Silence some clang warnings
This commit is contained in:
@@ -859,7 +859,6 @@ static void alc_initconfig(void)
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size_t len;
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const char *next = str;
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i = 0;
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do {
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str = next;
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next = strchr(str, ',');
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+2
-2
@@ -146,7 +146,7 @@ static ALuint WaveProc(ALvoid *ptr)
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{
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fs = fwrite(data->buffer, frameSize, Device->UpdateSize,
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data->f);
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fs = fs;
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(void)fs;
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}
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if(ferror(data->f))
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{
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@@ -257,7 +257,7 @@ static ALCboolean wave_reset_playback(ALCdevice *device)
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fwrite32le(channel_masks[channels], data->f);
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// 16 byte GUID, sub-type format
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val = fwrite(((bits==32) ? SUBTYPE_FLOAT : SUBTYPE_PCM), 1, 16, data->f);
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val = val;
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(void)val;
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fprintf(data->f, "data");
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fwrite32le(0xFFFFFFFF, data->f); // 'data' header len; filled in at close
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@@ -855,7 +855,14 @@ static inline ALfloat CalcDampingCoeff(ALfloat hfRatio, ALfloat length, ALfloat
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// Damping is done with a 1-pole filter, so g needs to be squared.
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g *= g;
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coeff = lpCoeffCalc(g, cw);
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if(g < 0.9999f) /* 1-epsilon */
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{
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/* Be careful with gains < 0.001, as that causes the coefficient
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* head towards 1, which will flatten the signal. */
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g = maxf(g, 0.001f);
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coeff = (1 - g*cw - sqrtf(2*g*(1-cw) - g*g*(1 - cw*cw))) /
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(1 - g);
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}
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// Very low decay times will produce minimal output, so apply an
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// upper bound to the coefficient.
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+3
-3
@@ -161,7 +161,7 @@ ALvoid ComputeAngleGains(const ALCdevice *device, ALfloat angle, ALfloat hwidth,
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SpeakerAngle[i] = device->SpeakerAngle[i];
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/* Some easy special-cases first... */
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if(device->NumChan == 1 || hwidth >= F_PI)
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if(device->NumChan <= 1 || hwidth >= F_PI)
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{
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/* Full coverage for all speakers. */
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for(i = 0;i < device->NumChan;i++)
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@@ -203,7 +203,7 @@ ALvoid ComputeAngleGains(const ALCdevice *device, ALfloat angle, ALfloat hwidth,
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* within -pi...+pi. */
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if(angle > 0.0f)
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{
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ALuint done = 0;
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ALuint done;
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ALuint i = 0;
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while(i < device->NumChan && device->SpeakerAngle[i]-angle < -F_PI)
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i++;
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@@ -226,7 +226,7 @@ ALvoid ComputeAngleGains(const ALCdevice *device, ALfloat angle, ALfloat hwidth,
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* we need to handle index 0. Because the iterators are unsigned,
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* they'll underflow and wrap to become 0xFFFFFFFF, which will
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* break as expected. */
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ALuint done = device->NumChan-1;
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ALuint done;
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ALuint i = device->NumChan-1;
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while(i < device->NumChan && device->SpeakerAngle[i]-angle > F_PI)
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i--;
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@@ -9,12 +9,6 @@ extern "C" {
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#define LOWPASSFREQREF (5000)
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/* Calculates the low-pass filter coefficient given the pre-scaled gain and
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* cos(w) value. Note that g should be pre-scaled (sqr(gain) for one-pole,
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* sqrt(gain) for four-pole, etc) */
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ALfloat lpCoeffCalc(ALfloat g, ALfloat cw);
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typedef enum ALfilterType {
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ALfilterType_HighShelf,
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ALfilterType_LowShelf,
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+7
-6
@@ -455,7 +455,7 @@ AL_API ALvoid AL_APIENTRY alBufferSubDataSOFT(ALuint buffer, ALenum format, cons
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offset = offset/OldBytes * Bytes;
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length = length/OldBytes/Channels;
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}
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ConvertData(&((ALubyte*)ALBuf->data)[offset], ALBuf->FmtType,
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ConvertData((char*)ALBuf->data+offset, (enum UserFmtType)ALBuf->FmtType,
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data, SrcType, Channels, length);
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WriteUnlock(&ALBuf->lock);
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}
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@@ -531,7 +531,7 @@ AL_API void AL_APIENTRY alBufferSubSamplesSOFT(ALuint buffer,
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/* offset -> byte offset */
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offset *= FrameSize;
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ConvertData(&((ALubyte*)ALBuf->data)[offset], ALBuf->FmtType,
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ConvertData((char*)ALBuf->data+offset, (enum UserFmtType)ALBuf->FmtType,
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data, type, ChannelsFromFmt(ALBuf->FmtChannels), samples);
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WriteUnlock(&ALBuf->lock);
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}
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@@ -581,7 +581,8 @@ AL_API void AL_APIENTRY alGetBufferSamplesSOFT(ALuint buffer,
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/* offset -> byte offset */
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offset *= FrameSize;
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ConvertData(data, type, &((ALubyte*)ALBuf->data)[offset], ALBuf->FmtType,
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ConvertData(data, type,
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(char*)ALBuf->data+offset, (enum UserFmtType)ALBuf->FmtType,
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ChannelsFromFmt(ALBuf->FmtChannels), samples);
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ReadUnlock(&ALBuf->lock);
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}
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@@ -1972,7 +1973,7 @@ static ALenum LoadData(ALbuffer *ALBuf, ALuint freq, ALenum NewFormat, ALsizei f
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ALBuf->data = temp;
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if(data != NULL)
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ConvertData(ALBuf->data, DstType, data, SrcType, NewChannels, frames);
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ConvertData(ALBuf->data, (enum UserFmtType)DstType, data, SrcType, NewChannels, frames);
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if(storesrc)
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{
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@@ -1985,8 +1986,8 @@ static ALenum LoadData(ALbuffer *ALBuf, ALuint freq, ALenum NewFormat, ALsizei f
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}
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else
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{
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ALBuf->OriginalChannels = DstChannels;
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ALBuf->OriginalType = DstType;
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ALBuf->OriginalChannels = (enum UserFmtChannels)DstChannels;
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ALBuf->OriginalType = (enum UserFmtType)DstType;
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ALBuf->OriginalSize = frames * NewBytes * NewChannels;
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}
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@@ -403,23 +403,6 @@ void ALfilterState_setParams(ALfilterState *filter, ALfilterType type, ALfloat g
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}
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ALfloat lpCoeffCalc(ALfloat g, ALfloat cw)
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{
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ALfloat a = 0.0f;
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if(g < 0.9999f) /* 1-epsilon */
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{
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/* Be careful with gains < 0.001, as that causes the coefficient head
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* towards 1, which will flatten the signal */
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g = maxf(g, 0.001f);
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a = (1 - g*cw - sqrtf(2*g*(1-cw) - g*g*(1 - cw*cw))) /
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(1 - g);
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}
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return a;
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}
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static void lp_SetParami(ALfilter *filter, ALCcontext *context, ALenum param, ALint val)
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{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); (void)filter;(void)param;(void)val; }
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static void lp_SetParamiv(ALfilter *filter, ALCcontext *context, ALenum param, const ALint *vals)
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+2
-2
@@ -534,13 +534,13 @@ static ALenum SetSourcefv(ALsource *Source, ALCcontext *Context, SrcFloatProp pr
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case sfAuxSendFilterGainHFAuto:
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case sfDirectChannelsSOFT:
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ival = (ALint)values[0];
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return SetSourceiv(Source, Context, prop, &ival);
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return SetSourceiv(Source, Context, (SrcIntProp)prop, &ival);
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case sfBuffer:
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case sfBuffersQueued:
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case sfBuffersProcessed:
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ival = (ALint)((ALuint)values[0]);
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return SetSourceiv(Source, Context, prop, &ival);
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return SetSourceiv(Source, Context, (SrcIntProp)prop, &ival);
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}
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ERR("Unexpected property: 0x%04x\n", prop);
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