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Merge 7c12e2b5eb into 53b54406bd

This commit is contained in:
Tygyh 2025-05-27 21:43:06 +02:00 committed by GitHub
parent 53b54406bd 7c12e2b5eb
commit eeba1f8b0a
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GPG Key ID: B5690EEEBB952194
6 changed files with 34 additions and 28 deletions
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13
Source/Core/AudioCommon/AudioCommon.cpp
View File

@ -166,7 +166,8 @@ void UpdateSoundStream(Core::System& system)
if (sound_stream)
{
int volume = Config::Get(Config::MAIN_AUDIO_MUTED) ? 0 : Config::Get(Config::MAIN_AUDIO_VOLUME);
const int volume =
Config::Get(Config::MAIN_AUDIO_MUTED) ? 0 : Config::Get(Config::MAIN_AUDIO_VOLUME);
sound_stream->SetVolume(volume);
}
}
@ -192,7 +193,7 @@ void SetSoundStreamRunning(Core::System& system, bool running)
void SendAIBuffer(Core::System& system, const short* samples, unsigned int num_samples)
{
SoundStream* sound_stream = system.GetSoundStream();
const SoundStream* sound_stream = system.GetSoundStream();
if (!sound_stream)
return;
@ -212,9 +213,9 @@ void SendAIBuffer(Core::System& system, const short* samples, unsigned int num_s
void StartAudioDump(Core::System& system)
{
SoundStream* sound_stream = system.GetSoundStream();
const SoundStream* sound_stream = system.GetSoundStream();
std::time_t start_time = std::time(nullptr);
const std::time_t start_time = std::time(nullptr);
std::string path_prefix = File::GetUserPath(D_DUMPAUDIO_IDX) + SConfig::GetInstance().GetGameID();
@ -232,7 +233,7 @@ void StartAudioDump(Core::System& system)
void StopAudioDump(Core::System& system)
{
SoundStream* sound_stream = system.GetSoundStream();
const SoundStream* sound_stream = system.GetSoundStream();
if (!sound_stream)
return;
@ -265,7 +266,7 @@ void DecreaseVolume(Core::System& system, unsigned short offset)
void ToggleMuteVolume(Core::System& system)
{
bool isMuted = Config::Get(Config::MAIN_AUDIO_MUTED);
const bool isMuted = Config::Get(Config::MAIN_AUDIO_MUTED);
Config::SetBaseOrCurrent(Config::MAIN_AUDIO_MUTED, !isMuted);
UpdateSoundStream(system);
}

4
Source/Core/AudioCommon/CubebStream.cpp
View File

@ -23,7 +23,7 @@ constexpr u32 BUFFER_SAMPLES = 512;
long CubebStream::DataCallback(cubeb_stream* stream, void* user_data, const void* /*input_buffer*/,
void* output_buffer, long num_frames)
{
auto* self = static_cast<CubebStream*>(user_data);
const auto* self = static_cast<CubebStream*>(user_data);
if (self->m_stereo)
self->m_mixer->Mix(static_cast<short*>(output_buffer), num_frames);
@ -44,7 +44,7 @@ CubebStream::CubebStream()
Common::Event sync_event;
m_work_queue.Push([this, &sync_event] {
Common::ScopeGuard sync_event_guard([&sync_event] { sync_event.Set(); });
auto result = ::CoInitializeEx(nullptr, COINIT_MULTITHREADED | COINIT_DISABLE_OLE1DDE);
const auto result = CoInitializeEx(nullptr, COINIT_MULTITHREADED | COINIT_DISABLE_OLE1DDE);
m_coinit_success = result == S_OK;
m_should_couninit = result == S_OK || result == S_FALSE;
});

19
Source/Core/AudioCommon/Mixer.cpp
View File

@ -182,7 +182,8 @@ std::size_t Mixer::MixSurround(float* samples, std::size_t num_samples)
memset(samples, 0, num_samples * SURROUND_CHANNELS * sizeof(float));
std::size_t needed_frames = m_surround_decoder.QueryFramesNeededForSurroundOutput(num_samples);
const std::size_t needed_frames =
m_surround_decoder.QueryFramesNeededForSurroundOutput(num_samples);
constexpr std::size_t max_samples = 0x8000;
ASSERT_MSG(AUDIO, needed_frames <= max_samples,
@ -190,7 +191,7 @@ std::size_t Mixer::MixSurround(float* samples, std::size_t num_samples)
needed_frames, max_samples);
std::array<s16, max_samples> buffer;
std::size_t available_frames = Mix(buffer.data(), static_cast<std::size_t>(needed_frames));
const std::size_t available_frames = Mix(buffer.data(), static_cast<std::size_t>(needed_frames));
if (available_frames != needed_frames)
{
ERROR_LOG_FMT(AUDIO,
@ -229,7 +230,7 @@ void Mixer::PushSamples(const s16* samples, std::size_t num_samples)
if (m_log_dsp_audio)
{
const s32 sample_rate_divisor = m_dma_mixer.GetInputSampleRateDivisor();
auto volume = m_dma_mixer.GetVolume();
const auto volume = m_dma_mixer.GetVolume();
m_wave_writer_dsp.AddStereoSamplesBE(samples, static_cast<u32>(num_samples),
sample_rate_divisor, volume.first, volume.second);
}
@ -241,7 +242,7 @@ void Mixer::PushStreamingSamples(const s16* samples, std::size_t num_samples)
if (m_log_dtk_audio)
{
const s32 sample_rate_divisor = m_streaming_mixer.GetInputSampleRateDivisor();
auto volume = m_streaming_mixer.GetVolume();
const auto volume = m_streaming_mixer.GetVolume();
m_wave_writer_dtk.AddStereoSamplesBE(samples, static_cast<u32>(num_samples),
sample_rate_divisor, volume.first, volume.second);
}
@ -286,7 +287,8 @@ void Mixer::PushSkylanderPortalSamples(const u8* samples, std::size_t num_sample
{
for (std::size_t i = 0; i < num_samples; ++i)
{
s16 sample = static_cast<u16>(samples[i * 2 + 1]) << 8 | static_cast<u16>(samples[i * 2]);
const s16 sample =
static_cast<u16>(samples[i * 2 + 1]) << 8 | static_cast<u16>(samples[i * 2]);
samples_stereo[i * 2] = sample;
samples_stereo[i * 2 + 1] = sample;
}
@ -335,7 +337,8 @@ void Mixer::StartLogDTKAudio(const std::string& filename)
{
if (!m_log_dtk_audio)
{
bool success = m_wave_writer_dtk.Start(filename, m_streaming_mixer.GetInputSampleRateDivisor());
const bool success =
m_wave_writer_dtk.Start(filename, m_streaming_mixer.GetInputSampleRateDivisor());
if (success)
{
m_log_dtk_audio = true;
@ -372,7 +375,7 @@ void Mixer::StartLogDSPAudio(const std::string& filename)
{
if (!m_log_dsp_audio)
{
bool success = m_wave_writer_dsp.Start(filename, m_dma_mixer.GetInputSampleRateDivisor());
const bool success = m_wave_writer_dsp.Start(filename, m_dma_mixer.GetInputSampleRateDivisor());
if (success)
{
m_log_dsp_audio = true;
@ -497,7 +500,7 @@ void Mixer::MixerFifo::Enqueue()
const std::size_t head = m_queue_head.load(std::memory_order_acquire);
// Check if we run out of space in the circular queue. (rare)
std::size_t next_head = (head + 1) & GRANULE_QUEUE_MASK;
const std::size_t next_head = (head + 1) & GRANULE_QUEUE_MASK;
if (next_head == m_queue_tail.load(std::memory_order_acquire))
{
WARN_LOG_FMT(AUDIO,

19
Source/Core/AudioCommon/OpenALStream.cpp
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@ -76,7 +76,7 @@ static bool InitLibrary()
if (!InitFunctions())
{
::FreeLibrary(s_openal_dll);
FreeLibrary(s_openal_dll);
s_openal_dll = nullptr;
return false;
}
@ -168,7 +168,7 @@ bool OpenALStream::SetRunning(bool running)
static ALenum CheckALError(const char* desc)
{
ALenum err = palGetError();
const ALenum err = palGetError();
if (err != AL_NO_ERROR)
{
@ -211,16 +211,16 @@ void OpenALStream::SoundLoop()
{
Common::SetCurrentThreadName("Audio thread - openal");
bool float32_capable = palIsExtensionPresent("AL_EXT_float32") != 0;
bool surround_capable = palIsExtensionPresent("AL_EXT_MCFORMATS") || IsCreativeXFi();
const bool float32_capable = palIsExtensionPresent("AL_EXT_float32") != 0;
const bool surround_capable = palIsExtensionPresent("AL_EXT_MCFORMATS") || IsCreativeXFi();
bool use_surround = Config::ShouldUseDPL2Decoder() && surround_capable;
// As there is no extension to check for 32-bit fixed point support
// and we know that only a X-Fi with hardware OpenAL supports it,
// we just check if one is being used.
bool fixed32_capable = IsCreativeXFi();
const bool fixed32_capable = IsCreativeXFi();
u32 frequency = m_mixer->GetSampleRate();
const u32 frequency = m_mixer->GetSampleRate();
u32 frames_per_buffer;
// Can't have zero samples per buffer
@ -288,12 +288,12 @@ void OpenALStream::SoundLoop()
num_buffers_queued -= num_buffers_processed;
}
unsigned int min_frames = frames_per_buffer;
const unsigned int min_frames = frames_per_buffer;
if (use_surround)
{
std::array<float, OAL_MAX_FRAMES * SURROUND_CHANNELS> dpl2;
u32 rendered_frames = static_cast<u32>(m_mixer->MixSurround(dpl2.data(), min_frames));
const u32 rendered_frames = static_cast<u32>(m_mixer->MixSurround(dpl2.data(), min_frames));
if (rendered_frames < min_frames)
continue;
@ -351,7 +351,8 @@ void OpenALStream::SoundLoop()
}
else
{
u32 rendered_frames = static_cast<u32>(m_mixer->Mix(m_realtime_buffer.data(), min_frames));
const u32 rendered_frames =
static_cast<u32>(m_mixer->Mix(m_realtime_buffer.data(), min_frames));
if (!rendered_frames)
continue;

2
Source/Core/AudioCommon/SurroundDecoder.cpp
View File

@ -32,7 +32,7 @@ size_t SurroundDecoder::QueryFramesNeededForSurroundOutput(const size_t output_f
if (m_decoded_fifo.size() < output_frames * SURROUND_CHANNELS)
{
// Output stereo frames needed to have at least the desired number of surround frames
size_t frames_needed = output_frames - m_decoded_fifo.size() / SURROUND_CHANNELS;
const size_t frames_needed = output_frames - m_decoded_fifo.size() / SURROUND_CHANNELS;
return frames_needed + m_frame_block_size - frames_needed % m_frame_block_size;
}

5
Source/Core/AudioCommon/WASAPIStream.cpp
View File

@ -159,8 +159,9 @@ ComPtr<IMMDevice> WASAPIStream::GetDeviceByName(std::string_view name)
bool WASAPIStream::Init()
{
ASSERT(m_enumerator == nullptr);
HRESULT result = CoCreateInstance(__uuidof(MMDeviceEnumerator), nullptr, CLSCTX_INPROC_SERVER,
IID_PPV_ARGS(m_enumerator.GetAddressOf()));
const HRESULT result =
CoCreateInstance(__uuidof(MMDeviceEnumerator), nullptr, CLSCTX_INPROC_SERVER,
IID_PPV_ARGS(m_enumerator.GetAddressOf()));
if (!HandleWinAPI("Failed to create MMDeviceEnumerator", result))
return false;