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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;
}
const char* defaultDeviceName = palcGetString(nullptr, ALC_DEFAULT_DEVICE_SPECIFIER);
INFO_LOG(AUDIO, "Found OpenAL device %s", defaultDeviceName);
const char* default_device_dame = palcGetString(nullptr, ALC_DEFAULT_DEVICE_SPECIFIER);
INFO_LOG(AUDIO, "Found OpenAL device %s", default_device_dame);
ALCdevice* pDevice = palcOpenDevice(defaultDeviceName);
if (!pDevice)
ALCdevice* device = palcOpenDevice(default_device_dame);
if (!device)
{
PanicAlertT("OpenAL: can't open device %s", defaultDeviceName);
PanicAlertT("OpenAL: can't open device %s", default_device_dame);
return false;
}
ALCcontext* pContext = palcCreateContext(pDevice, nullptr);
if (!pContext)
ALCcontext* context = palcCreateContext(device, nullptr);
if (!context)
{
palcCloseDevice(pDevice);
PanicAlertT("OpenAL: can't create context for device %s", defaultDeviceName);
palcCloseDevice(device);
PanicAlertT("OpenAL: can't create context for device %s", default_device_dame);
return false;
}
palcMakeContextCurrent(pContext);
palcMakeContextCurrent(context);
m_run_thread.Set();
thread = std::thread(&OpenALStream::SoundLoop, this);
m_thread = std::thread(&OpenALStream::SoundLoop, this);
return true;
}
@ -128,37 +128,37 @@ void OpenALStream::Stop()
{
m_run_thread.Clear();
// kick the thread if it's waiting
soundSyncEvent.Set();
m_sound_sync_event.Set();
thread.join();
m_thread.join();
palSourceStop(uiSource);
palSourcei(uiSource, AL_BUFFER, 0);
palSourceStop(m_source);
palSourcei(m_source, AL_BUFFER, 0);
// Clean up buffers and sources
palDeleteSources(1, &uiSource);
uiSource = 0;
palDeleteBuffers(OAL_BUFFERS, uiBuffers.data());
palDeleteSources(1, &m_source);
m_source = 0;
palDeleteBuffers(OAL_BUFFERS, m_buffers.data());
ALCcontext* pContext = palcGetCurrentContext();
ALCdevice* pDevice = palcGetContextsDevice(pContext);
ALCcontext* context = palcGetCurrentContext();
ALCdevice* device = palcGetContextsDevice(context);
palcMakeContextCurrent(nullptr);
palcDestroyContext(pContext);
palcCloseDevice(pDevice);
palcDestroyContext(context);
palcCloseDevice(device);
}
void OpenALStream::SetVolume(int volume)
{
fVolume = (float)volume / 100.0f;
m_volume = (float)volume / 100.0f;
if (uiSource)
palSourcef(uiSource, AL_GAIN, fVolume);
if (m_source)
palSourcef(m_source, AL_GAIN, m_volume);
}
void OpenALStream::Update()
{
soundSyncEvent.Set();
m_sound_sync_event.Set();
}
void OpenALStream::Clear(bool mute)
@ -167,11 +167,11 @@ void OpenALStream::Clear(bool mute)
if (m_muted)
{
palSourceStop(uiSource);
palSourceStop(m_source);
}
else
{
palSourcePlay(uiSource);
palSourcePlay(m_source);
}
}
@ -229,17 +229,17 @@ void OpenALStream::SoundLoop()
// we just check if one is being used.
bool fixed32_capable = IsCreativeXFi();
u32 ulFrequency = m_mixer->GetSampleRate();
u32 frequency = m_mixer->GetSampleRate();
u32 frames_per_buffer;
// Can't have zero samples per buffer
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
{
frames_per_buffer = ulFrequency / 1000 * 1 / OAL_BUFFERS;
frames_per_buffer = frequency / 1000 * 1 / OAL_BUFFERS;
}
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
// than what we request?
realtimeBuffer.resize(frames_per_buffer * STEREO_CHANNELS);
sampleBuffer.resize(frames_per_buffer * STEREO_CHANNELS);
uiSource = 0;
m_realtime_buffer.resize(frames_per_buffer * STEREO_CHANNELS);
m_sample_buffer.resize(frames_per_buffer * STEREO_CHANNELS);
m_source = 0;
// Clear error state before querying or else we get false positives.
ALenum err = palGetError();
// Generate some AL Buffers for streaming
palGenBuffers(OAL_BUFFERS, (ALuint*)uiBuffers.data());
palGenBuffers(OAL_BUFFERS, (ALuint*)m_buffers.data());
err = CheckALError("generating buffers");
// Generate a Source to playback the Buffers
palGenSources(1, &uiSource);
palGenSources(1, &m_source);
err = CheckALError("generating sources");
// 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
// ALenum err = alGetError();
unsigned int nextBuffer = 0;
unsigned int numBuffersQueued = 0;
ALint iState = 0;
unsigned int next_buffer = 0;
unsigned int num_buffers_queued = 0;
ALint state = 0;
while (m_run_thread.IsSet())
{
// Block until we have a free buffer
int numBuffersProcessed;
palGetSourcei(uiSource, AL_BUFFERS_PROCESSED, &numBuffersProcessed);
if (OAL_BUFFERS == numBuffersQueued && !numBuffersProcessed)
int num_buffers_processed;
palGetSourcei(m_source, AL_BUFFERS_PROCESSED, &num_buffers_processed);
if (num_buffers_queued == OAL_BUFFERS && !num_buffers_processed)
{
soundSyncEvent.Wait();
m_sound_sync_event.Wait();
continue;
}
// Remove the Buffer from the Queue.
if (numBuffersProcessed)
if (num_buffers_processed)
{
ALuint unqueuedBufferIds[OAL_BUFFERS];
palSourceUnqueueBuffers(uiSource, numBuffersProcessed, unqueuedBufferIds);
ALuint unqueued_buffer_ids[OAL_BUFFERS];
palSourceUnqueueBuffers(m_source, num_buffers_processed, unqueued_buffer_ids);
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)
{
// DPL2 accepts 240 samples minimum (FWRDURATION)
unsigned int minSamples = 240;
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;
// 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
// AL_FORMAT_50CHN32 to make this super-explicit.
// 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;
}
if (float32_capable)
{
palBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN32, dpl2,
numSamples * FRAME_SURROUND_FLOAT, ulFrequency);
palBufferData(m_buffers[next_buffer], AL_FORMAT_51CHN32, dpl2,
rendered_frames * FRAME_SURROUND_FLOAT, frequency);
}
else if (fixed32_capable)
{
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.
// 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];
}
palBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN32, surround_int32,
numSamples * FRAME_SURROUND_INT32, ulFrequency);
palBufferData(m_buffers[next_buffer], AL_FORMAT_51CHN32, surround_int32,
rendered_frames * FRAME_SURROUND_INT32, frequency);
}
else
{
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);
if (dpl2[i] > SHRT_MAX)
@ -367,8 +364,8 @@ void OpenALStream::SoundLoop()
surround_short[i] = (int)dpl2[i];
}
palBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN16, surround_short,
numSamples * FRAME_SURROUND_SHORT, ulFrequency);
palBufferData(m_buffers[next_buffer], AL_FORMAT_51CHN16, surround_short,
rendered_frames * FRAME_SURROUND_SHORT, frequency);
}
err = CheckALError("buffering data");
@ -382,19 +379,19 @@ void OpenALStream::SoundLoop()
}
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
for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i)
sampleBuffer[i] = static_cast<float>(realtimeBuffer[i]) / (1 << 15);
for (u32 i = 0; i < rendered_frames * STEREO_CHANNELS; ++i)
m_sample_buffer[i] = static_cast<float>(m_realtime_buffer[i]) / (1 << 15);
if (!numSamples)
if (!rendered_frames)
continue;
if (float32_capable)
{
palBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO_FLOAT32, sampleBuffer.data(),
numSamples * FRAME_STEREO_FLOAT, ulFrequency);
palBufferData(m_buffers[next_buffer], AL_FORMAT_STEREO_FLOAT32, m_sample_buffer.data(),
rendered_frames * FRAME_STEREO_FLOAT, frequency);
err = CheckALError("buffering float32 data");
if (err == AL_INVALID_ENUM)
@ -406,35 +403,35 @@ void OpenALStream::SoundLoop()
{
// Clamping is not necessary here, samples are always between (-1,1)
int stereo_int32[OAL_MAX_FRAMES * STEREO_CHANNELS];
for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i)
stereo_int32[i] = (int)((float)sampleBuffer[i] * (INT64_C(1) << 31));
for (u32 i = 0; i < rendered_frames * STEREO_CHANNELS; ++i)
stereo_int32[i] = (int)((float)m_sample_buffer[i] * (INT64_C(1) << 31));
palBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO32, stereo_int32,
numSamples * FRAME_STEREO_INT32, ulFrequency);
palBufferData(m_buffers[next_buffer], AL_FORMAT_STEREO32, stereo_int32,
rendered_frames * FRAME_STEREO_INT32, frequency);
}
else
{
// Convert the samples from float to short
short stereo[OAL_MAX_FRAMES * STEREO_CHANNELS];
for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i)
stereo[i] = (short)((float)sampleBuffer[i] * (1 << 15));
for (u32 i = 0; i < rendered_frames * STEREO_CHANNELS; ++i)
stereo[i] = (short)((float)m_sample_buffer[i] * (1 << 15));
palBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO16, stereo,
numSamples * FRAME_STEREO_SHORT, ulFrequency);
palBufferData(m_buffers[next_buffer], AL_FORMAT_STEREO16, stereo,
rendered_frames * FRAME_STEREO_SHORT, frequency);
}
}
palSourceQueueBuffers(uiSource, 1, &uiBuffers[nextBuffer]);
palSourceQueueBuffers(m_source, 1, &m_buffers[next_buffer]);
err = CheckALError("queuing buffers");
numBuffersQueued++;
nextBuffer = (nextBuffer + 1) % OAL_BUFFERS;
num_buffers_queued++;
next_buffer = (next_buffer + 1) % OAL_BUFFERS;
palGetSourcei(uiSource, AL_SOURCE_STATE, &iState);
if (iState != AL_PLAYING)
palGetSourcei(m_source, AL_SOURCE_STATE, &state);
if (state != AL_PLAYING)
{
// Buffer underrun occurred, resume playback
palSourcePlay(uiSource);
palSourcePlay(m_source);
err = CheckALError("occurred resuming playback");
}
}

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

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