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Renamed some variables to the current coding standard and some to better 

fit what they really are.
This commit is contained in:
lfsafady 2017-06-17 17:16:32 -03:00
parent 8fd1af6783
commit d9d51fe0c4
2 changed files with 88 additions and 91 deletions
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163
Source/Core/AudioCommon/OpenALStream.cpp
View File

@ -100,27 +100,27 @@ bool OpenALStream::Start()
return false; return false;
} }
const char* defaultDeviceName = palcGetString(nullptr, ALC_DEFAULT_DEVICE_SPECIFIER); const char* default_device_dame = palcGetString(nullptr, ALC_DEFAULT_DEVICE_SPECIFIER);
INFO_LOG(AUDIO, "Found OpenAL device %s", defaultDeviceName); INFO_LOG(AUDIO, "Found OpenAL device %s", default_device_dame);
ALCdevice* pDevice = palcOpenDevice(defaultDeviceName); ALCdevice* device = palcOpenDevice(default_device_dame);
if (!pDevice) if (!device)
{ {
PanicAlertT("OpenAL: can't open device %s", defaultDeviceName); PanicAlertT("OpenAL: can't open device %s", default_device_dame);
return false; return false;
} }
ALCcontext* pContext = palcCreateContext(pDevice, nullptr); ALCcontext* context = palcCreateContext(device, nullptr);
if (!pContext) if (!context)
{ {
palcCloseDevice(pDevice); palcCloseDevice(device);
PanicAlertT("OpenAL: can't create context for device %s", defaultDeviceName); PanicAlertT("OpenAL: can't create context for device %s", default_device_dame);
return false; return false;
} }
palcMakeContextCurrent(pContext); palcMakeContextCurrent(context);
m_run_thread.Set(); m_run_thread.Set();
thread = std::thread(&OpenALStream::SoundLoop, this); m_thread = std::thread(&OpenALStream::SoundLoop, this);
return true; return true;
} }
@ -128,37 +128,37 @@ void OpenALStream::Stop()
{ {
m_run_thread.Clear(); m_run_thread.Clear();
// kick the thread if it's waiting // kick the thread if it's waiting
soundSyncEvent.Set(); m_sound_sync_event.Set();
thread.join(); m_thread.join();
palSourceStop(uiSource); palSourceStop(m_source);
palSourcei(uiSource, AL_BUFFER, 0); palSourcei(m_source, AL_BUFFER, 0);
// Clean up buffers and sources // Clean up buffers and sources
palDeleteSources(1, &uiSource); palDeleteSources(1, &m_source);
uiSource = 0; m_source = 0;
palDeleteBuffers(OAL_BUFFERS, uiBuffers.data()); palDeleteBuffers(OAL_BUFFERS, m_buffers.data());
ALCcontext* pContext = palcGetCurrentContext(); ALCcontext* context = palcGetCurrentContext();
ALCdevice* pDevice = palcGetContextsDevice(pContext); ALCdevice* device = palcGetContextsDevice(context);
palcMakeContextCurrent(nullptr); palcMakeContextCurrent(nullptr);
palcDestroyContext(pContext); palcDestroyContext(context);
palcCloseDevice(pDevice); palcCloseDevice(device);
} }
void OpenALStream::SetVolume(int volume) void OpenALStream::SetVolume(int volume)
{ {
fVolume = (float)volume / 100.0f; m_volume = (float)volume / 100.0f;
if (uiSource) if (m_source)
palSourcef(uiSource, AL_GAIN, fVolume); palSourcef(m_source, AL_GAIN, m_volume);
} }
void OpenALStream::Update() void OpenALStream::Update()
{ {
soundSyncEvent.Set(); m_sound_sync_event.Set();
} }
void OpenALStream::Clear(bool mute) void OpenALStream::Clear(bool mute)
@ -167,11 +167,11 @@ void OpenALStream::Clear(bool mute)
if (m_muted) if (m_muted)
{ {
palSourceStop(uiSource); palSourceStop(m_source);
} }
else else
{ {
palSourcePlay(uiSource); palSourcePlay(m_source);
} }
} }
@ -229,17 +229,17 @@ void OpenALStream::SoundLoop()
// we just check if one is being used. // we just check if one is being used.
bool fixed32_capable = IsCreativeXFi(); bool fixed32_capable = IsCreativeXFi();
u32 ulFrequency = m_mixer->GetSampleRate(); u32 frequency = m_mixer->GetSampleRate();
u32 frames_per_buffer; u32 frames_per_buffer;
// Can't have zero samples per buffer // Can't have zero samples per buffer
if (SConfig::GetInstance().iLatency > 0) if (SConfig::GetInstance().iLatency > 0)
{ {
frames_per_buffer = ulFrequency / 1000 * SConfig::GetInstance().iLatency / OAL_BUFFERS; frames_per_buffer = frequency / 1000 * SConfig::GetInstance().iLatency / OAL_BUFFERS;
} }
else else
{ {
frames_per_buffer = ulFrequency / 1000 * 1 / OAL_BUFFERS; frames_per_buffer = frequency / 1000 * 1 / OAL_BUFFERS;
} }
if (frames_per_buffer > OAL_MAX_FRAMES) if (frames_per_buffer > OAL_MAX_FRAMES)
@ -258,84 +258,81 @@ void OpenALStream::SoundLoop()
// Should we make these larger just in case the mixer ever sends more samples // Should we make these larger just in case the mixer ever sends more samples
// than what we request? // than what we request?
realtimeBuffer.resize(frames_per_buffer * STEREO_CHANNELS); m_realtime_buffer.resize(frames_per_buffer * STEREO_CHANNELS);
sampleBuffer.resize(frames_per_buffer * STEREO_CHANNELS); m_sample_buffer.resize(frames_per_buffer * STEREO_CHANNELS);
uiSource = 0; m_source = 0;
// Clear error state before querying or else we get false positives. // Clear error state before querying or else we get false positives.
ALenum err = palGetError(); ALenum err = palGetError();
// Generate some AL Buffers for streaming // Generate some AL Buffers for streaming
palGenBuffers(OAL_BUFFERS, (ALuint*)uiBuffers.data()); palGenBuffers(OAL_BUFFERS, (ALuint*)m_buffers.data());
err = CheckALError("generating buffers"); err = CheckALError("generating buffers");
// Generate a Source to playback the Buffers // Generate a Source to playback the Buffers
palGenSources(1, &uiSource); palGenSources(1, &m_source);
err = CheckALError("generating sources"); err = CheckALError("generating sources");
// Set the default sound volume as saved in the config file. // Set the default sound volume as saved in the config file.
palSourcef(uiSource, AL_GAIN, fVolume); palSourcef(m_source, AL_GAIN, m_volume);
// TODO: Error handling // TODO: Error handling
// ALenum err = alGetError(); // ALenum err = alGetError();
unsigned int nextBuffer = 0; unsigned int next_buffer = 0;
unsigned int numBuffersQueued = 0; unsigned int num_buffers_queued = 0;
ALint iState = 0; ALint state = 0;
while (m_run_thread.IsSet()) while (m_run_thread.IsSet())
{ {
// Block until we have a free buffer // Block until we have a free buffer
int numBuffersProcessed; int num_buffers_processed;
palGetSourcei(uiSource, AL_BUFFERS_PROCESSED, &numBuffersProcessed); palGetSourcei(m_source, AL_BUFFERS_PROCESSED, &num_buffers_processed);
if (OAL_BUFFERS == numBuffersQueued && !numBuffersProcessed) if (num_buffers_queued == OAL_BUFFERS && !num_buffers_processed)
{ {
soundSyncEvent.Wait(); m_sound_sync_event.Wait();
continue; continue;
} }
// Remove the Buffer from the Queue. // Remove the Buffer from the Queue.
if (numBuffersProcessed) if (num_buffers_processed)
{ {
ALuint unqueuedBufferIds[OAL_BUFFERS]; ALuint unqueued_buffer_ids[OAL_BUFFERS];
palSourceUnqueueBuffers(uiSource, numBuffersProcessed, unqueuedBufferIds); palSourceUnqueueBuffers(m_source, num_buffers_processed, unqueued_buffer_ids);
err = CheckALError("unqueuing buffers"); err = CheckALError("unqueuing buffers");
numBuffersQueued -= numBuffersProcessed; num_buffers_queued -= num_buffers_processed;
} }
unsigned int numSamples = frames_per_buffer; unsigned int min_frames = frames_per_buffer;
if (use_surround) if (use_surround)
{ {
// DPL2 accepts 240 samples minimum (FWRDURATION)
unsigned int minSamples = 240;
float dpl2[OAL_MAX_FRAMES * SURROUND_CHANNELS]; float dpl2[OAL_MAX_FRAMES * SURROUND_CHANNELS];
numSamples = m_mixer->MixSurround(dpl2, numSamples); u32 rendered_frames = m_mixer->MixSurround(dpl2, min_frames);
if (numSamples < minSamples) if (rendered_frames < min_frames)
continue; continue;
// zero-out the subwoofer channel - DPL2Decode generates a pretty // zero-out the subwoofer channel - DPL2Decode generates a pretty
// good 5.0 but not a good 5.1 output. Sadly there is not a 5.0 // good 5.0 but not a good 5.1 output. Sadly there is not a 5.0
// AL_FORMAT_50CHN32 to make this super-explicit. // AL_FORMAT_50CHN32 to make this super-explicit.
// DPL2Decode output: LEFTFRONT, RIGHTFRONT, CENTREFRONT, (sub), LEFTREAR, RIGHTREAR // DPL2Decode output: LEFTFRONT, RIGHTFRONT, CENTREFRONT, (sub), LEFTREAR, RIGHTREAR
for (u32 i = 0; i < numSamples; ++i) for (u32 i = 0; i < rendered_frames; ++i)
{ {
dpl2[i * SURROUND_CHANNELS + 3 /*sub/lfe*/] = 0.0f; dpl2[i * SURROUND_CHANNELS + 3 /*sub/lfe*/] = 0.0f;
} }
if (float32_capable) if (float32_capable)
{ {
palBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN32, dpl2, palBufferData(m_buffers[next_buffer], AL_FORMAT_51CHN32, dpl2,
numSamples * FRAME_SURROUND_FLOAT, ulFrequency); rendered_frames * FRAME_SURROUND_FLOAT, frequency);
} }
else if (fixed32_capable) else if (fixed32_capable)
{ {
int surround_int32[OAL_MAX_FRAMES * SURROUND_CHANNELS]; int surround_int32[OAL_MAX_FRAMES * SURROUND_CHANNELS];
for (u32 i = 0; i < numSamples * SURROUND_CHANNELS; ++i) for (u32 i = 0; i < rendered_frames * SURROUND_CHANNELS; ++i)
{ {
// For some reason the ffdshow's DPL2 decoder outputs samples bigger than 1. // For some reason the ffdshow's DPL2 decoder outputs samples bigger than 1.
// Most are close to 2.5 and some go up to 8. Hard clamping here, we need to // Most are close to 2.5 and some go up to 8. Hard clamping here, we need to
@ -349,14 +346,14 @@ void OpenALStream::SoundLoop()
surround_int32[i] = (int)dpl2[i]; surround_int32[i] = (int)dpl2[i];
} }
palBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN32, surround_int32, palBufferData(m_buffers[next_buffer], AL_FORMAT_51CHN32, surround_int32,
numSamples * FRAME_SURROUND_INT32, ulFrequency); rendered_frames * FRAME_SURROUND_INT32, frequency);
} }
else else
{ {
short surround_short[OAL_MAX_FRAMES * SURROUND_CHANNELS]; short surround_short[OAL_MAX_FRAMES * SURROUND_CHANNELS];
for (u32 i = 0; i < numSamples * SURROUND_CHANNELS; ++i) for (u32 i = 0; i < rendered_frames * SURROUND_CHANNELS; ++i)
{ {
dpl2[i] = dpl2[i] * (1 << 15); dpl2[i] = dpl2[i] * (1 << 15);
if (dpl2[i] > SHRT_MAX) if (dpl2[i] > SHRT_MAX)
@ -367,8 +364,8 @@ void OpenALStream::SoundLoop()
surround_short[i] = (int)dpl2[i]; surround_short[i] = (int)dpl2[i];
} }
palBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN16, surround_short, palBufferData(m_buffers[next_buffer], AL_FORMAT_51CHN16, surround_short,
numSamples * FRAME_SURROUND_SHORT, ulFrequency); rendered_frames * FRAME_SURROUND_SHORT, frequency);
} }
err = CheckALError("buffering data"); err = CheckALError("buffering data");
@ -382,19 +379,19 @@ void OpenALStream::SoundLoop()
} }
else else
{ {
numSamples = m_mixer->Mix(realtimeBuffer.data(), numSamples); u32 rendered_frames = m_mixer->Mix(m_realtime_buffer.data(), min_frames);
// Convert the samples from short to float // Convert the samples from short to float
for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i) for (u32 i = 0; i < rendered_frames * STEREO_CHANNELS; ++i)
sampleBuffer[i] = static_cast<float>(realtimeBuffer[i]) / (1 << 15); m_sample_buffer[i] = static_cast<float>(m_realtime_buffer[i]) / (1 << 15);
if (!numSamples) if (!rendered_frames)
continue; continue;
if (float32_capable) if (float32_capable)
{ {
palBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO_FLOAT32, sampleBuffer.data(), palBufferData(m_buffers[next_buffer], AL_FORMAT_STEREO_FLOAT32, m_sample_buffer.data(),
numSamples * FRAME_STEREO_FLOAT, ulFrequency); rendered_frames * FRAME_STEREO_FLOAT, frequency);
err = CheckALError("buffering float32 data"); err = CheckALError("buffering float32 data");
if (err == AL_INVALID_ENUM) if (err == AL_INVALID_ENUM)
@ -406,35 +403,35 @@ void OpenALStream::SoundLoop()
{ {
// Clamping is not necessary here, samples are always between (-1,1) // Clamping is not necessary here, samples are always between (-1,1)
int stereo_int32[OAL_MAX_FRAMES * STEREO_CHANNELS]; int stereo_int32[OAL_MAX_FRAMES * STEREO_CHANNELS];
for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i) for (u32 i = 0; i < rendered_frames * STEREO_CHANNELS; ++i)
stereo_int32[i] = (int)((float)sampleBuffer[i] * (INT64_C(1) << 31)); stereo_int32[i] = (int)((float)m_sample_buffer[i] * (INT64_C(1) << 31));
palBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO32, stereo_int32, palBufferData(m_buffers[next_buffer], AL_FORMAT_STEREO32, stereo_int32,
numSamples * FRAME_STEREO_INT32, ulFrequency); rendered_frames * FRAME_STEREO_INT32, frequency);
} }
else else
{ {
// Convert the samples from float to short // Convert the samples from float to short
short stereo[OAL_MAX_FRAMES * STEREO_CHANNELS]; short stereo[OAL_MAX_FRAMES * STEREO_CHANNELS];
for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i) for (u32 i = 0; i < rendered_frames * STEREO_CHANNELS; ++i)
stereo[i] = (short)((float)sampleBuffer[i] * (1 << 15)); stereo[i] = (short)((float)m_sample_buffer[i] * (1 << 15));
palBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO16, stereo, palBufferData(m_buffers[next_buffer], AL_FORMAT_STEREO16, stereo,
numSamples * FRAME_STEREO_SHORT, ulFrequency); rendered_frames * FRAME_STEREO_SHORT, frequency);
} }
} }
palSourceQueueBuffers(uiSource, 1, &uiBuffers[nextBuffer]); palSourceQueueBuffers(m_source, 1, &m_buffers[next_buffer]);
err = CheckALError("queuing buffers"); err = CheckALError("queuing buffers");
numBuffersQueued++; num_buffers_queued++;
nextBuffer = (nextBuffer + 1) % OAL_BUFFERS; next_buffer = (next_buffer + 1) % OAL_BUFFERS;
palGetSourcei(uiSource, AL_SOURCE_STATE, &iState); palGetSourcei(m_source, AL_SOURCE_STATE, &state);
if (iState != AL_PLAYING) if (state != AL_PLAYING)
{ {
// Buffer underrun occurred, resume playback // Buffer underrun occurred, resume playback
palSourcePlay(uiSource); palSourcePlay(m_source);
err = CheckALError("occurred resuming playback"); err = CheckALError("occurred resuming playback");
} }
} }

16
Source/Core/AudioCommon/OpenALStream.h
View File

@ -56,7 +56,7 @@ class OpenALStream final : public SoundStream
{ {
#ifdef _WIN32 #ifdef _WIN32
public: public:
OpenALStream() : uiSource(0) {} OpenALStream() : m_source(0) {}
bool Start() override; bool Start() override;
void SoundLoop() override; void SoundLoop() override;
void SetVolume(int volume) override; void SetVolume(int volume) override;
@ -67,16 +67,16 @@ public:
static bool isValid(); static bool isValid();
private: private:
std::thread thread; std::thread m_thread;
Common::Flag m_run_thread; Common::Flag m_run_thread;
Common::Event soundSyncEvent; Common::Event m_sound_sync_event;
std::vector<short> realtimeBuffer; std::vector<short> m_realtime_buffer;
std::vector<float> sampleBuffer; std::vector<float> m_sample_buffer;
std::array<ALuint, OAL_BUFFERS> uiBuffers; std::array<ALuint, OAL_BUFFERS> m_buffers;
ALuint uiSource; ALuint m_source;
ALfloat fVolume; ALfloat m_volume;
#endif // _WIN32 #endif // _WIN32
}; };