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Clean up more space/tab mismatches in AudioCommon, Common, and VideoCommon.

Browse Source 
Not planning to touch Core since it's the most actively changed part of the project.
This commit is contained in:
lioncash
2013-03-19 21:51:12 -04:00
parent 0e3d8e2e9f
commit edd9d0e0ef
91 changed files with 2151 additions and 2169 deletions
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6
Source/Core/AudioCommon/Src/AOSoundStream.cpp
View File

@ -34,7 +34,7 @@ void AOSound::SoundLoop()
format.channels = 2;
format.rate = m_mixer->GetSampleRate();
format.byte_format = AO_FMT_LITTLE;
device = ao_open_live(default_driver, &format, NULL /* no options */);
if (!device)
{
@ -49,7 +49,7 @@ void AOSound::SoundLoop()
while (!threadData)
{
m_mixer->Mix(realtimeBuffer, numBytesToRender >> 2);
{
std::lock_guard<std::mutex> lk(soundCriticalSection);
ao_play(device, (char*)realtimeBuffer, numBytesToRender);
@ -62,7 +62,7 @@ void AOSound::SoundLoop()
bool AOSound::Start()
{
memset(realtimeBuffer, 0, sizeof(realtimeBuffer));
thread = std::thread(std::mem_fun(&AOSound::SoundLoop), this);
return true;
}

12
Source/Core/AudioCommon/Src/AOSoundStream.h
View File

@ -45,21 +45,21 @@ public:
AOSound(CMixer *mixer) : SoundStream(mixer) {}
virtual ~AOSound();
virtual bool Start();
virtual void SoundLoop();
virtual void Stop();
static bool isValid() {
return true;
}
virtual bool usesMixer() const {
return true;
}
virtual void Update();
#else

32
Source/Core/AudioCommon/Src/AlsaSoundStream.cpp
View File

@ -88,7 +88,7 @@ bool AlsaSound::AlsaInit()
snd_pcm_hw_params_t *hwparams;
snd_pcm_uframes_t buffer_size,buffer_size_max;
unsigned int periods;
err = snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0);
if (err < 0)
{
@ -97,7 +97,7 @@ bool AlsaSound::AlsaInit()
}
snd_pcm_hw_params_alloca(&hwparams);
err = snd_pcm_hw_params_any(handle, hwparams);
if (err < 0)
{
@ -111,8 +111,8 @@ bool AlsaSound::AlsaInit()
ERROR_LOG(AUDIO, "Access type not available: %s\n", snd_strerror(err));
return false;
}
err = snd_pcm_hw_params_set_format(handle, hwparams, SND_PCM_FORMAT_S16_LE);
err = snd_pcm_hw_params_set_format(handle, hwparams, SND_PCM_FORMAT_S16_LE);
if (err < 0)
{
ERROR_LOG(AUDIO, "Sample format not available: %s\n", snd_strerror(err));
@ -126,14 +126,14 @@ bool AlsaSound::AlsaInit()
ERROR_LOG(AUDIO, "Rate not available: %s\n", snd_strerror(err));
return false;
}
err = snd_pcm_hw_params_set_channels(handle, hwparams, 2);
err = snd_pcm_hw_params_set_channels(handle, hwparams, 2);
if (err < 0)
{
ERROR_LOG(AUDIO, "Channels count not available: %s\n", snd_strerror(err));
return false;
}
periods = BUFFER_SIZE_MAX / FRAME_COUNT_MIN;
err = snd_pcm_hw_params_set_periods_max(handle, hwparams, &periods, &dir);
if (err < 0)
@ -153,10 +153,10 @@ bool AlsaSound::AlsaInit()
err = snd_pcm_hw_params(handle, hwparams);
if (err < 0)
{
ERROR_LOG(AUDIO, "Unable to install hw params: %s\n", snd_strerror(err));
ERROR_LOG(AUDIO, "Unable to install hw params: %s\n", snd_strerror(err));
return false;
}
err = snd_pcm_hw_params_get_buffer_size(hwparams, &buffer_size);
if (err < 0)
{
@ -176,10 +176,10 @@ bool AlsaSound::AlsaInit()
frames_to_deliver = buffer_size / periods;
//limit the minimum size. pulseaudio advertises a minimum of 32 samples.
if (frames_to_deliver < FRAME_COUNT_MIN)
frames_to_deliver = FRAME_COUNT_MIN;
frames_to_deliver = FRAME_COUNT_MIN;
//it is probably a bad idea to try to send more than one buffer of data
if ((unsigned int)frames_to_deliver > buffer_size)
frames_to_deliver = buffer_size;
frames_to_deliver = buffer_size;
NOTICE_LOG(AUDIO, "ALSA gave us a %ld sample \"hardware\" buffer with %d periods. Will send %d samples per fragments.\n", buffer_size, periods, frames_to_deliver);
snd_pcm_sw_params_alloca(&swparams);
@ -187,21 +187,21 @@ bool AlsaSound::AlsaInit()
err = snd_pcm_sw_params_current(handle, swparams);
if (err < 0)
{
ERROR_LOG(AUDIO, "cannot init sw params: %s\n", snd_strerror(err));
ERROR_LOG(AUDIO, "cannot init sw params: %s\n", snd_strerror(err));
return false;
}
err = snd_pcm_sw_params_set_start_threshold(handle, swparams, 0U);
if (err < 0)
{
ERROR_LOG(AUDIO, "cannot set start thresh: %s\n", snd_strerror(err));
ERROR_LOG(AUDIO, "cannot set start thresh: %s\n", snd_strerror(err));
return false;
}
err = snd_pcm_sw_params(handle, swparams);
if (err < 0)
{
ERROR_LOG(AUDIO, "cannot set sw params: %s\n", snd_strerror(err));
ERROR_LOG(AUDIO, "cannot set sw params: %s\n", snd_strerror(err));
return false;
}

6
Source/Core/AudioCommon/Src/CoreAudioSoundStream.h
View File

@ -30,19 +30,19 @@ class CoreAudioSound : public SoundStream
public:
CoreAudioSound(CMixer *mixer);
virtual ~CoreAudioSound();
virtual bool Start();
virtual void SetVolume(int volume);
virtual void SoundLoop();
virtual void Stop();
static bool isValid() {
return true;
}
virtual bool usesMixer() const {
return true;
}
virtual void Update();
private:

52
Source/Core/AudioCommon/Src/DPL2Decoder.cpp
View File

@ -130,15 +130,15 @@ returns 0 if OK, -1 if fail
*/
float* design_fir(unsigned int *n, float* fc, float opt)
{
unsigned int o = *n & 1; // Indicator for odd filter length
unsigned int end = ((*n + 1) >> 1) - o; // Loop end
unsigned int i; // Loop index
unsigned int o = *n & 1; // Indicator for odd filter length
unsigned int end = ((*n + 1) >> 1) - o; // Loop end
unsigned int i; // Loop index
float k1 = 2 * float(M_PI); // 2*pi*fc1
float k2 = 0.5f * (float)(1 - o);// Constant used if the filter has even length
float g = 0.0f; // Gain
float t1; // Temporary variables
float fc1; // Cutoff frequencies
float k1 = 2 * float(M_PI); // 2*pi*fc1
float k2 = 0.5f * (float)(1 - o); // Constant used if the filter has even length
float g = 0.0f; // Gain
float t1; // Temporary variables
float fc1; // Cutoff frequencies
// Sanity check
if(*n==0) return NULL;
@ -241,18 +241,18 @@ void matrix_decode(const float *in, const int k, const int il,
float *_rr, float *_cf)
{
static const float M9_03DB = 0.3535533906f;
static const float MATAGCTRIG = 8.0f; /* (Fuzzy) AGC trigger */
static const float MATAGCDECAY = 1.0f; /* AGC baseline decay rate (1/samp.) */
static const float MATAGCTRIG = 8.0f; /* (Fuzzy) AGC trigger */
static const float MATAGCDECAY = 1.0f; /* AGC baseline decay rate (1/samp.) */
static const float MATCOMPGAIN = 0.37f; /* Cross talk compensation gain, 0.50 - 0.55 is full cancellation. */
const int kr = (k + olddelay) % _dlbuflen;
float l_gain = (_l_fwr + _r_fwr) / (1 + _l_fwr + _l_fwr);
float r_gain = (_l_fwr + _r_fwr) / (1 + _r_fwr + _r_fwr);
/* The 2nd axis has strong gain fluctuations, and therefore require
limits. The factor corresponds to the 1 / amplification of (Lt
- Rt) when (Lt, Rt) is strongly correlated. (e.g. during
dialogues). It should be bigger than -12 dB to prevent
distortion. */
// The 2nd axis has strong gain fluctuations, and therefore require
// limits. The factor corresponds to the 1 / amplification of (Lt
// - Rt) when (Lt, Rt) is strongly correlated. (e.g. during
// dialogues). It should be bigger than -12 dB to prevent
// distortion.
float lmr_lim_fwr = _lmr_fwr > M9_03DB * _lpr_fwr ? _lmr_fwr : M9_03DB * _lpr_fwr;
float lpr_gain = (_lpr_fwr + lmr_lim_fwr) / (1 + _lpr_fwr + _lpr_fwr);
float lmr_gain = (_lpr_fwr + lmr_lim_fwr) / (1 + lmr_lim_fwr + lmr_lim_fwr);
@ -275,9 +275,9 @@ void matrix_decode(const float *in, const int k, const int il,
if (decode_rear)
{
_lr[kr] = _rr[kr] = (l_agc - r_agc) * (float)M_SQRT1_2;
/* Stereo rear channel is steered with the same AGC steering as
the decoding matrix. Note this requires a fast updating AGC
at the order of 20 ms (which is the case here). */
// Stereo rear channel is steered with the same AGC steering as
// the decoding matrix. Note this requires a fast updating AGC
// at the order of 20 ms (which is the case here).
_lr[kr] *= (_l_fwr + _l_fwr) / (1 + _l_fwr + _r_fwr);
_rr[kr] *= (_r_fwr + _r_fwr) / (1 + _l_fwr + _r_fwr);
}
@ -298,16 +298,16 @@ void matrix_decode(const float *in, const int k, const int il,
_rf[k] = (lpr_agc - lmr_agc) * (float)M_SQRT1_2;
/*** CENTER FRONT CANCELLATION ***/
/* A heuristic approach exploits that Lt + Rt gain contains the
information about Lt, Rt correlation. This effectively reshapes
the front and rear "cones" to concentrate Lt + Rt to C and
introduce Lt - Rt in L, R. */
// A heuristic approach exploits that Lt + Rt gain contains the
// information about Lt, Rt correlation. This effectively reshapes
// the front and rear "cones" to concentrate Lt + Rt to C and
// introduce Lt - Rt in L, R.
/* 0.67677 is the empirical lower bound for lpr_gain. */
c_gain = 8 * (*_adapt_lpr_gain - 0.67677f);
c_gain = c_gain > 0 ? c_gain : 0;
/* c_gain should not be too high, not even reaching full
cancellation (~ 0.50 - 0.55 at current AGC implementation), or
the center will sound too narrow. */
// c_gain should not be too high, not even reaching full
// cancellation (~ 0.50 - 0.55 at current AGC implementation), or
// the center will sound too narrow. */
c_gain = MATCOMPGAIN / (1 + c_gain * c_gain);
c_agc_cfk = c_gain * _cf[k];
_lf[k] -= c_agc_cfk;
@ -317,7 +317,7 @@ void matrix_decode(const float *in, const int k, const int il,
void dpl2decode(float *samples, int numsamples, float *out)
{
static const unsigned int FWRDURATION = 240; /* FWR average duration (samples) */
static const unsigned int FWRDURATION = 240; // FWR average duration (samples)
static const int cfg_delay = 0;
static const unsigned int fmt_freq = 48000;
static const unsigned int fmt_nchannels = 2; // input channels

2
Source/Core/AudioCommon/Src/DSoundStream.cpp
View File

@ -122,7 +122,7 @@ void DSound::SoundLoop()
bool DSound::Start()
{
if (FAILED(DirectSoundCreate8(0, &ds, 0)))
return false;
return false;
if (hWnd)
{
HRESULT hr = ds->SetCooperativeLevel((HWND)hWnd, DSSCL_PRIORITY);

50
Source/Core/AudioCommon/Src/DSoundStream.h
View File

@ -31,33 +31,33 @@
class DSound : public SoundStream
{
#ifdef _WIN32
std::thread thread;
Common::Event soundSyncEvent;
void *hWnd;
std::thread thread;
Common::Event soundSyncEvent;
void *hWnd;
IDirectSound8* ds;
IDirectSoundBuffer* dsBuffer;
int bufferSize; //i bytes
IDirectSound8* ds;
IDirectSoundBuffer* dsBuffer;
int bufferSize; //i bytes
int m_volume;
// playback position
int currentPos;
int lastPos;
short realtimeBuffer[BUFSIZE / sizeof(short)];
inline int FIX128(int x)
// playback position
int currentPos;
int lastPos;
short realtimeBuffer[BUFSIZE / sizeof(short)];
inline int FIX128(int x)
{
return x & (~127);
}
}
inline int ModBufferSize(int x)
inline int ModBufferSize(int x)
{
return (x + bufferSize) % bufferSize;
}
}
bool CreateBuffer();
bool WriteDataToBuffer(DWORD dwOffset, char* soundData, DWORD dwSoundBytes);
bool CreateBuffer();
bool WriteDataToBuffer(DWORD dwOffset, char* soundData, DWORD dwSoundBytes);
public:
DSound(CMixer *mixer, void *_hWnd = NULL)
@ -70,16 +70,16 @@ public:
, hWnd(_hWnd)
{}
virtual ~DSound() {}
virtual ~DSound() {}
virtual bool Start();
virtual void SoundLoop();
virtual void SoundLoop();
virtual void SetVolume(int volume);
virtual void Stop();
virtual void Stop();
virtual void Clear(bool mute);
static bool isValid() { return true; }
virtual bool usesMixer() const { return true; }
virtual void Update();
static bool isValid() { return true; }
virtual bool usesMixer() const { return true; }
virtual void Update();
#else
public:

11
Source/Core/AudioCommon/Src/Mixer.cpp
View File

@ -67,7 +67,7 @@ unsigned int CMixer::Mix(short* samples, unsigned int numSamples)
{
static const __m128i sr_mask =
_mm_set_epi32(0x0C0D0E0FL, 0x08090A0BL,
0x04050607L, 0x00010203L);
0x04050607L, 0x00010203L);
for (unsigned int i = 0; i < numLeft * 2; i += 8)
{
@ -99,12 +99,11 @@ unsigned int CMixer::Mix(short* samples, unsigned int numSamples)
if ((m_indexR2 & INDEX_MASK) == (m_indexW & INDEX_MASK)) //..if it exists
m_indexR2 = m_indexR;
s16 l1 = Common::swap16(m_buffer[m_indexR & INDEX_MASK]); //current
s16 l2 = Common::swap16(m_buffer[m_indexR2 & INDEX_MASK]); //next
int sampleL = ((l1 << 16) + (l2 - l1) * (u16)frac) >> 16;
samples[i+1] = sampleL;
int sampleL = ((l1 << 16) + (l2 - l1) * (u16)frac) >> 16;
samples[i+1] = sampleL;
s16 r1 = Common::swap16(m_buffer[(m_indexR + 1) & INDEX_MASK]); //current
s16 r2 = Common::swap16(m_buffer[(m_indexR2 + 1) & INDEX_MASK]); //next
int sampleR = ((r1 << 16) + (r2 - r1) * (u16)frac) >> 16;
@ -116,8 +115,6 @@ unsigned int CMixer::Mix(short* samples, unsigned int numSamples)
}
}
} else {
numLeft = 0;
}

18
Source/Core/AudioCommon/Src/NullSoundStream.h
View File

@ -25,24 +25,24 @@
class NullSound : public SoundStream
{
// playback position
short realtimeBuffer[BUF_SIZE / sizeof(short)];
// playback position
short realtimeBuffer[BUF_SIZE / sizeof(short)];
public:
NullSound(CMixer *mixer, void *hWnd = NULL)
: SoundStream(mixer)
{}
virtual ~NullSound() {}
virtual ~NullSound() {}
virtual bool Start();
virtual void SoundLoop();
virtual void SoundLoop();
virtual void SetVolume(int volume);
virtual void Stop();
virtual void Stop();
virtual void Clear(bool mute);
static bool isValid() { return true; }
virtual bool usesMixer() const { return true; }
virtual void Update();
static bool isValid() { return true; }
virtual bool usesMixer() const { return true; }
virtual void Update();
};
#endif //_NULLSOUNDSTREAM_H_

2
Source/Core/AudioCommon/Src/OpenALStream.cpp
View File

@ -295,7 +295,6 @@ void OpenALStream::SoundLoop()
{
ERROR_LOG(AUDIO, "Error occurred while buffering float32 data: %08x", err);
}
}
#endif
if (!float32_capable)
@ -308,7 +307,6 @@ void OpenALStream::SoundLoop()
stereo[i * 2 + 1] = (short)((float)sampleBuffer[i * 2 + 1] * (1 << 16));
}
alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO16, stereo, nSamples * 2 * 2, ulFrequency);
}
}

10
Source/Core/AudioCommon/Src/SoundStream.h
View File

@ -27,17 +27,17 @@ class SoundStream
protected:
CMixer *m_mixer;
// We set this to shut down the sound thread.
// 0=keep playing, 1=stop playing NOW.
volatile int threadData;
bool m_logAudio;
// We set this to shut down the sound thread.
// 0=keep playing, 1=stop playing NOW.
volatile int threadData;
bool m_logAudio;
WaveFileWriter g_wave_writer;
bool m_muted;
public:
SoundStream(CMixer *mixer) : m_mixer(mixer), threadData(0), m_logAudio(false), m_muted(false) {}
virtual ~SoundStream() { delete m_mixer;}
static bool isValid() { return false; }
virtual CMixer *GetMixer() const { return m_mixer; }
virtual bool Start() { return false; }

10
Source/Core/AudioCommon/Src/WaveFile.cpp
View File

@ -104,21 +104,21 @@ void WaveFileWriter::AddStereoSamples(const short *sample_data, u32 count)
{
if (!file)
PanicAlertT("WaveFileWriter - file not open.");
if (skip_silence)
{
bool all_zero = true;
for (u32 i = 0; i < count * 2; i++)
{
if (sample_data[i])
all_zero = false;
}
if (all_zero)
return;
}
file.WriteBytes(sample_data, count * 4);
audio_size += count * 4;
}
@ -134,7 +134,7 @@ void WaveFileWriter::AddStereoSamplesBE(const short *sample_data, u32 count)
if (skip_silence)
{
bool all_zero = true;
for (u32 i = 0; i < count * 2; i++)
{
if (sample_data[i])

6
Source/Core/AudioCommon/Src/XAudio2Stream.cpp
View File

@ -95,10 +95,10 @@ void StreamingVoiceContext::OnBufferEnd(void* context)
if (!m_source_voice || !context)
return;
//m_sound_sync_event->Wait(); // sync
//m_sound_sync_event->Spin(); // or tight sync
m_mixer->Mix(static_cast<short*>(context), SAMPLES_PER_BUFFER);
SubmitBuffer(static_cast<BYTE*>(context));
}
@ -183,6 +183,6 @@ void XAudio2::Stop()
m_mastering_voice->DestroyVoice();
m_mastering_voice = nullptr;
}
m_xaudio2.reset(); // release interface
}