Don't leave the negative frequencies as 0 for inverse FFT

alc/effects/convolution.cpp

@@ -266,8 +266,7 @@ void ConvolutionState::update(const ALCcontext *context, const ALeffectslot *slo
     /* The iFFT'd response is scaled up by the number of bins, so apply the
      * inverse to the output mixing gain.
      */
-    constexpr size_t m{ConvolveUpdateSize/2 + 1};
-    const float gain{slot->Params.Gain * (1.0f/m)};
+    const float gain{slot->Params.Gain * (1.0f/float{ConvolveUpdateSize})};
     auto &chans = *mChans;
     if(mChannels == FmtBFormat3D || mChannels == FmtBFormat2D)
     {
@@ -386,6 +385,12 @@ void ConvolutionState::process(const size_t samplesToDo,
                     mFftBuffer[i] += *input * *filter;
             }
 
+            /* Reconstruct the mirrored/negative frequencies to do a proper
+             * inverse FFT.
+             */
+            for(size_t i{m};i < ConvolveUpdateSize;++i)
+                mFftBuffer[i] = std::conj(mFftBuffer[ConvolveUpdateSize-i]);
+
             /* Apply iFFT to get the 1024 (really 1023) samples for output. The
              * 512 output samples are combined with the last output's 511
              * second-half samples (and this output's second half is

alc/effects/pshifter.cpp

@@ -226,15 +226,15 @@ void PshifterState::process(const size_t samplesToDo, const al::span<const Float
 
             mFftBuffer[k] = std::polar(mSynthesisBuffer[k].Amplitude, mSumPhase[k]);
         }
-        /* Clear negative frequencies to recontruct the time-domain signal. */
-        std::fill(mFftBuffer.begin()+STFT_HALF_SIZE+1, mFftBuffer.end(), complex_d{});
+        for(size_t k{STFT_HALF_SIZE+1};k < STFT_SIZE;++k)
+            mFftBuffer[k] = std::conj(mFftBuffer[STFT_SIZE-k]);
 
         /* Apply an inverse FFT to get the time-domain siganl, and accumulate
          * for the output with windowing.
          */
         complex_fft(mFftBuffer, 1.0);
         for(size_t k{0u};k < STFT_SIZE;k++)
-            mOutputAccum[k] += HannWindow[k]*mFftBuffer[k].real() * (2.0/STFT_HALF_SIZE/OVERSAMP);
+            mOutputAccum[k] += HannWindow[k]*mFftBuffer[k].real() * (2.0/STFT_SIZE/OVERSAMP);
 
         /* Shift FIFO and accumulator. */
         fifo_iter = std::copy(mFIFO.begin()+STFT_STEP, mFIFO.end(), mFIFO.begin());

alc/uhjfilter.cpp

@@ -44,18 +44,21 @@ std::array<float,Uhj2Encoder::sFilterSize> GenerateFilter()
      */
     constexpr complex_d c0{0.0,  1.0};
     constexpr complex_d c1{0.0, -1.0};
-    constexpr size_t half_size{32768};
+    constexpr size_t fft_size{65536};
+    constexpr size_t half_size{fft_size / 2};
 
-    /* Generate a frequency domain impulse with a +90 degree phase offset. Keep
-     * the mirrored frequencies clear for converting to the time domain.
+    /* Generate a frequency domain impulse with a +90 degree phase offset.
+     * Reconstruct the mirrored frequencies to convert to the time domain.
      */
-    auto fftBuffer = std::vector<complex_d>(half_size*2, complex_d{});
+    auto fftBuffer = std::vector<complex_d>(fft_size, complex_d{});
     for(size_t i{0};i < half_size;i += 2)
     {
         fftBuffer[i  ] = c0;
         fftBuffer[i+1] = c1;
     }
     fftBuffer[half_size] = c0;
+    for(size_t i{half_size+1};i < fft_size;++i)
+        fftBuffer[i] = std::conj(fftBuffer[fft_size - i]);
     complex_fft(fftBuffer, 1.0);
 
     /* Reverse and truncate the filter to a usable size, and store only the
@@ -65,7 +68,7 @@ std::array<float,Uhj2Encoder::sFilterSize> GenerateFilter()
     auto fftiter = fftBuffer.data() + half_size + (Uhj2Encoder::sFilterSize-1);
     for(float &coeff : ret)
     {
-        coeff = static_cast<float>(fftiter->real() / (half_size+1));
+        coeff = static_cast<float>(fftiter->real() / double{fft_size});
         fftiter -= 2;
     }
     return ret;