Post merge fixup

Externals/FreeSurround/source/FreeSurroundDecoder.cpp

@@ -92,7 +92,7 @@ void DPL2FSDecoder::Init(channel_setup chsetup, unsigned int blsize,
         memcpy(&outbuf[i], &outbuf[prev_size - C], sizeof(float) * C);
     }
   }
-  for (unsigned int k = 0; k < std::min(C, unsigned int(signal.size())); k++)
+  for (unsigned int k = 0; k < std::min(C, static_cast<unsigned int>(signal.size())); k++)
     signal[k].resize(N);
   signal.resize(C, std::vector<cplx>(N));
 

Source/Core/AudioCommon/AudioCommon.cpp

@@ -60,49 +60,33 @@ static std::unique_ptr<SoundStream> CreateSoundStreamForBackend(std::string_view
 
 void InitSoundStream()
 {
+  std::lock_guard<std::mutex> guard(g_sound_stream_mutex);
+
+  std::string backend = SConfig::GetInstance().sBackend;
+  g_sound_stream = CreateSoundStreamForBackend(backend);
+  g_selected_sound_stream_failed = false;
+
+  if (!g_sound_stream)
   {
-    WARN_LOG_FMT(AUDIO, "Unknown backend {}, using {} instead.", backend, GetDefaultSoundBackend());
+    WARN_LOG_FMT(AUDIO, "Unknown backend {}, using {} instead (default)", backend,
+                  GetDefaultSoundBackend());
     backend = GetDefaultSoundBackend();
-    g_sound_stream = CreateSoundStreamForBackend(GetDefaultSoundBackend());
+    g_sound_stream = CreateSoundStreamForBackend(backend);
   }
 
   if (!g_sound_stream || !g_sound_stream->Init())
   {
-    WARN_LOG_FMT(AUDIO, "Could not initialize backend {}, using {} instead.", backend,
-                 BACKEND_NULLSOUND);
+    WARN_LOG_FMT(AUDIO, "Could not initialize backend {}, using {} instead", backend,
+                  NullSound::GetName());
     g_sound_stream = std::make_unique<NullSound>();
-    g_sound_stream->Init();
+    g_sound_stream->Init();  // NullSound can't fail
+    g_selected_sound_stream_failed = true;
   }
 }
 
+// This needs to be called after AudioInterface::Init where input sample rates are set
 void PostInitSoundStream()
 {
-  // This needs to be called after AudioInterface::Init where input sample rates are set
-  UpdateSoundStream();
-    std::lock_guard<std::mutex> guard(g_sound_stream_mutex);
-
-    std::string backend = SConfig::GetInstance().sBackend;
-    g_sound_stream = CreateSoundStreamForBackend(backend);
-    g_selected_sound_stream_failed = false;
-
-    if (!g_sound_stream)
-    {
-      WARN_LOG_FMT(AUDIO, "Unknown backend {}, using {} instead (default)", backend,
-               GetDefaultSoundBackend());
-      backend = GetDefaultSoundBackend();
-      g_sound_stream = CreateSoundStreamForBackend(backend);
-    }
-
-    if (!g_sound_stream || !g_sound_stream->Init())
-    {
-      WARN_LOG_FMT(AUDIO, "Could not initialize backend {}, using {} instead", backend,
-               NullSound::GetName());
-      g_sound_stream = std::make_unique<NullSound>();
-      g_sound_stream->Init();  // NullSound can't fail
-      g_selected_sound_stream_failed = true;
-    }
-  }
-
   UpdateSoundStreamSettings(true);
   // This can fail, but we don't really care as it just won't produce any sounds,
   // also the user might be able to fix it up by changing their device settings
@@ -111,13 +95,6 @@ void PostInitSoundStream()
   // and true again, so basically the backend is "restarted" with every emulation state change
   SetSoundStreamRunning(true, true);
 
-  //To review: are these still needed after merge with master?
-  // Ideally these two calls would be done in AudioInterface::Init so that we don't
-  // need to have a dependency on AudioInterface here, but this has to be done
-  // after creating g_sound_stream (above) and before starting audio dumping (below)
-  g_sound_stream->GetMixer()->SetDMAInputSampleRate(AudioInterface::GetAIDSampleRate());
-  g_sound_stream->GetMixer()->SetStreamingInputSampleRate(AudioInterface::GetAISSampleRate());
-
   if (SConfig::GetInstance().m_DumpAudio && !s_audio_dump_started)
     StartAudioDump();
 }

Source/Core/AudioCommon/Mixer.cpp

@@ -45,7 +45,8 @@ Mixer::Mixer(u32 sample_rate)
 
 Mixer::~Mixer()
 {
-  INFO_LOG_FMT(AUDIO_INTERFACE, "Mixer is initialized");
+  INFO_LOG_FMT(AUDIO_INTERFACE, "Mixer is uninitialized");
+  Core::RemoveOnStateChangedCallback(&m_on_state_changed_handle);
 }
 
 void Mixer::SetPaused(bool paused)

Source/Core/AudioCommon/Mixer.h

@@ -89,8 +89,8 @@ class Mixer final
   {
   public:
     MixerFifo(Mixer* mixer, unsigned sample_rate, bool big_endians, bool constantly_pushed = true)
-        : m_mixer(mixer), m_input_sample_rate(sample_rate),
-          m_big_endians(big_endians), m_constantly_pushed(constantly_pushed)
+        : m_mixer(mixer), m_input_sample_rate(sample_rate), m_constantly_pushed(constantly_pushed),
+          m_big_endians(big_endians)
     {
     }
     void DoState(PointerWrap& p);
@@ -191,7 +191,7 @@ class Mixer final
 
   u32 m_sample_rate;  // Only changed by main or emulation thread when the backend is not running
   bool m_stretching = false;
-  bool m_update_surround_latency = false;
+  bool m_update_surround_latency = false; //To delete?
   AudioCommon::AudioStretcher m_stretcher;
   AudioCommon::SurroundDecoder m_surround_decoder;
 

Source/Core/Core/ConfigLoaders/IsSettingSaveable.cpp

@@ -44,9 +44,9 @@ bool IsSettingSaveable(const Config::Location& config_location)
       &Config::MAIN_ALLOW_SD_WRITES.GetLocation(),
       &Config::MAIN_DPL2_DECODER.GetLocation(),
       &Config::MAIN_DPL2_QUALITY.GetLocation(),
-      &Config::MAIN_DPL2_BASS_REDIRECTION.location,
-      &Config::MAIN_AUDIO_MIXER_MIN_LATENCY.location,
-      &Config::MAIN_AUDIO_MIXER_MAX_LATENCY.location,
+      &Config::MAIN_DPL2_BASS_REDIRECTION.GetLocation(),
+      &Config::MAIN_AUDIO_MIXER_MIN_LATENCY.GetLocation(),
+      &Config::MAIN_AUDIO_MIXER_MAX_LATENCY.GetLocation(),
       &Config::MAIN_RAM_OVERRIDE_ENABLE.GetLocation(),
       &Config::MAIN_MEM1_SIZE.GetLocation(),
       &Config::MAIN_MEM2_SIZE.GetLocation(),

Source/Core/Core/HW/AudioInterface.cpp

@@ -154,9 +154,8 @@ void Init()
 
   event_type_ai = CoreTiming::RegisterEvent("AICallback", Update);
 
-  //To review: this is now duplicate?
   g_sound_stream->GetMixer()->SetDMAInputSampleRate(GetAIDSampleRate());
-  g_sound_stream->GetMixer()->SetStreamInputSampleRate(GetAISSampleRate());
+  g_sound_stream->GetMixer()->SetStreamingInputSampleRate(GetAISSampleRate());
 }
 
 void Shutdown()

Source/Core/Core/HW/WiimoteEmu/Speaker.cpp

@@ -91,7 +91,7 @@ void stopdamnwav()
 }
 #endif
 
-void SpeakerLogic::SpeakerData(const u8* data, int length)
+void SpeakerLogic::SpeakerData(const u8* data, int length, float speaker_pan)
 {
   if (reg_data.sample_rate == 0 || length == 0)
     return;
@@ -174,7 +174,7 @@ void SpeakerLogic::SpeakerData(const u8* data, int length)
     // We should play the samples from the wiimote at the native volume they came with,
     // because you can lower their volume from the wii settings, and because they are
     // already extremely low quality, so any additional quality loss isn't welcome.
-    float speaker_pan = std::clamp(float(m_speaker_pan_setting.GetValue()) / 100, -1.f, 1.f);
+    speaker_pan = std::clamp(speaker_pan, -1.f, 1.f);
     const u32 l_volume = std::min(u32(std::min(1.f - speaker_pan, 1.f) * volume), 255u);
     const u32 r_volume = std::min(u32(std::min(1.f + speaker_pan, 1.f) * volume), 255u);
 

Source/Core/Core/HW/WiimoteEmu/WiimoteEmu.cpp

@@ -166,7 +166,6 @@ void Wiimote::Reset()
   m_i2c_bus.AddSlave(&m_speaker_logic);
   m_i2c_bus.AddSlave(&m_camera_logic);
 
-  //To review: is this needed now?
   m_speaker_logic.m_index = m_index;
 
   // Reset extension connections to NONE:

Source/Core/Core/State.cpp

@@ -74,7 +74,7 @@ static Common::Event g_compressAndDumpStateSyncEvent;
 static std::thread g_save_thread;
 
 // Don't forget to increase this after doing changes on the savestate system
-constexpr u32 STATE_VERSION = 131;  // Last changed in PR ????
+constexpr u32 STATE_VERSION = 130;  // Last changed in PR 9545
 
 // Maps savestate versions to Dolphin versions.
 // Versions after 42 don't need to be added to this list,

Source/Core/DolphinLib.props

@@ -2,6 +2,7 @@
 <Project xmlns="http://schemas.microsoft.com/developer/msbuild/2003">
   <ItemGroup>
     <ClInclude Include="AudioCommon\AudioCommon.h" />
+    <ClInclude Include="AudioCommon\AudioSpeedCounter.h" />
     <ClInclude Include="AudioCommon\AudioStretcher.h" />
     <ClInclude Include="AudioCommon\CubebStream.h" />
     <ClInclude Include="AudioCommon\CubebUtils.h" />
@@ -667,12 +668,14 @@
   </ItemGroup>
   <ItemGroup>
     <ClCompile Include="AudioCommon\AudioCommon.cpp" />
+    <ClCompile Include="AudioCommon\AudioSpeedCounter.cpp" />
     <ClCompile Include="AudioCommon\AudioStretcher.cpp" />
     <ClCompile Include="AudioCommon\CubebStream.cpp" />
     <ClCompile Include="AudioCommon\CubebUtils.cpp" />
     <ClCompile Include="AudioCommon\Mixer.cpp" />
     <ClCompile Include="AudioCommon\NullSoundStream.cpp" />
     <ClCompile Include="AudioCommon\OpenALStream.cpp" />
+    <ClCompile Include="AudioCommon\SoundStream.cpp" />
     <ClCompile Include="AudioCommon\SurroundDecoder.cpp" />
     <ClCompile Include="AudioCommon\WASAPIStream.cpp" />
     <ClCompile Include="AudioCommon\WaveFile.cpp" />