer/audacity
synced 2025-05-05 14:18:53 +02:00
Fix Windows build differently
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parent
781de82d02
commit
b5e410d987
@ -448,7 +448,7 @@ sampleCount Mixer::MixVariableRates(int *channelFlags, WaveTrackCache &cache,
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memmove(queue, &queue[*queueStart], (*queueLen) * sampleSize);
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*queueStart = 0;
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const auto getLen = limitSampleBufferSize(
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auto getLen = limitSampleBufferSize(
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mQueueMaxLen - *queueLen,
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backwards ? *pos - endPos : endPos - *pos
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);
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@ -476,7 +476,7 @@ sampleCount Mixer::MixVariableRates(int *channelFlags, WaveTrackCache &cache,
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*pos += getLen;
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}
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for (auto i = 0; i < getLen; i++) {
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for (decltype(getLen) i = 0; i < getLen; i++) {
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queue[(*queueLen) + i] *= mEnvValues[i];
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}
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@ -848,12 +848,12 @@ double VoiceKey::TestEnergy (WaveTrack & t, sampleCount start, sampleCount len)
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while(len > 0)
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{
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//Figure out how much to grab
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const auto block = limitSampleBufferSize ( t.GetBestBlockSize(s), len );
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auto block = limitSampleBufferSize ( t.GetBestBlockSize(s), len );
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t.Get((samplePtr)buffer,floatSample, s,block); //grab the block;
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//Now, go through the block and calculate energy
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for(auto i = 0; i< block; i++)
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for(decltype(block) i = 0; i< block; i++)
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{
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sum += buffer[i]*buffer[i];
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}
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@ -891,7 +891,7 @@ double VoiceKey::TestSignChanges(WaveTrack & t, sampleCount start, sampleCount l
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while(len > 0) {
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//Figure out how much to grab
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const auto block = limitSampleBufferSize ( t.GetBestBlockSize(s), len );
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auto block = limitSampleBufferSize ( t.GetBestBlockSize(s), len );
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t.Get((samplePtr)buffer,floatSample, s,block); //grab the block;
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@ -903,7 +903,7 @@ double VoiceKey::TestSignChanges(WaveTrack & t, sampleCount start, sampleCount l
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//Now, go through the block and calculate zero crossings
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for(auto i = 0; i< block; i++)
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for(decltype(block) i = 0; i< block; i++)
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{
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if( sgn(buffer[i]) != currentsign)
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{
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@ -948,7 +948,7 @@ double VoiceKey::TestDirectionChanges(WaveTrack & t, sampleCount start, sampleCo
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while(len > 0) {
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//Figure out how much to grab
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const auto block = limitSampleBufferSize ( t.GetBestBlockSize(s), len );
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auto block = limitSampleBufferSize ( t.GetBestBlockSize(s), len );
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t.Get((samplePtr)buffer,floatSample, s,block); //grab the block;
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@ -960,7 +960,7 @@ double VoiceKey::TestDirectionChanges(WaveTrack & t, sampleCount start, sampleCo
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//Now, go through the block and calculate zero crossings
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for(auto i = 0; i< block; i++){
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for(decltype(block) i = 0; i< block; i++){
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if( sgn(buffer[i]-lastval) != lastdirection) {
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directionchanges++;
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@ -1459,11 +1459,11 @@ bool WaveTrack::Disjoin(double t0, double t1)
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sampleCount len = ( end - start );
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for( sampleCount done = 0; done < len; done += maxAtOnce )
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{
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const auto numSamples = limitSampleBufferSize( maxAtOnce, len - done );
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auto numSamples = limitSampleBufferSize( maxAtOnce, len - done );
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clip->GetSamples( ( samplePtr )buffer, floatSample, start + done,
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numSamples );
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for( auto i = 0; i < numSamples; i++ )
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for( decltype(numSamples) i = 0; i < numSamples; i++ )
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{
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sampleCount curSamplePos = start + done + i;
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@ -110,13 +110,13 @@ bool CompareAudioCommand::Apply(CommandExecutionContext context)
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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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const auto block = limitSampleBufferSize(
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auto block = limitSampleBufferSize(
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mTrack0->GetBestBlockSize(position), s1 - position
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);
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mTrack0->Get((samplePtr)buff0, floatSample, position, block);
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mTrack1->Get((samplePtr)buff1, floatSample, position, block);
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for (auto buffPos = 0; buffPos < block; ++buffPos)
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for (decltype(block) buffPos = 0; buffPos < block; ++buffPos)
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{
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if (CompareSample(buff0[buffPos], buff1[buffPos]) > errorThreshold)
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{
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@ -229,13 +229,13 @@ bool EffectReverse::ProcessOneClip(int count, WaveTrack *track,
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sampleCount originalLen = (sampleCount)originalEnd-originalStart;
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while (len > 1) {
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const auto block =
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auto block =
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limitSampleBufferSize( track->GetBestBlockSize(first), len / 2 );
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second = first + (len - block);
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track->Get((samplePtr)buffer1, floatSample, first, block);
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track->Get((samplePtr)buffer2, floatSample, second, block);
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for (auto i = 0; i < block; i++) {
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for (decltype(block) i = 0; i < block; i++) {
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tmp = buffer1[i];
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buffer1[i] = buffer2[block-i-1];
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buffer2[block-i-1] = tmp;
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@ -98,7 +98,7 @@ long resampleCB(void *cb_data, SBSMSFrame *data)
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{
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ResampleBuf *r = (ResampleBuf*) cb_data;
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const auto blockSize = limitSampleBufferSize(
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auto blockSize = limitSampleBufferSize(
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r->leftTrack->GetBestBlockSize(r->offset),
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r->end - r->offset
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);
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@ -108,7 +108,7 @@ long resampleCB(void *cb_data, SBSMSFrame *data)
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r->rightTrack->Get((samplePtr)(r->rightBuffer), floatSample, r->offset, blockSize);
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// convert to sbsms audio format
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for(auto i=0; i<blockSize; i++) {
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for(decltype(blockSize) i=0; i<blockSize; i++) {
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r->buf[i][0] = r->leftBuffer[i];
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r->buf[i][1] = r->rightBuffer[i];
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}
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@ -281,7 +281,7 @@ bool EffectSoundTouch::ProcessStereo(WaveTrack* leftTrack, WaveTrack* rightTrack
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while (sourceSampleCount < end) {
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//Get a block of samples (smaller than the size of the buffer)
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//Adjust the block size if it is the final block in the track
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const auto blockSize = limitSampleBufferSize(
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auto blockSize = limitSampleBufferSize(
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leftTrack->GetBestBlockSize(sourceSampleCount),
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end - sourceSampleCount
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);
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@ -291,7 +291,7 @@ bool EffectSoundTouch::ProcessStereo(WaveTrack* leftTrack, WaveTrack* rightTrack
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rightTrack->Get((samplePtr)(rightBuffer), floatSample, sourceSampleCount, blockSize);
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// Interleave into soundTouchBuffer.
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for (auto index = 0; index < blockSize; index++) {
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for (decltype(blockSize) index = 0; index < blockSize; index++) {
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soundTouchBuffer[index*2] = leftBuffer[index];
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soundTouchBuffer[(index*2)+1] = rightBuffer[index];
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}
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@ -144,8 +144,8 @@ bool EffectStereoToMono::ProcessOne(int count)
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while (index < mEnd) {
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bResult &= mLeftTrack->Get((samplePtr)leftBuffer, floatSample, index, idealBlockLen);
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bResult &= mRightTrack->Get((samplePtr)rightBuffer, floatSample, index, idealBlockLen);
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const auto limit = limitSampleBufferSize( idealBlockLen, mEnd - index );
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for (auto i = 0; i < limit; ++i) {
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auto limit = limitSampleBufferSize( idealBlockLen, mEnd - index );
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for (decltype(limit) i = 0; i < limit; ++i) {
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index++;
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curLeftFrame = leftBuffer[i];
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curRightFrame = rightBuffer[i];
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@ -624,13 +624,13 @@ bool EffectTruncSilence::Analyze(RegionList& silenceList,
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// End of optimization
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// Limit size of current block if we've reached the end
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const auto count = limitSampleBufferSize( blockLen, end - *index );
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auto count = limitSampleBufferSize( blockLen, end - *index );
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// Fill buffer
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wt->Get((samplePtr)(buffer), floatSample, *index, count);
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// Look for silenceList in current block
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for (auto i = 0; i < count; ++i) {
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for (decltype(count) i = 0; i < count; ++i) {
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if (inputLength && ((outLength >= previewLen) || (outLength > wt->TimeToLongSamples(*minInputLength)))) {
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*inputLength = wt->LongSamplesToTime(*index + i) - wt->LongSamplesToTime(start);
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break;
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