Fixed crackling on some loud samples

Source/Core/AudioCommon/AudioStretcher.cpp

@@ -39,22 +39,22 @@ void AudioStretcher::PushSamples(const s16* in, u32 num_in)
   }
 }
 
+// This won't return any samples a lot of times as they are processed in batches
 u32 AudioStretcher::GetStretchedSamples(s16* out, u32 num_out, bool pad)
 {
-  // This won't return any samples a lot of times as they are processed in batches
   u32 samples_received = m_sound_touch.receiveSamples(out, num_out);
 
-  if (!pad)
-  {
-    return samples_received;
-  }
-
   if (samples_received != 0)
   {
     m_last_stretched_sample[0] = out[samples_received * 2 - 2];
     m_last_stretched_sample[1] = out[samples_received * 2 - 1];
   }
 
+  if (!pad)
+  {
+    return samples_received;
+  }
+
   // Perform padding if we've run out of samples
   for (u32 i = samples_received; i < num_out; ++i)
   {

Source/Core/AudioCommon/Mixer.cpp

@@ -20,7 +20,7 @@
 
 #include "Common/Logging/Log.h" //To delete and all the uses (or make debug log)
 #include "VideoCommon/OnScreenDisplay.h" //To delete and all the uses
-#pragma optimize("", off) //To delete
+//#pragma optimize("", off) //To delete
 
 Mixer::Mixer(u32 sample_rate)
     : m_sample_rate(sample_rate), m_stretcher(sample_rate),
@@ -102,7 +102,7 @@ u32 Mixer::MixerFifo::Mix(s16* samples, u32 num_samples, bool stretching)
   s32 lVolume = m_lVolume.load();
   s32 rVolume = m_rVolume.load();
 
-  s16 s[NC]; // Padding samples
+  s32 s[NC];  // Padding samples
   s[0] = m_last_output_samples[1];
   s[1] = m_last_output_samples[1];
 
@@ -194,10 +194,10 @@ u32 Mixer::MixerFifo::Mix(s16* samples, u32 num_samples, bool stretching)
     unsigned int current_sample = actual_samples_count * 2;
     for (; current_sample < num_samples * 2; current_sample += 2)
     {
-      samples[current_sample + 0] =
-          std::clamp(m_last_output_samples[0] + samples[current_sample + 0], -SHRT_MAX, SHRT_MAX);
-      samples[current_sample + 1] =
-          std::clamp(m_last_output_samples[1] + samples[current_sample + 1], -SHRT_MAX, SHRT_MAX);
+      samples[current_sample + 0] = std::clamp(
+          samples[current_sample + 0] + m_last_output_samples[0], SHRT_MIN, SHRT_MAX);
+      samples[current_sample + 1] = std::clamp(
+          samples[current_sample + 1] + m_last_output_samples[1], SHRT_MIN, SHRT_MAX);
     }
   }
   //if (indexW > 8 && !m_constantly_pushed) OSD::AddMessage("last sample: " + std::to_string(m_last_output_samples[0]), 0U);
@@ -209,7 +209,7 @@ u32 Mixer::MixerFifo::Mix(s16* samples, u32 num_samples, bool stretching)
 
 // Sounds better than linear interpolation
 u32 Mixer::MixerFifo::CubicInterpolation(s16* samples, u32 num_samples, double rate, u32& indexR,
-                                         u32 indexW, s16& l_s, s16& r_s, s32 lVolume, s32 rVolume,
+                                         u32 indexW, s32& l_s, s32& r_s, s32 lVolume, s32 rVolume,
                                          bool forwards)
 {
   // Coefficients copied from SoundTouch Copyright © Olli Parviainen 2001-2019
@@ -301,14 +301,14 @@ u32 Mixer::MixerFifo::CubicInterpolation(s16* samples, u32 num_samples, double r
                   y3 * interpolation_buffer[(indexR + 6 * direction + 1) & INDEX_MASK];
     //To test left and right on HW source
 
-    l_s = (s16(std::round(l_s_f)) * lVolume) >> 8;
-    r_s = (s16(std::round(r_s_f)) * rVolume) >> 8;
+    l_s = (s32(std::round(l_s_f)) * lVolume) >> 8; //To round again post vol? //To use std::rint?
+    r_s = (s32(std::round(r_s_f)) * rVolume) >> 8;
     // Clamp after adding to current sample, as if the cubic interpolation produced a sample over
     // the limits and the current sample has the opposite sign, then we'd keep the excess value,
     // while if it had the same sign, it would have been clamped anyway.
-    // We don't clamp to SHRT_MIN to keep the center at 0, though audio devices can take SHRT_MIN
-    samples[i * NC + 0] = std::clamp(samples[i * NC + 0] + l_s, -SHRT_MAX, SHRT_MAX);
-    samples[i * NC + 1] = std::clamp(samples[i * NC + 1] + r_s, -SHRT_MAX, SHRT_MAX);
+    // Instead of clamping to -SHORT_MAX like before to keep 0 centered, we use the full short range
+    samples[i * NC + 0] = std::clamp(samples[i * NC + 0] + l_s, SHRT_MIN, SHRT_MAX);
+    samples[i * NC + 1] = std::clamp(samples[i * NC + 1] + r_s, SHRT_MIN, SHRT_MAX);
 
     ++i;
   }
@@ -597,6 +597,7 @@ u32 Mixer::MixSurround(float* samples, u32 num_samples)
   if (available_samples != needed_samples)
   {
     ERROR_LOG(AUDIO, "Error decoding surround samples");
+    // This needs to do padding
     return 0;
   }
 

Source/Core/AudioCommon/Mixer.h

@@ -100,9 +100,11 @@ class Mixer final
     u32 GetNextIndexR(u32 indexR) const;
     void UpdatePush(double time);
 
-    // Returns the actual number of samples written. Outputs the last played sample for padding
+    // Returns the actual number of samples written. Outputs the last played sample for padding.
+    // We pass in in/out samples to cache the last calculated value. It's in int32 range because
+    // cubic interpolation can produce values over int16 and we don't want to lose the extra
     u32 CubicInterpolation(s16* samples, u32 num_samples, double rate, u32& indexR, u32 indexW,
-                           s16& l_s, s16& r_s, s32 lVolume, s32 rVolume, bool forwards = true);
+                           s32& l_s, s32& r_s, s32 lVolume, s32 rVolume, bool forwards = true);
 
   private:
     Mixer* m_mixer;
@@ -128,7 +130,8 @@ class Mixer final
     // Volume range is 0-256
     std::atomic<s32> m_lVolume{256};
     std::atomic<s32> m_rVolume{256};
-    s16 m_last_output_samples[NC]{};
+    // Se comment on CubicInterpolation()
+    s32 m_last_output_samples[NC]{};
     double m_fract = -1.0;
     double m_backwards_fract = -1.0;
     //To review: if this was off, that mixer won't gather a buffer/latency, and it would always be on the brink of playback

Source/Core/AudioCommon/SurroundDecoder.cpp

@@ -10,7 +10,7 @@
 
 #include "AudioCommon/Enums.h"
 #include "AudioCommon/SurroundDecoder.h"
-#pragma optimize("", off) //To delete
+//#pragma optimize("", off) //To delete
 
 namespace AudioCommon
 {