Revert of Convert RELEASE_ASSERT()/ASSERT(...) to CHECK()/DCHECK_op(...) in...

Revert of Convert RELEASE_ASSERT()/ASSERT(...) to CHECK()/DCHECK_op(...) in platform/audio (patchset #3 id:40001 of https://codereview.chromium.org/2803073002/ )

Reason for revert:
Unexpected compilation error from buildbot Android Builder.

Original issue's description:
> Convert RELEASE_ASSERT()/ASSERT(...) to CHECK()/DCHECK_op(...) in platform/audio
>
> This CL is the second part of the conversion. It manually replaces:
>
>  - RELEASE_ASSERT() with CHECK().
>  - ASSERT(...args) with one or more DCHECK_op(arg).
>
> With this CL the following script returns 0 result.
>
> ```
> cd ${CHROME_SRC}/third_party/WebKit/Source/platform/audio
> grep -rnw . -e 'ASSERT('
> ```
>
> BUG=707655
>
> Review-Url: https://codereview.chromium.org/2803073002
> Cr-Commit-Position: refs/heads/master@{#462904}
> Committed: https://chromium.googlesource.com/chromium/src/+/f90cae07269c724f98ca3c196ea37ce7f6a801d5

TBR=rtoy@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=707655

Review-Url: https://codereview.chromium.org/2807593003
Cr-Commit-Position: refs/heads/master@{#462908}

third_party/WebKit/Source/platform/audio/AudioArray.h

@@ -58,7 +58,7 @@ class AudioArray {
     // Although n is a size_t, its true limit is max unsigned because we use
     // unsigned in zeroRange() and copyToRange(). Also check for integer
     // overflow.
-    CHECK_LE(n, std::numeric_limits<unsigned>::max() / sizeof(T));
+    RELEASE_ASSERT(n <= std::numeric_limits<unsigned>::max() / sizeof(T));
 
     unsigned initialSize = sizeof(T) * n;
 
@@ -79,12 +79,12 @@ class AudioArray {
       static size_t extraAllocationBytes = 0;
 
       // Again, check for integer overflow.
-      CHECK_GE(initialSize + extraAllocationBytes, initialSize);
+      RELEASE_ASSERT(initialSize + extraAllocationBytes >= initialSize);
 
       T* allocation = static_cast<T*>(WTF::Partitions::fastMalloc(
           initialSize + extraAllocationBytes,
           WTF_HEAP_PROFILER_TYPE_NAME(AudioArray<T>)));
-      CHECK(allocation);
+      RELEASE_ASSERT(allocation);
 
       T* alignedData = alignedAddress(allocation, alignment);
 

third_party/WebKit/Source/platform/audio/AudioBus.cpp

@@ -49,7 +49,7 @@ const unsigned MaxBusChannels = 32;
 PassRefPtr<AudioBus> AudioBus::create(unsigned numberOfChannels,
                                       size_t length,
                                       bool allocate) {
-  DCHECK_LE(numberOfChannels, MaxBusChannels);
+  ASSERT(numberOfChannels <= MaxBusChannels);
   if (numberOfChannels > MaxBusChannels)
     return nullptr;
 
@@ -81,7 +81,7 @@ void AudioBus::setChannelMemory(unsigned channelIndex,
 }
 
 void AudioBus::resizeSmaller(size_t newLength) {
-  DCHECK_LE(newLength, m_length);
+  ASSERT(newLength <= m_length);
   if (newLength <= m_length)
     m_length = newLength;
 
@@ -479,7 +479,7 @@ void AudioBus::copyWithGainFrom(const AudioBus& sourceBus,
   }
 
   unsigned numberOfChannels = this->numberOfChannels();
-  DCHECK_LE(numberOfChannels, MaxBusChannels);
+  ASSERT(numberOfChannels <= MaxBusChannels);
   if (numberOfChannels > MaxBusChannels)
     return;
 
@@ -615,8 +615,7 @@ PassRefPtr<AudioBus> AudioBus::createBySampleRateConverting(
     bool mixToMono,
     double newSampleRate) {
   // sourceBus's sample-rate must be known.
-  DCHECK(sourceBus);
-  DCHECK(sourceBus->sampleRate());
+  ASSERT(sourceBus && sourceBus->sampleRate());
   if (!sourceBus || !sourceBus->sampleRate())
     return nullptr;
 

third_party/WebKit/Source/platform/audio/AudioChannel.cpp

@@ -36,7 +36,7 @@ namespace blink {
 using namespace VectorMath;
 
 void AudioChannel::resizeSmaller(size_t newLength) {
-  DCHECK_LE(newLength, m_length);
+  ASSERT(newLength <= m_length);
   if (newLength <= m_length)
     m_length = newLength;
 }

third_party/WebKit/Source/platform/audio/AudioDSPKernelProcessor.cpp

@@ -67,8 +67,7 @@ void AudioDSPKernelProcessor::uninitialize() {
 void AudioDSPKernelProcessor::process(const AudioBus* source,
                                       AudioBus* destination,
                                       size_t framesToProcess) {
-  DCHECK(source);
-  DCHECK(destination);
+  ASSERT(source && destination);
   if (!source || !destination)
     return;
 

third_party/WebKit/Source/platform/audio/AudioDelayDSPKernel.cpp

@@ -45,8 +45,7 @@ AudioDelayDSPKernel::AudioDelayDSPKernel(double maxDelayTime, float sampleRate)
       m_maxDelayTime(maxDelayTime),
       m_writeIndex(0),
       m_firstTime(true) {
-  DCHECK_GT(maxDelayTime, 0.0);
-  DCHECK(!std::isnan(maxDelayTime));
+  ASSERT(maxDelayTime > 0.0 && !std::isnan(maxDelayTime));
   if (maxDelayTime <= 0.0 || std::isnan(maxDelayTime))
     return;
 
@@ -92,8 +91,7 @@ void AudioDelayDSPKernel::process(const float* source,
   if (!bufferLength)
     return;
 
-  DCHECK(source);
-  DCHECK(destination);
+  ASSERT(source && destination);
   if (!source || !destination)
     return;
 

third_party/WebKit/Source/platform/audio/AudioResamplerKernel.cpp

@@ -46,7 +46,7 @@ AudioResamplerKernel::AudioResamplerKernel(AudioResampler* resampler)
 float* AudioResamplerKernel::getSourcePointer(
     size_t framesToProcess,
     size_t* numberOfSourceFramesNeededP) {
-  DCHECK_LE(framesToProcess, MaxFramesToProcess);
+  ASSERT(framesToProcess <= MaxFramesToProcess);
 
   // Calculate the next "virtual" index.  After process() is called,
   // m_virtualReadIndex will equal this value.
@@ -75,7 +75,7 @@ float* AudioResamplerKernel::getSourcePointer(
 }
 
 void AudioResamplerKernel::process(float* destination, size_t framesToProcess) {
-  DCHECK_LE(framesToProcess, MaxFramesToProcess);
+  ASSERT(framesToProcess <= MaxFramesToProcess);
 
   float* source = m_sourceBuffer.data();
 
@@ -92,11 +92,11 @@ void AudioResamplerKernel::process(float* destination, size_t framesToProcess) {
   double virtualReadIndex = m_virtualReadIndex;
 
   // Sanity check source buffer access.
-  DCHECK_GT(framesToProcess, 0u);
-  DCHECK_GE(virtualReadIndex, 0);
-  DCHECK_LT(1 + static_cast<unsigned>(virtualReadIndex +
-                                      (framesToProcess - 1) * rate),
-            m_sourceBuffer.size());
+  ASSERT(framesToProcess > 0);
+  ASSERT(virtualReadIndex >= 0 &&
+         1 + static_cast<unsigned>(virtualReadIndex +
+                                   (framesToProcess - 1) * rate) <
+             m_sourceBuffer.size());
 
   // Do the linear interpolation.
   int n = framesToProcess;

third_party/WebKit/Source/platform/audio/AudioUtilities.cpp

@@ -36,7 +36,7 @@ float decibelsToLinear(float decibels) {
 }
 
 float linearToDecibels(float linear) {
-  DCHECK_GE(linear, 0);
+  ASSERT(linear >= 0);
 
   return 20 * log10f(linear);
 }
@@ -47,7 +47,7 @@ double discreteTimeConstantForSampleRate(double timeConstant,
 }
 
 size_t timeToSampleFrame(double time, double sampleRate) {
-  DCHECK_GE(time, 0);
+  ASSERT(time >= 0);
   double frame = round(time * sampleRate);
 
   // Just return the largest possible size_t value if necessary.

third_party/WebKit/Source/platform/audio/DirectConvolver.cpp

@@ -50,13 +50,13 @@ void DirectConvolver::process(AudioFloatArray* convolutionKernel,
                               const float* sourceP,
                               float* destP,
                               size_t framesToProcess) {
-  DCHECK_EQ(framesToProcess, m_inputBlockSize);
+  ASSERT(framesToProcess == m_inputBlockSize);
   if (framesToProcess != m_inputBlockSize)
     return;
 
   // Only support kernelSize <= m_inputBlockSize
   size_t kernelSize = convolutionKernel->size();
-  DCHECK_LE(kernelSize, m_inputBlockSize);
+  ASSERT(kernelSize <= m_inputBlockSize);
   if (kernelSize > m_inputBlockSize)
     return;
 

third_party/WebKit/Source/platform/audio/DynamicsCompressor.cpp

@@ -51,7 +51,7 @@ DynamicsCompressor::DynamicsCompressor(float sampleRate,
 }
 
 void DynamicsCompressor::setParameterValue(unsigned parameterID, float value) {
-  DCHECK_LT(parameterID, static_cast<unsigned>(ParamLast));
+  ASSERT(parameterID < ParamLast);
   if (parameterID < ParamLast)
     m_parameters[parameterID] = value;
 }
@@ -84,7 +84,7 @@ void DynamicsCompressor::initializeParameters() {
 }
 
 float DynamicsCompressor::parameterValue(unsigned parameterID) {
-  DCHECK_LT(parameterID, static_cast<unsigned>(ParamLast));
+  ASSERT(parameterID < ParamLast);
   return m_parameters[parameterID];
 }
 
@@ -100,8 +100,7 @@ void DynamicsCompressor::process(const AudioBus* sourceBus,
   unsigned numberOfChannels = destinationBus->numberOfChannels();
   unsigned numberOfSourceChannels = sourceBus->numberOfChannels();
 
-  DCHECK_EQ(numberOfChannels, m_numberOfChannels);
-  DCHECK(numberOfSourceChannels);
+  ASSERT(numberOfChannels == m_numberOfChannels && numberOfSourceChannels);
 
   if (numberOfChannels != m_numberOfChannels || !numberOfSourceChannels) {
     destinationBus->zero();

third_party/WebKit/Source/platform/audio/DynamicsCompressorKernel.cpp

@@ -216,7 +216,7 @@ void DynamicsCompressorKernel::process(
     float releaseZone2,
     float releaseZone3,
     float releaseZone4) {
-  DCHECK_EQ(m_preDelayBuffers.size(), numberOfChannels);
+  ASSERT(m_preDelayBuffers.size() == numberOfChannels);
 
   float sampleRate = this->sampleRate();
 

third_party/WebKit/Source/platform/audio/HRTFDatabase.cpp

@@ -76,7 +76,7 @@ HRTFDatabase::HRTFDatabase(float sampleRate)
             static_cast<float>(jj) / static_cast<float>(InterpolationFactor);
         m_elevations[i + jj] = HRTFElevation::createByInterpolatingSlices(
             m_elevations[i].get(), m_elevations[j].get(), x, sampleRate);
-        DCHECK(m_elevations[i + jj].get());
+        ASSERT(m_elevations[i + jj].get());
       }
     }
   }

third_party/WebKit/Source/platform/audio/HRTFDatabaseLoader.cpp

@@ -52,7 +52,7 @@ HRTFDatabaseLoader::createAndLoadAsynchronouslyIfNecessary(float sampleRate) {
 
   RefPtr<HRTFDatabaseLoader> loader = getLoaderMap().at(sampleRate);
   if (loader) {
-    DCHECK_EQ(sampleRate, loader->databaseSampleRate());
+    ASSERT(sampleRate == loader->databaseSampleRate());
     return loader.release();
   }
 

third_party/WebKit/Source/platform/audio/HRTFElevation.cpp

@@ -309,13 +309,11 @@ std::unique_ptr<HRTFElevation> HRTFElevation::createByInterpolatingSlices(
     HRTFElevation* hrtfElevation2,
     float x,
     float sampleRate) {
-  DCHECK(hrtfElevation1);
-  DCHECK(hrtfElevation2);
+  ASSERT(hrtfElevation1 && hrtfElevation2);
   if (!hrtfElevation1 || !hrtfElevation2)
     return nullptr;
 
-  DCHECK_GE(x, 0.0);
-  DCHECK_LT(x, 1.0);
+  ASSERT(x >= 0.0 && x < 1.0);
 
   std::unique_ptr<HRTFKernelList> kernelListL =
       WTF::makeUnique<HRTFKernelList>(NumberOfTotalAzimuths);

third_party/WebKit/Source/platform/audio/HRTFKernel.cpp

@@ -53,8 +53,8 @@ static float extractAverageGroupDelay(AudioChannel* channel,
     return 0;
 
   // Check for power-of-2.
-  DCHECK_EQ(1UL << static_cast<unsigned>(log2(analysisFFTSize)),
-            analysisFFTSize);
+  ASSERT(1UL << static_cast<unsigned>(log2(analysisFFTSize)) ==
+         analysisFFTSize);
 
   FFTFrame estimationFrame(analysisFFTSize);
   estimationFrame.doFFT(impulseP);
@@ -83,7 +83,7 @@ HRTFKernel::HRTFKernel(AudioChannel* channel, size_t fftSize, float sampleRate)
   // Quick fade-out (apply window) at truncation point
   unsigned numberOfFadeOutFrames = static_cast<unsigned>(
       sampleRate / 4410);  // 10 sample-frames @44.1KHz sample-rate
-  DCHECK_LT(numberOfFadeOutFrames, truncatedResponseLength);
+  ASSERT(numberOfFadeOutFrames < truncatedResponseLength);
   if (numberOfFadeOutFrames < truncatedResponseLength) {
     for (unsigned i = truncatedResponseLength - numberOfFadeOutFrames;
          i < truncatedResponseLength; ++i) {
@@ -116,18 +116,16 @@ std::unique_ptr<HRTFKernel> HRTFKernel::createInterpolatedKernel(
     HRTFKernel* kernel1,
     HRTFKernel* kernel2,
     float x) {
-  DCHECK(kernel1);
-  DCHECK(kernel2);
+  ASSERT(kernel1 && kernel2);
   if (!kernel1 || !kernel2)
     return nullptr;
 
-  DCHECK_GE(x, 0.0);
-  DCHECK_LT(x, 1.0);
+  ASSERT(x >= 0.0 && x < 1.0);
   x = clampTo(x, 0.0f, 1.0f);
 
   float sampleRate1 = kernel1->sampleRate();
   float sampleRate2 = kernel2->sampleRate();
-  DCHECK_EQ(sampleRate1, sampleRate2);
+  ASSERT(sampleRate1 == sampleRate2);
   if (sampleRate1 != sampleRate2)
     return nullptr;
 

third_party/WebKit/Source/platform/audio/HRTFPanner.cpp

@@ -210,7 +210,7 @@ void HRTFPanner::pan(double desiredAzimuth,
 
   // This algorithm currently requires that we process in power-of-two size
   // chunks at least AudioUtilities::kRenderQuantumFrames.
-  DCHECK_EQ(1UL << static_cast<int>(log2(framesToProcess)), framesToProcess);
+  ASSERT(1UL << static_cast<int>(log2(framesToProcess)) == framesToProcess);
   DCHECK_GE(framesToProcess, AudioUtilities::kRenderQuantumFrames);
 
   const unsigned framesPerSegment = AudioUtilities::kRenderQuantumFrames;
@@ -240,10 +240,10 @@ void HRTFPanner::pan(double desiredAzimuth,
       return;
     }
 
-    DCHECK_LT(frameDelayL1 / sampleRate(), MaxDelayTimeSeconds);
-    DCHECK_LT(frameDelayR1 / sampleRate(), MaxDelayTimeSeconds);
-    DCHECK_LT(frameDelayL2 / sampleRate(), MaxDelayTimeSeconds);
-    DCHECK_LT(frameDelayR2 / sampleRate(), MaxDelayTimeSeconds);
+    ASSERT(frameDelayL1 / sampleRate() < MaxDelayTimeSeconds &&
+           frameDelayR1 / sampleRate() < MaxDelayTimeSeconds);
+    ASSERT(frameDelayL2 / sampleRate() < MaxDelayTimeSeconds &&
+           frameDelayR2 / sampleRate() < MaxDelayTimeSeconds);
 
     // Crossfade inter-aural delays based on transitions.
     double frameDelayL =

third_party/WebKit/Source/platform/audio/IIRFilter.cpp

@@ -65,7 +65,7 @@ void IIRFilter::process(const float* sourceP,
 
   // Sanity check to see if the feedback coefficients have been scaled
   // appropriately. It must be EXACTLY 1!
-  DCHECK_EQ(feedback[0], 1);
+  ASSERT(feedback[0] == 1);
 
   int feedbackLength = m_feedback->size();
   int feedforwardLength = m_feedforward->size();

third_party/WebKit/Source/platform/audio/MultiChannelResampler.cpp

@@ -74,7 +74,7 @@ class ChannelProvider final : public AudioSourceProvider {
       return;
 
     // Copy the channel data from what we received from m_multiChannelProvider.
-    DCHECK_LE(m_currentChannel, m_numberOfChannels);
+    ASSERT(m_currentChannel <= m_numberOfChannels);
     if (m_currentChannel < m_numberOfChannels) {
       memcpy(bus->channel(0)->mutableData(),
              m_multiChannelBus->channel(m_currentChannel)->data(),

third_party/WebKit/Source/platform/audio/ReverbConvolverStage.cpp

@@ -61,7 +61,7 @@ ReverbConvolverStage::ReverbConvolverStage(
     m_fftConvolver = WTF::makeUnique<FFTConvolver>(fftSize);
   } else {
     DCHECK(!stageOffset);
-    DCHECK_LE(stageLength, fftSize / 2);
+    ASSERT(stageLength <= fftSize / 2);
 
     m_directKernel = WTF::wrapUnique(new AudioFloatArray(fftSize / 2));
     m_directKernel->copyToRange(impulseResponse, 0, stageLength);
@@ -77,7 +77,7 @@ ReverbConvolverStage::ReverbConvolverStage(
   // this out...
   size_t halfSize = fftSize / 2;
   if (!m_directMode) {
-    DCHECK_GE(totalDelay, halfSize);
+    ASSERT(totalDelay >= halfSize);
     if (totalDelay >= halfSize)
       totalDelay -= halfSize;
   }
@@ -177,7 +177,7 @@ void ReverbConvolverStage::process(const float* source,
     memcpy(preDelayedDestination, source, sizeof(float) * framesToProcess);
     m_preReadWriteIndex += framesToProcess;
 
-    DCHECK_LE(m_preReadWriteIndex, m_preDelayLength);
+    ASSERT(m_preReadWriteIndex <= m_preDelayLength);
     if (m_preReadWriteIndex >= m_preDelayLength)
       m_preReadWriteIndex = 0;
   }

third_party/WebKit/Source/platform/audio/ReverbInputBuffer.cpp

@@ -44,7 +44,7 @@ void ReverbInputBuffer::write(const float* sourceP, size_t numberOfFrames) {
          sizeof(float) * numberOfFrames);
 
   m_writeIndex += numberOfFrames;
-  DCHECK_LE(m_writeIndex, bufferLength);
+  ASSERT(m_writeIndex <= bufferLength);
 
   if (m_writeIndex >= bufferLength)
     m_writeIndex = 0;

third_party/WebKit/Source/platform/audio/SincResampler.cpp

@@ -156,8 +156,7 @@ class BufferSourceProvider final : public AudioSourceProvider {
 
   // Consumes samples from the in-memory buffer.
   void provideInput(AudioBus* bus, size_t framesToProcess) override {
-    DCHECK(m_source);
-    DCHECK(bus);
+    ASSERT(m_source && bus);
     if (!m_source || !bus)
       return;
 

third_party/WebKit/Source/platform/audio/android/FFTFrameOpenMAXDLAndroid.cpp

@@ -49,7 +49,7 @@ FFTFrame::FFTFrame(unsigned fftSize)
       m_inverseContext(nullptr),
       m_complexData(fftSize) {
   // We only allow power of two.
-  DCHECK_EQ(1UL << m_log2FFTSize, m_FFTSize);
+  ASSERT(1UL << m_log2FFTSize == m_FFTSize);
 
   m_forwardContext = contextForSize(m_log2FFTSize);
   m_inverseContext = contextForSize(m_log2FFTSize);
@@ -143,7 +143,7 @@ void FFTFrame::doInverseFFT(float* data) {
 
 OMXFFTSpec_R_F32* FFTFrame::contextForSize(unsigned log2FFTSize) {
   DCHECK(log2FFTSize);
-  DCHECK_LE(log2FFTSize, kMaxFFTPow2Size);
+  ASSERT(log2FFTSize <= kMaxFFTPow2Size);
   int bufSize;
   OMXResult status = omxSP_FFTGetBufSize_R_F32(log2FFTSize, &bufSize);
 

third_party/WebKit/Source/platform/audio/ffmpeg/FFTFrameFFMPEG.cpp

@@ -42,9 +42,11 @@ extern "C" {
 
 namespace blink {
 
+#if DCHECK_IS_ON()
 // Max FFT size for FFMPEG.  WebAudio currently only uses FFTs up to size 15
 // (2^15 points).
 const int kMaxFFTPow2Size = 16;
+#endif
 
 // Normal constructor: allocates for a given fftSize.
 FFTFrame::FFTFrame(unsigned fftSize)
@@ -56,7 +58,7 @@ FFTFrame::FFTFrame(unsigned fftSize)
       m_inverseContext(nullptr),
       m_complexData(fftSize) {
   // We only allow power of two.
-  DCHECK_EQ(1UL << m_log2FFTSize, m_FFTSize);
+  ASSERT(1UL << m_log2FFTSize == m_FFTSize);
 
   m_forwardContext = contextForSize(fftSize, DFT_R2C);
   m_inverseContext = contextForSize(fftSize, IDFT_C2R);
@@ -155,7 +157,7 @@ RDFTContext* FFTFrame::contextForSize(unsigned fftSize, int trans) {
   // by sharing the FFTFrames on a per-thread basis.
   DCHECK(fftSize);
   int pow2size = static_cast<int>(log2(fftSize));
-  DCHECK_LT(pow2size, kMaxFFTPow2Size);
+  ASSERT(pow2size < kMaxFFTPow2Size);
 
   RDFTContext* context = av_rdft_init(pow2size, (RDFTransformType)trans);
   return context;

third_party/WebKit/Source/platform/audio/ipp/FFTFrameIPP.cpp

@@ -47,8 +47,8 @@ FFTFrame::FFTFrame(unsigned fftSize)
       m_imagData(fftSize / 2),
       m_complexData(fftSize) {
   // We only allow power of two.
-  DCHECK_EQ(1UL << m_log2FFTSize, m_FFTSize);
-  DCHECK_LE(m_log2FFTSize, maximumFFTPower2Size);
+  ASSERT(1UL << m_log2FFTSize == m_FFTSize);
+  ASSERT(m_log2FFTSize <= maximumFFTPower2Size);
 
   ippsDFTInitAlloc_R_32f(&m_DFTSpec, m_FFTSize, IPP_FFT_NODIV_BY_ANY,
                          ippAlgHintFast);

third_party/WebKit/Source/platform/audio/mac/FFTFrameMac.cpp

@@ -49,7 +49,7 @@ FFTFrame::FFTFrame(unsigned fftSize)
   m_log2FFTSize = static_cast<unsigned>(log2(fftSize));
 
   // We only allow power of two
-  DCHECK_EQ(1UL << m_log2FFTSize, m_FFTSize);
+  ASSERT(1UL << m_log2FFTSize == m_FFTSize);
 
   // Lazily create and share fftSetup with other frames
   m_FFTSetup = fftSetupForSize(fftSize);
@@ -116,7 +116,7 @@ FFTSetup FFTFrame::fftSetupForSize(unsigned fftSize) {
   }
 
   int pow2size = static_cast<int>(log2(fftSize));
-  DCHECK_LT(pow2size, kMaxFFTPow2Size);
+  ASSERT(pow2size < kMaxFFTPow2Size);
   if (!fftSetups[pow2size])
     fftSetups[pow2size] = vDSP_create_fftsetup(pow2size, FFT_RADIX2);