Fixups in audio backend

Removes 's_' prefix from variables that are no longer static and
thread_local. Removes superfluous comments left behind due to copy-paste
mistakes.

rpcs3/Emu/Audio/ALSA/ALSABackend.cpp

@@ -28,7 +28,7 @@ ALSABackend::ALSABackend()
 
 ALSABackend::~ALSABackend()
 {
-	if (s_tls_sw_params || s_tls_hw_params || s_tls_handle)
+	if (tls_sw_params || tls_hw_params || tls_handle)
 	{
 		Close();
 	}
@@ -36,71 +36,71 @@ ALSABackend::~ALSABackend()
 
 void ALSABackend::Open(u32 num_buffers)
 {
-	if (!check(snd_pcm_open(&s_tls_handle, "default", SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK), "snd_pcm_open"))
+	if (!check(snd_pcm_open(&tls_handle, "default", SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK), "snd_pcm_open"))
 		return;
 
-	if (!check(snd_pcm_hw_params_malloc(&s_tls_hw_params), "snd_pcm_hw_params_malloc"))
+	if (!check(snd_pcm_hw_params_malloc(&tls_hw_params), "snd_pcm_hw_params_malloc"))
 		return;
 
-	if (!check(snd_pcm_hw_params_any(s_tls_handle, s_tls_hw_params), "snd_pcm_hw_params_any"))
+	if (!check(snd_pcm_hw_params_any(tls_handle, tls_hw_params), "snd_pcm_hw_params_any"))
 		return;
 
-	if (!check(snd_pcm_hw_params_set_access(s_tls_handle, s_tls_hw_params, SND_PCM_ACCESS_RW_INTERLEAVED), "snd_pcm_hw_params_set_access"))
+	if (!check(snd_pcm_hw_params_set_access(tls_handle, tls_hw_params, SND_PCM_ACCESS_RW_INTERLEAVED), "snd_pcm_hw_params_set_access"))
 		return;
 
-	if (!check(snd_pcm_hw_params_set_format(s_tls_handle, s_tls_hw_params, g_cfg.audio.convert_to_u16 ? SND_PCM_FORMAT_S16_LE : SND_PCM_FORMAT_FLOAT_LE), "snd_pcm_hw_params_set_format"))
+	if (!check(snd_pcm_hw_params_set_format(tls_handle, tls_hw_params, g_cfg.audio.convert_to_u16 ? SND_PCM_FORMAT_S16_LE : SND_PCM_FORMAT_FLOAT_LE), "snd_pcm_hw_params_set_format"))
 		return;
 
 	uint rate = get_sampling_rate();
-	if (!check(snd_pcm_hw_params_set_rate_near(s_tls_handle, s_tls_hw_params, &rate, nullptr), "snd_pcm_hw_params_set_rate_near"))
+	if (!check(snd_pcm_hw_params_set_rate_near(tls_handle, tls_hw_params, &rate, nullptr), "snd_pcm_hw_params_set_rate_near"))
 		return;
 
-	if (!check(snd_pcm_hw_params_set_channels(s_tls_handle, s_tls_hw_params, get_channels()), "snd_pcm_hw_params_set_channels"))
+	if (!check(snd_pcm_hw_params_set_channels(tls_handle, tls_hw_params, get_channels()), "snd_pcm_hw_params_set_channels"))
 		return;
 
 	//uint period = 5333;
-	//if (!check(snd_pcm_hw_params_set_period_time_near(s_tls_handle, s_tls_hw_params, &period, nullptr), "snd_pcm_hw_params_set_period_time_near"))
+	//if (!check(snd_pcm_hw_params_set_period_time_near(tls_handle, tls_hw_params, &period, nullptr), "snd_pcm_hw_params_set_period_time_near"))
 	//	return;
 
-	//if (!check(snd_pcm_hw_params_set_periods(s_tls_handle, s_tls_hw_params, 4, 0), "snd_pcm_hw_params_set_periods"))
+	//if (!check(snd_pcm_hw_params_set_periods(tls_handle, tls_hw_params, 4, 0), "snd_pcm_hw_params_set_periods"))
 	//	return;
 
 	snd_pcm_uframes_t bufsize_frames = num_buffers * AUDIO_BUFFER_SAMPLES;
 	snd_pcm_uframes_t period_frames = AUDIO_BUFFER_SAMPLES;
 
-	if (!check(snd_pcm_hw_params_set_buffer_size_near(s_tls_handle, s_tls_hw_params, &bufsize_frames), "snd_pcm_hw_params_set_buffer_size_near"))
+	if (!check(snd_pcm_hw_params_set_buffer_size_near(tls_handle, tls_hw_params, &bufsize_frames), "snd_pcm_hw_params_set_buffer_size_near"))
 		return;
 
-	if (!check(snd_pcm_hw_params_set_period_size_near(s_tls_handle, s_tls_hw_params, &period_frames, 0), "snd_pcm_hw_params_set_period_size"))
+	if (!check(snd_pcm_hw_params_set_period_size_near(tls_handle, tls_hw_params, &period_frames, 0), "snd_pcm_hw_params_set_period_size"))
 		return;
 
-	if (!check(snd_pcm_hw_params(s_tls_handle, s_tls_hw_params), "snd_pcm_hw_params"))
+	if (!check(snd_pcm_hw_params(tls_handle, tls_hw_params), "snd_pcm_hw_params"))
 		return;
 
-	if (!check(snd_pcm_hw_params_get_buffer_size(s_tls_hw_params, &bufsize_frames), "snd_pcm_hw_params_get_buffer_size"))
+	if (!check(snd_pcm_hw_params_get_buffer_size(tls_hw_params, &bufsize_frames), "snd_pcm_hw_params_get_buffer_size"))
 		return;
 
-	if (!check(snd_pcm_hw_params_get_period_size(s_tls_hw_params, &period_frames, nullptr), "snd_pcm_hw_params_get_period_size"))
+	if (!check(snd_pcm_hw_params_get_period_size(tls_hw_params, &period_frames, nullptr), "snd_pcm_hw_params_get_period_size"))
 		return;
 
-	if (!check(snd_pcm_sw_params_malloc(&s_tls_sw_params), "snd_pcm_sw_params_malloc"))
+	if (!check(snd_pcm_sw_params_malloc(&tls_sw_params), "snd_pcm_sw_params_malloc"))
 		return;
 
-	if (!check(snd_pcm_sw_params_current(s_tls_handle, s_tls_sw_params), "snd_pcm_sw_params_current"))
+	if (!check(snd_pcm_sw_params_current(tls_handle, tls_sw_params), "snd_pcm_sw_params_current"))
 		return;
 
 	period_frames *= g_cfg.audio.startt;
 
-	if (!check(snd_pcm_sw_params_set_start_threshold(s_tls_handle, s_tls_sw_params, period_frames + 1), "snd_pcm_sw_params_set_start_threshold"))
+	if (!check(snd_pcm_sw_params_set_start_threshold(tls_handle, tls_sw_params, period_frames + 1), "snd_pcm_sw_params_set_start_threshold"))
 		return;
 
-	if (!check(snd_pcm_sw_params_set_stop_threshold(s_tls_handle, s_tls_sw_params, bufsize_frames), "snd_pcm_sw_params_set_stop_threshold"))
+	if (!check(snd_pcm_sw_params_set_stop_threshold(tls_handle, tls_sw_params, bufsize_frames), "snd_pcm_sw_params_set_stop_threshold"))
 		return;
 
-	if (!check(snd_pcm_sw_params(s_tls_handle, s_tls_sw_params), "snd_pcm_sw_params"))
+	if (!check(snd_pcm_sw_params(tls_handle, tls_sw_params), "snd_pcm_sw_params"))
 		return;
 
-	if (!check(snd_pcm_prepare(s_tls_handle), "snd_pcm_prepare"))
+	if (!check(snd_pcm_prepare(tls_handle), "snd_pcm_prepare"))
 		return;
 
 	LOG_NOTICE(GENERAL, "ALSA: bufsize_frames=%u, period_frames=%u", bufsize_frames, period_frames);
@@ -108,22 +108,22 @@ void ALSABackend::Open(u32 num_buffers)
 
 void ALSABackend::Close()
 {
-	if (s_tls_sw_params)
+	if (tls_sw_params)
 	{
-		snd_pcm_sw_params_free(s_tls_sw_params);
+		snd_pcm_sw_params_free(tls_sw_params);
-		s_tls_sw_params = nullptr;
+		tls_sw_params = nullptr;
 	}
 
-	if (s_tls_hw_params)
+	if (tls_hw_params)
 	{
-		snd_pcm_hw_params_free(s_tls_hw_params);
+		snd_pcm_hw_params_free(tls_hw_params);
-		s_tls_hw_params = nullptr;
+		tls_hw_params = nullptr;
 	}
 
-	if (s_tls_handle)
+	if (tls_handle)
 	{
-		snd_pcm_close(s_tls_handle);
+		snd_pcm_close(tls_handle);
-		s_tls_handle = nullptr;
+		tls_handle = nullptr;
 	}
 }
 
@@ -131,7 +131,7 @@ bool ALSABackend::AddData(const void* src, u32 num_samples)
 {
 	u32 num_frames = num_samples / get_channels();
 
-	int res = snd_pcm_writei(s_tls_handle, src, num_frames);
+	int res = snd_pcm_writei(tls_handle, src, num_frames);
 
 	if (res == -EAGAIN)
 	{
@@ -141,7 +141,7 @@ bool ALSABackend::AddData(const void* src, u32 num_samples)
 
 	if (res < 0)
 	{
-		res = snd_pcm_recover(s_tls_handle, res, 0);
+		res = snd_pcm_recover(tls_handle, res, 0);
 
 		if (res < 0)
 		{
@@ -149,7 +149,7 @@ bool ALSABackend::AddData(const void* src, u32 num_samples)
 			return false;
 		}
 
-		res = snd_pcm_writei(s_tls_handle, src, num_frames);
+		res = snd_pcm_writei(tls_handle, src, num_frames);
 	}
 
 	if (res != num_frames)

rpcs3/Emu/Audio/ALSA/ALSABackend.h

@@ -8,9 +8,9 @@
 
 class ALSABackend : public AudioBackend
 {
-	snd_pcm_t* s_tls_handle{nullptr};
+	snd_pcm_t* tls_handle{nullptr};
-	snd_pcm_hw_params_t* s_tls_hw_params{nullptr};
+	snd_pcm_hw_params_t* tls_hw_params{nullptr};
-	snd_pcm_sw_params_t* s_tls_sw_params{nullptr};
+	snd_pcm_sw_params_t* tls_sw_params{nullptr};
 	
 public:
 	ALSABackend();

rpcs3/Emu/Audio/XAudio2/XAudio27Backend.cpp

@@ -9,14 +9,14 @@
 
 class XAudio27Library : public XAudio2Backend::XAudio2Library
 {
-	const HMODULE s_tls_xaudio2_lib;
+	const HMODULE tls_xaudio2_lib;
-	IXAudio2* s_tls_xaudio2_instance{};
+	IXAudio2* tls_xaudio2_instance{};
-	IXAudio2MasteringVoice* s_tls_master_voice{};
+	IXAudio2MasteringVoice* tls_master_voice{};
-	IXAudio2SourceVoice* s_tls_source_voice{};
+	IXAudio2SourceVoice* tls_source_voice{};
 
 public:
 	XAudio27Library(void* lib2_7)
-		: s_tls_xaudio2_lib(static_cast<HMODULE>(lib2_7))
+		: tls_xaudio2_lib(static_cast<HMODULE>(lib2_7))
 	{
 		HRESULT hr = S_OK;
 
@@ -28,7 +28,7 @@ public:
 			return;
 		}
 
-		hr = XAudio2Create(&s_tls_xaudio2_instance, 0, XAUDIO2_DEFAULT_PROCESSOR);
+		hr = XAudio2Create(&tls_xaudio2_instance, 0, XAUDIO2_DEFAULT_PROCESSOR);
 		if (FAILED(hr))
 		{
 			LOG_ERROR(GENERAL, "XAudio2Backend : XAudio2Create() failed(0x%08x)", (u32)hr);
@@ -36,42 +36,42 @@ public:
 			return;
 		}
 
-		hr = s_tls_xaudio2_instance->CreateMasteringVoice(&s_tls_master_voice, g_cfg.audio.downmix_to_2ch ? 2 : 8, 48000);
+		hr = tls_xaudio2_instance->CreateMasteringVoice(&tls_master_voice, g_cfg.audio.downmix_to_2ch ? 2 : 8, 48000);
 		if (FAILED(hr))
 		{
 			LOG_ERROR(GENERAL, "XAudio2Backend : CreateMasteringVoice() failed(0x%08x)", (u32)hr);
-			s_tls_xaudio2_instance->Release();
+			tls_xaudio2_instance->Release();
 			Emu.Pause();
 		}
 	}
 
 	~XAudio27Library()
 	{
-		if (s_tls_source_voice != nullptr)
+		if (tls_source_voice != nullptr)
 		{
-			s_tls_source_voice->Stop();
+			tls_source_voice->Stop();
-			s_tls_source_voice->DestroyVoice();
+			tls_source_voice->DestroyVoice();
 		}
 
-		if (s_tls_master_voice != nullptr)
+		if (tls_master_voice != nullptr)
 		{
-			s_tls_master_voice->DestroyVoice();
+			tls_master_voice->DestroyVoice();
 		}
 
-		if (s_tls_xaudio2_instance != nullptr)
+		if (tls_xaudio2_instance != nullptr)
 		{
-			s_tls_xaudio2_instance->StopEngine();
+			tls_xaudio2_instance->StopEngine();
-			s_tls_xaudio2_instance->Release();
+			tls_xaudio2_instance->Release();
 		}
 
 		CoUninitialize();
 
-		FreeLibrary(s_tls_xaudio2_lib);
+		FreeLibrary(tls_xaudio2_lib);
 	}
 
 	virtual void play() override
 	{
-		HRESULT hr = s_tls_source_voice->Start();
+		HRESULT hr = tls_source_voice->Start();
 		if (FAILED(hr))
 		{
 			LOG_ERROR(GENERAL, "XAudio2Backend : Start() failed(0x%08x)", (u32)hr);
@@ -81,7 +81,7 @@ public:
 
 	virtual void flush() override
 	{
-		HRESULT hr = s_tls_source_voice->FlushSourceBuffers();
+		HRESULT hr = tls_source_voice->FlushSourceBuffers();
 		if (FAILED(hr))
 		{
 			LOG_ERROR(GENERAL, "XAudio2Backend : FlushSourceBuffers() failed(0x%08x)", (u32)hr);
@@ -91,7 +91,7 @@ public:
 
 	virtual void stop() override
 	{
-		HRESULT hr = s_tls_source_voice->Stop();
+		HRESULT hr = tls_source_voice->Stop();
 		if (FAILED(hr))
 		{
 			LOG_ERROR(GENERAL, "XAudio2Backend : Stop() failed(0x%08x)", (u32)hr);
@@ -102,7 +102,7 @@ public:
 	virtual bool is_playing() override
 	{
 		XAUDIO2_VOICE_STATE state;
-		s_tls_source_voice->GetState(&state);
+		tls_source_voice->GetState(&state);
 
 		return state.BuffersQueued > 0 || state.pCurrentBufferContext != nullptr;
 	}
@@ -124,7 +124,7 @@ public:
 		waveformatex.wBitsPerSample = sample_size * 8;
 		waveformatex.cbSize = 0;
 
-		hr = s_tls_xaudio2_instance->CreateSourceVoice(&s_tls_source_voice, &waveformatex, 0, XAUDIO2_DEFAULT_FREQ_RATIO);
+		hr = tls_xaudio2_instance->CreateSourceVoice(&tls_source_voice, &waveformatex, 0, XAUDIO2_DEFAULT_FREQ_RATIO);
 		if (FAILED(hr))
 		{
 			LOG_ERROR(GENERAL, "XAudio2Backend : CreateSourceVoice() failed(0x%08x)", (u32)hr);
@@ -132,13 +132,13 @@ public:
 			return;
 		}
 
-		s_tls_source_voice->SetVolume(channels == 2 ? 1.0f : 4.0f);
+		tls_source_voice->SetVolume(channels == 2 ? 1.0f : 4.0f);
 	}
 
 	virtual bool add(const void* src, u32 num_samples) override
 	{
 		XAUDIO2_VOICE_STATE state;
-		s_tls_source_voice->GetState(&state);
+		tls_source_voice->GetState(&state);
 
 		// XAudio 2.7 bug workaround, when it says "SimpList: non-growable list ran out of room for new elements" and hits int 3
 		if (state.BuffersQueued >= MAX_AUDIO_BUFFERS)
@@ -159,7 +159,7 @@ public:
 		buffer.PlayBegin = 0;
 		buffer.PlayLength = AUDIO_BUFFER_SAMPLES;
 
-		HRESULT hr = s_tls_source_voice->SubmitSourceBuffer(&buffer);
+		HRESULT hr = tls_source_voice->SubmitSourceBuffer(&buffer);
 		if (FAILED(hr))
 		{
 			LOG_ERROR(GENERAL, "XAudio2Backend : AddData() failed(0x%08x)", (u32)hr);
@@ -173,7 +173,7 @@ public:
 	virtual u64 enqueued_samples() override
 	{
 		XAUDIO2_VOICE_STATE state;
-		s_tls_source_voice->GetState(&state);
+		tls_source_voice->GetState(&state);
 
 		// all buffers contain AUDIO_BUFFER_SAMPLES, so we can easily calculate how many samples there are remaining
 		return (AUDIO_BUFFER_SAMPLES - state.SamplesPlayed % AUDIO_BUFFER_SAMPLES) + (state.BuffersQueued * AUDIO_BUFFER_SAMPLES);
@@ -183,7 +183,7 @@ public:
 	{
 		new_ratio = std::clamp(new_ratio, XAUDIO2_MIN_FREQ_RATIO, XAUDIO2_DEFAULT_FREQ_RATIO);
 
-		HRESULT hr = s_tls_source_voice->SetFrequencyRatio(new_ratio);
+		HRESULT hr = tls_source_voice->SetFrequencyRatio(new_ratio);
 		if (FAILED(hr))
 		{
 			LOG_ERROR(GENERAL, "XAudio2Backend : SetFrequencyRatio() failed(0x%08x)", (u32)hr);

rpcs3/Emu/Audio/XAudio2/XAudio28Backend.cpp

@@ -9,16 +9,16 @@
 
 class XAudio28Library : public XAudio2Backend::XAudio2Library
 {
-	const HMODULE s_tls_xaudio2_lib;
+	const HMODULE tls_xaudio2_lib;
-	IXAudio2* s_tls_xaudio2_instance{};
+	IXAudio2* tls_xaudio2_instance{};
-	IXAudio2MasteringVoice* s_tls_master_voice{};
+	IXAudio2MasteringVoice* tls_master_voice{};
-	IXAudio2SourceVoice* s_tls_source_voice{};
+	IXAudio2SourceVoice* tls_source_voice{};
 
 public:
 	XAudio28Library(void* lib2_8)
-		: s_tls_xaudio2_lib(static_cast<HMODULE>(lib2_8))
+		: tls_xaudio2_lib(static_cast<HMODULE>(lib2_8))
 	{
-		const auto create = (XAudio2Create)GetProcAddress(s_tls_xaudio2_lib, "XAudio2Create");
+		const auto create = (XAudio2Create)GetProcAddress(tls_xaudio2_lib, "XAudio2Create");
 
 		HRESULT hr = S_OK;
 
@@ -30,7 +30,7 @@ public:
 			return;
 		}
 
-		hr = create(&s_tls_xaudio2_instance, 0, XAUDIO2_DEFAULT_PROCESSOR);
+		hr = create(&tls_xaudio2_instance, 0, XAUDIO2_DEFAULT_PROCESSOR);
 		if (FAILED(hr))
 		{
 			LOG_ERROR(GENERAL, "XAudio2Backend : XAudio2Create() failed(0x%08x)", (u32)hr);
@@ -38,44 +38,44 @@ public:
 			return;
 		}
 
-		hr = s_tls_xaudio2_instance->CreateMasteringVoice(&s_tls_master_voice, g_cfg.audio.downmix_to_2ch ? 2 : 8, 48000);
+		hr = tls_xaudio2_instance->CreateMasteringVoice(&tls_master_voice, g_cfg.audio.downmix_to_2ch ? 2 : 8, 48000);
 		if (FAILED(hr))
 		{
 			LOG_ERROR(GENERAL, "XAudio2Backend : CreateMasteringVoice() failed(0x%08x)", (u32)hr);
-			s_tls_xaudio2_instance->Release();
+			tls_xaudio2_instance->Release();
 			Emu.Pause();
 		}
 	}
 
 	~XAudio28Library()
 	{
-		if (s_tls_source_voice != nullptr)
+		if (tls_source_voice != nullptr)
 		{
-			s_tls_source_voice->Stop();
+			tls_source_voice->Stop();
-			s_tls_source_voice->DestroyVoice();
+			tls_source_voice->DestroyVoice();
 		}
 
-		if (s_tls_master_voice != nullptr)
+		if (tls_master_voice != nullptr)
 		{
-			s_tls_master_voice->DestroyVoice();
+			tls_master_voice->DestroyVoice();
 		}
 
-		if (s_tls_xaudio2_instance != nullptr)
+		if (tls_xaudio2_instance != nullptr)
 		{
-			s_tls_xaudio2_instance->StopEngine();
+			tls_xaudio2_instance->StopEngine();
-			s_tls_xaudio2_instance->Release();
+			tls_xaudio2_instance->Release();
 		}
 
 		CoUninitialize();
 
-		FreeLibrary(s_tls_xaudio2_lib);
+		FreeLibrary(tls_xaudio2_lib);
 	}
 
 	virtual void play() override
 	{
-		AUDIT(s_tls_source_voice != nullptr);
+		AUDIT(tls_source_voice != nullptr);
 
-		HRESULT hr = s_tls_source_voice->Start();
+		HRESULT hr = tls_source_voice->Start();
 		if (FAILED(hr))
 		{
 			LOG_ERROR(GENERAL, "XAudio2Backend : Start() failed(0x%08x)", (u32)hr);
@@ -85,9 +85,9 @@ public:
 
 	virtual void flush() override
 	{
-		AUDIT(s_tls_source_voice != nullptr);
+		AUDIT(tls_source_voice != nullptr);
 
-		HRESULT hr = s_tls_source_voice->FlushSourceBuffers();
+		HRESULT hr = tls_source_voice->FlushSourceBuffers();
 		if (FAILED(hr))
 		{
 			LOG_ERROR(GENERAL, "XAudio2Backend : FlushSourceBuffers() failed(0x%08x)", (u32)hr);
@@ -97,9 +97,9 @@ public:
 
 	virtual void stop() override
 	{
-		AUDIT(s_tls_source_voice != nullptr);
+		AUDIT(tls_source_voice != nullptr);
 
-		HRESULT hr = s_tls_source_voice->Stop();
+		HRESULT hr = tls_source_voice->Stop();
 		if (FAILED(hr))
 		{
 			LOG_ERROR(GENERAL, "XAudio2Backend : Stop() failed(0x%08x)", (u32)hr);
@@ -109,10 +109,10 @@ public:
 
 	virtual bool is_playing() override
 	{
-		AUDIT(s_tls_source_voice != nullptr);
+		AUDIT(tls_source_voice != nullptr);
 
 		XAUDIO2_VOICE_STATE state;
-		s_tls_source_voice->GetState(&state, XAUDIO2_VOICE_NOSAMPLESPLAYED);
+		tls_source_voice->GetState(&state, XAUDIO2_VOICE_NOSAMPLESPLAYED);
 
 		return state.BuffersQueued > 0 || state.pCurrentBufferContext != nullptr;
 	}
@@ -134,7 +134,7 @@ public:
 		waveformatex.wBitsPerSample = sample_size * 8;
 		waveformatex.cbSize = 0;
 
-		hr = s_tls_xaudio2_instance->CreateSourceVoice(&s_tls_source_voice, &waveformatex, 0, XAUDIO2_DEFAULT_FREQ_RATIO);
+		hr = tls_xaudio2_instance->CreateSourceVoice(&tls_source_voice, &waveformatex, 0, XAUDIO2_DEFAULT_FREQ_RATIO);
 		if (FAILED(hr))
 		{
 			LOG_ERROR(GENERAL, "XAudio2Backend : CreateSourceVoice() failed(0x%08x)", (u32)hr);
@@ -142,16 +142,16 @@ public:
 			return;
 		}
 
-		AUDIT(s_tls_source_voice != nullptr);
+		AUDIT(tls_source_voice != nullptr);
-		s_tls_source_voice->SetVolume(channels == 2 ? 1.0f : 4.0f);
+		tls_source_voice->SetVolume(channels == 2 ? 1.0f : 4.0f);
 	}
 
 	virtual bool add(const void* src, u32 num_samples) override
 	{
-		AUDIT(s_tls_source_voice != nullptr);
+		AUDIT(tls_source_voice != nullptr);
 
 		XAUDIO2_VOICE_STATE state;
-		s_tls_source_voice->GetState(&state, XAUDIO2_VOICE_NOSAMPLESPLAYED);
+		tls_source_voice->GetState(&state, XAUDIO2_VOICE_NOSAMPLESPLAYED);
 
 		if (state.BuffersQueued >= MAX_AUDIO_BUFFERS)
 		{
@@ -171,7 +171,7 @@ public:
 		buffer.PlayBegin = 0;
 		buffer.PlayLength = AUDIO_BUFFER_SAMPLES;
 
-		HRESULT hr = s_tls_source_voice->SubmitSourceBuffer(&buffer);
+		HRESULT hr = tls_source_voice->SubmitSourceBuffer(&buffer);
 		if (FAILED(hr))
 		{
 			LOG_ERROR(GENERAL, "XAudio2Backend : AddData() failed(0x%08x)", (u32)hr);
@@ -185,7 +185,7 @@ public:
 	virtual u64 enqueued_samples() override
 	{
 		XAUDIO2_VOICE_STATE state;
-		s_tls_source_voice->GetState(&state);
+		tls_source_voice->GetState(&state);
 
 		// all buffers contain AUDIO_BUFFER_SAMPLES, so we can easily calculate how many samples there are remaining
 		return (AUDIO_BUFFER_SAMPLES - state.SamplesPlayed % AUDIO_BUFFER_SAMPLES) + (state.BuffersQueued * AUDIO_BUFFER_SAMPLES);
@@ -195,7 +195,7 @@ public:
 	{
 		new_ratio = std::clamp(new_ratio, XAUDIO2_MIN_FREQ_RATIO, XAUDIO2_DEFAULT_FREQ_RATIO);
 
-		HRESULT hr = s_tls_source_voice->SetFrequencyRatio(new_ratio);
+		HRESULT hr = tls_source_voice->SetFrequencyRatio(new_ratio);
 		if (FAILED(hr))
 		{
 			LOG_ERROR(GENERAL, "XAudio2Backend : SetFrequencyRatio() failed(0x%08x)", (u32)hr);

rpcs3/Emu/Audio/XAudio2/XAudio2Backend.cpp

@@ -45,14 +45,12 @@ void XAudio2Backend::Open(u32 /* num_buffers */)
 		}
 		else if (hmodule = LoadLibraryExW(L"XAudio2_8.dll", nullptr, LOAD_LIBRARY_SEARCH_SYSTEM32))
 		{
-			// XAudio 2.9 uses the same code as XAudio 2.8
 			lib.reset(xa28_init(hmodule));
 
 			LOG_SUCCESS(GENERAL, "XAudio 2.8 initialized");
 		}
 		else if (hmodule = LoadLibraryExW(L"XAudio2_7.dll", nullptr, LOAD_LIBRARY_SEARCH_SYSTEM32))
 		{
-			// XAudio 2.9 uses the same code as XAudio 2.8
 			lib.reset(xa27_init(hmodule));
 
 			LOG_SUCCESS(GENERAL, "XAudio 2.7 initialized");