Update juce

source/modules/juce_audio_basics/buffers/juce_AudioDataConverters.h

@@ -487,13 +487,10 @@ public:
         }
 
         /** Scans a block of data, returning the lowest and highest levels as floats */
-        void findMinAndMax (size_t numSamples, float& minValue, float& maxValue) const noexcept
+        Range<float> findMinAndMax (size_t numSamples) const noexcept
         {
             if (numSamples == 0)
-            {
-                minValue = maxValue = 0;
-                return;
-            }
+                return Range<float>();
 
             Pointer dest (*this);
 
@@ -512,27 +509,32 @@ public:
                     if (v < mn)  mn = v;
                 }
 
-                minValue = mn;
-                maxValue = mx;
+                return Range<float> (mn, mx);
             }
-            else
+
+            int32 mn = dest.getAsInt32();
+            dest.advance();
+            int32 mx = mn;
+
+            while (--numSamples > 0)
             {
-                int32 mn = dest.getAsInt32();
+                const int v = dest.getAsInt32();
                 dest.advance();
-                int32 mx = mn;
 
-                while (--numSamples > 0)
-                {
-                    const int v = dest.getAsInt32();
-                    dest.advance();
+                if (mx < v)  mx = v;
+                if (v < mn)  mn = v;
+            }
 
-                    if (mx < v)  mx = v;
-                    if (v < mn)  mn = v;
-                }
+            return Range<float> (mn * (float) (1.0 / (1.0 + Int32::maxValue)),
+                                 mx * (float) (1.0 / (1.0 + Int32::maxValue)));
+        }
 
-                minValue = mn * (float) (1.0 / (1.0 + Int32::maxValue));
-                maxValue = mx * (float) (1.0 / (1.0 + Int32::maxValue));
-            }
+        /** Scans a block of data, returning the lowest and highest levels as floats */
+        void findMinAndMax (size_t numSamples, float& minValue, float& maxValue) const noexcept
+        {
+            Range<float> r (findMinAndMax (numSamples));
+            minValue = r.getStart();
+            maxValue = r.getEnd();
         }
 
         /** Returns true if the pointer is using a floating-point format. */

source/modules/juce_audio_basics/buffers/juce_FloatVectorOperations.cpp

@@ -134,6 +134,19 @@ namespace FloatVectorHelpers
         } \
         JUCE_FINISH_VEC_OP (normalOp)
 
+    #define JUCE_PERFORM_VEC_OP_SRC1_SRC2_DEST_DEST(normalOp, vecOp, locals, increment, setupOp) \
+        JUCE_BEGIN_VEC_OP \
+        setupOp \
+        { \
+            Mode::ParallelType (&loadSrc1) (const Mode::Type* v) = FloatVectorHelpers::isAligned (src1) ? Mode::loadA  : Mode::loadU;          \
+            Mode::ParallelType (&loadSrc2) (const Mode::Type* v) = FloatVectorHelpers::isAligned (src2) ? Mode::loadA  : Mode::loadU;          \
+            Mode::ParallelType (&loadDst)  (const Mode::Type* v) = FloatVectorHelpers::isAligned (dest) ? Mode::loadA  : Mode::loadU;          \
+            void (&storeDst) (Mode::Type* dest, Mode::ParallelType a) = FloatVectorHelpers::isAligned (dest) ? Mode::storeA  : Mode::storeU;   \
+            JUCE_VEC_LOOP_TWO_SOURCES_WITH_DEST_LOAD (vecOp, loadSrc1, loadSrc2, loadDst, storeDst, locals, increment);                        \
+        } \
+        JUCE_FINISH_VEC_OP (normalOp)
+
+
     //==============================================================================
    #elif JUCE_USE_ARM_NEON
 
@@ -211,6 +224,13 @@ namespace FloatVectorHelpers
         JUCE_VEC_LOOP_TWO_SOURCES (vecOp, Mode::loadU, Mode::loadU, Mode::storeU, locals, increment) \
         JUCE_FINISH_VEC_OP (normalOp)
 
+    #define JUCE_PERFORM_VEC_OP_SRC1_SRC2_DEST_DEST(normalOp, vecOp, locals, increment, setupOp) \
+        JUCE_BEGIN_VEC_OP \
+        setupOp \
+        JUCE_VEC_LOOP_TWO_SOURCES_WITH_DEST_LOAD (vecOp, Mode::loadU, Mode::loadU, Mode::loadU, Mode::storeU, locals, increment) \
+        JUCE_FINISH_VEC_OP (normalOp)
+
+
     //==============================================================================
    #else
     #define JUCE_PERFORM_VEC_OP_DEST(normalOp, vecOp, locals, setupOp) \
@@ -221,6 +241,10 @@ namespace FloatVectorHelpers
 
     #define JUCE_PERFORM_VEC_OP_SRC1_SRC2_DEST(normalOp, vecOp, locals, increment, setupOp) \
         for (int i = 0; i < num; ++i) normalOp;
+
+    #define JUCE_PERFORM_VEC_OP_SRC1_SRC2_DEST_DEST(normalOp, vecOp, locals, increment, setupOp) \
+        for (int i = 0; i < num; ++i) normalOp;
+
    #endif
 
     //==============================================================================
@@ -240,11 +264,20 @@ namespace FloatVectorHelpers
             increment; \
         }
 
+    #define JUCE_VEC_LOOP_TWO_SOURCES_WITH_DEST_LOAD(vecOp, src1Load, src2Load, dstLoad, dstStore, locals, increment) \
+        for (int i = 0; i < numLongOps; ++i) \
+        { \
+            locals (src1Load, src2Load, dstLoad); \
+            dstStore (dest, vecOp); \
+            increment; \
+        }
+
     #define JUCE_LOAD_NONE(srcLoad, dstLoad)
-    #define JUCE_LOAD_DEST(srcLoad, dstLoad)        const Mode::ParallelType d = dstLoad (dest);
-    #define JUCE_LOAD_SRC(srcLoad, dstLoad)         const Mode::ParallelType s = srcLoad (src);
-    #define JUCE_LOAD_SRC1_SRC2(src1Load, src2Load) const Mode::ParallelType s1 = src1Load (src1), s2 = src2Load (src2);
-    #define JUCE_LOAD_SRC_DEST(srcLoad, dstLoad)    const Mode::ParallelType d = dstLoad (dest), s = srcLoad (src);
+    #define JUCE_LOAD_DEST(srcLoad, dstLoad)                        const Mode::ParallelType d = dstLoad (dest);
+    #define JUCE_LOAD_SRC(srcLoad, dstLoad)                         const Mode::ParallelType s = srcLoad (src);
+    #define JUCE_LOAD_SRC1_SRC2(src1Load, src2Load)                 const Mode::ParallelType s1 = src1Load (src1), s2 = src2Load (src2);
+    #define JUCE_LOAD_SRC1_SRC2_DEST(src1Load, src2Load, dstLoad)   const Mode::ParallelType d = dstLoad (dest), s1 = src1Load (src1), s2 = src2Load (src2);
+    #define JUCE_LOAD_SRC_DEST(srcLoad, dstLoad)                    const Mode::ParallelType d = dstLoad (dest), s = srcLoad (src);
 
    #if JUCE_USE_SSE_INTRINSICS || JUCE_USE_ARM_NEON
     template<int typeSize> struct ModeType    { typedef BasicOps32 Mode; };
@@ -580,6 +613,28 @@ void JUCE_CALLTYPE FloatVectorOperations::addWithMultiply (double* dest, const d
                                   const Mode::ParallelType mult = Mode::load1 (multiplier);)
 }
 
+void JUCE_CALLTYPE FloatVectorOperations::addWithMultiply (float* dest, const float* src1, const float* src2, int num) noexcept
+{
+   #if JUCE_USE_VDSP_FRAMEWORK
+    vDSP_vma ((float*) src1, 1, (float*) src2, 1, dest, 1, dest, 1, (vDSP_Length) num);
+   #else
+    JUCE_PERFORM_VEC_OP_SRC1_SRC2_DEST_DEST (dest[i] += src1[i] * src2[i], Mode::add (d, Mode::mul (s1, s2)),
+                                             JUCE_LOAD_SRC1_SRC2_DEST,
+                                             JUCE_INCREMENT_SRC1_SRC2_DEST, )
+   #endif
+}
+
+void JUCE_CALLTYPE FloatVectorOperations::addWithMultiply (double* dest, const double* src1, const double* src2, int num) noexcept
+{
+   #if JUCE_USE_VDSP_FRAMEWORK
+    vDSP_vmaD ((double*) src1, 1, (double*) src2, 1, dest, 1, dest, 1, (vDSP_Length) num);
+   #else
+    JUCE_PERFORM_VEC_OP_SRC1_SRC2_DEST_DEST (dest[i] += src1[i] * src2[i], Mode::add (d, Mode::mul (s1, s2)),
+                                             JUCE_LOAD_SRC1_SRC2_DEST,
+                                             JUCE_INCREMENT_SRC1_SRC2_DEST, )
+   #endif
+}
+
 void JUCE_CALLTYPE FloatVectorOperations::multiply (float* dest, const float* src, int num) noexcept
 {
    #if JUCE_USE_VDSP_FRAMEWORK
@@ -682,6 +737,96 @@ void JUCE_CALLTYPE FloatVectorOperations::convertFixedToFloat (float* dest, cons
    #endif
 }
 
+void JUCE_CALLTYPE FloatVectorOperations::min (float* dest, const float* src, float comp, int num) noexcept
+{
+    JUCE_PERFORM_VEC_OP_SRC_DEST (dest[i] = jmin (src[i], comp), Mode::min (s, cmp),
+                                  JUCE_LOAD_SRC, JUCE_INCREMENT_SRC_DEST,
+                                  const Mode::ParallelType cmp = Mode::load1 (comp);)
+}
+
+void JUCE_CALLTYPE FloatVectorOperations::min (double* dest, const double* src, double comp, int num) noexcept
+{
+    JUCE_PERFORM_VEC_OP_SRC_DEST (dest[i] = jmin (src[i], comp), Mode::min (s, cmp),
+                                  JUCE_LOAD_SRC, JUCE_INCREMENT_SRC_DEST,
+                                  const Mode::ParallelType cmp = Mode::load1 (comp);)
+}
+
+void JUCE_CALLTYPE FloatVectorOperations::min (float* dest, const float* src1, const float* src2, int num) noexcept
+{
+   #if JUCE_USE_VDSP_FRAMEWORK
+    vDSP_vmin ((float*) src1, 1, (float*) src2, 1, dest, 1, (vDSP_Length) num);
+   #else
+    JUCE_PERFORM_VEC_OP_SRC1_SRC2_DEST (dest[i] = jmin (src1[i], src2[i]), Mode::min (s1, s2), JUCE_LOAD_SRC1_SRC2, JUCE_INCREMENT_SRC1_SRC2_DEST, )
+   #endif
+}
+
+void JUCE_CALLTYPE FloatVectorOperations::min (double* dest, const double* src1, const double* src2, int num) noexcept
+{
+   #if JUCE_USE_VDSP_FRAMEWORK
+    vDSP_vminD ((double*) src1, 1, (double*) src2, 1, dest, 1, (vDSP_Length) num);
+   #else
+    JUCE_PERFORM_VEC_OP_SRC1_SRC2_DEST (dest[i] = jmin (src1[i], src2[i]), Mode::min (s1, s2), JUCE_LOAD_SRC1_SRC2, JUCE_INCREMENT_SRC1_SRC2_DEST, )
+   #endif
+}
+
+void JUCE_CALLTYPE FloatVectorOperations::max (float* dest, const float* src, float comp, int num) noexcept
+{
+    JUCE_PERFORM_VEC_OP_SRC_DEST (dest[i] = jmax (src[i], comp), Mode::max (s, cmp),
+                                  JUCE_LOAD_SRC, JUCE_INCREMENT_SRC_DEST,
+                                  const Mode::ParallelType cmp = Mode::load1 (comp);)
+}
+
+void JUCE_CALLTYPE FloatVectorOperations::max (double* dest, const double* src, double comp, int num) noexcept
+{
+    JUCE_PERFORM_VEC_OP_SRC_DEST (dest[i] = jmax (src[i], comp), Mode::max (s, cmp),
+                                  JUCE_LOAD_SRC, JUCE_INCREMENT_SRC_DEST,
+                                  const Mode::ParallelType cmp = Mode::load1 (comp);)
+}
+
+void JUCE_CALLTYPE FloatVectorOperations::max (float* dest, const float* src1, const float* src2, int num) noexcept
+{
+   #if JUCE_USE_VDSP_FRAMEWORK
+    vDSP_vmax ((float*) src1, 1, (float*) src2, 1, dest, 1, (vDSP_Length) num);
+   #else
+    JUCE_PERFORM_VEC_OP_SRC1_SRC2_DEST (dest[i] = jmax (src1[i], src2[i]), Mode::max (s1, s2), JUCE_LOAD_SRC1_SRC2, JUCE_INCREMENT_SRC1_SRC2_DEST, )
+   #endif
+}
+
+void JUCE_CALLTYPE FloatVectorOperations::max (double* dest, const double* src1, const double* src2, int num) noexcept
+{
+   #if JUCE_USE_VDSP_FRAMEWORK
+    vDSP_vmaxD ((double*) src1, 1, (double*) src2, 1, dest, 1, (vDSP_Length) num);
+   #else
+    JUCE_PERFORM_VEC_OP_SRC1_SRC2_DEST (dest[i] = jmax (src1[i], src2[i]), Mode::max (s1, s2), JUCE_LOAD_SRC1_SRC2, JUCE_INCREMENT_SRC1_SRC2_DEST, )
+   #endif
+}
+
+void JUCE_CALLTYPE FloatVectorOperations::clip (float* dest, const float* src, float low, float high, int num) noexcept
+{
+    jassert(high >= low);
+
+   #if JUCE_USE_VDSP_FRAMEWORK
+    vDSP_vclip ((float*) src, 1, &low, &high, dest, 1, (vDSP_Length) num);
+   #else
+    JUCE_PERFORM_VEC_OP_SRC_DEST (dest[i] = jmax (jmin (src[i], high), low), Mode::max (Mode::min (s, hi), lo),
+                                  JUCE_LOAD_SRC, JUCE_INCREMENT_SRC_DEST,
+                                  const Mode::ParallelType lo = Mode::load1 (low); const Mode::ParallelType hi = Mode::load1 (high);)
+   #endif
+}
+
+void JUCE_CALLTYPE FloatVectorOperations::clip (double* dest, const double* src, double low, double high, int num) noexcept
+{
+    jassert(high >= low);
+
+   #if JUCE_USE_VDSP_FRAMEWORK
+    vDSP_vclipD ((double*) src, 1, &low, &high, dest, 1, (vDSP_Length) num);
+   #else
+    JUCE_PERFORM_VEC_OP_SRC_DEST (dest[i] = jmax (jmin (src[i], high), low), Mode::max (Mode::min (s, hi), lo),
+                                  JUCE_LOAD_SRC, JUCE_INCREMENT_SRC_DEST,
+                                  const Mode::ParallelType lo = Mode::load1 (low); const Mode::ParallelType hi = Mode::load1 (high);)
+   #endif
+}
+
 Range<float> JUCE_CALLTYPE FloatVectorOperations::findMinAndMax (const float* src, int num) noexcept
 {
    #if JUCE_USE_SSE_INTRINSICS || JUCE_USE_ARM_NEON
@@ -823,6 +968,11 @@ public:
 
             fillRandomly (random, int1, num);
             doConversionTest (u, data1, data2, int1, num);
+
+            FloatVectorOperations::fill (data1, (ValueType) 2, num);
+            FloatVectorOperations::fill (data2, (ValueType) 3, num);
+            FloatVectorOperations::addWithMultiply (data1, data1, data2, num);
+            u.expect (areAllValuesEqual (data1, num, (ValueType) 8));
         }
 
         static void doConversionTest (UnitTest& u, float* data1, float* data2, int* const int1, int num)

source/modules/juce_audio_basics/buffers/juce_FloatVectorOperations.h

@@ -101,6 +101,12 @@ public:
     /** Multiplies each source value by the given multiplier, then adds it to the destination value. */
     static void JUCE_CALLTYPE addWithMultiply (double* dest, const double* src, double multiplier, int numValues) noexcept;
 
+    /** Multiplies each source1 value by the corresponding source2 value, then adds it to the destination value. */
+    static void JUCE_CALLTYPE addWithMultiply (float* dest, const float* src1, const float* src2, int num) noexcept;
+
+    /** Multiplies each source1 value by the corresponding source2 value, then adds it to the destination value. */
+    static void JUCE_CALLTYPE addWithMultiply (double* dest, const double* src1, const double* src2, int num) noexcept;
+
     /** Multiplies the destination values by the source values. */
     static void JUCE_CALLTYPE multiply (float* dest, const float* src, int numValues) noexcept;
 
@@ -134,6 +140,36 @@ public:
     /** Converts a stream of integers to floats, multiplying each one by the given multiplier. */
     static void JUCE_CALLTYPE convertFixedToFloat (float* dest, const int* src, float multiplier, int numValues) noexcept;
 
+    /** Each element of dest will be the minimum of the corresponding element of the source array and the given comp value. */
+    static void JUCE_CALLTYPE min (float* dest, const float* src, float comp, int num) noexcept;
+
+    /** Each element of dest will be the minimum of the corresponding element of the source array and the given comp value. */
+    static void JUCE_CALLTYPE min (double* dest, const double* src, double comp, int num) noexcept;
+
+    /** Each element of dest will be the minimum of the corresponding source1 and source2 values. */
+    static void JUCE_CALLTYPE min (float* dest, const float* src1, const float* src2, int num) noexcept;
+
+    /** Each element of dest will be the minimum of the corresponding source1 and source2 values. */
+    static void JUCE_CALLTYPE min (double* dest, const double* src1, const double* src2, int num) noexcept;
+
+    /** Each element of dest will be the maximum of the corresponding element of the source array and the given comp value. */
+    static void JUCE_CALLTYPE max (float* dest, const float* src, float comp, int num) noexcept;
+
+    /** Each element of dest will be the maximum of the corresponding element of the source array and the given comp value. */
+    static void JUCE_CALLTYPE max (double* dest, const double* src, double comp, int num) noexcept;
+
+    /** Each element of dest will be the maximum of the corresponding source1 and source2 values. */
+    static void JUCE_CALLTYPE max (float* dest, const float* src1, const float* src2, int num) noexcept;
+
+    /** Each element of dest will be the maximum of the corresponding source1 and source2 values. */
+    static void JUCE_CALLTYPE max (double* dest, const double* src1, const double* src2, int num) noexcept;
+
+    /** Each element of dest is calculated by hard clipping the corresponding src element so that it is in the range specified by the arguments low and high. */
+    static void JUCE_CALLTYPE clip (float* dest, const float* src, float low, float high, int num) noexcept;
+
+    /** Each element of dest is calculated by hard clipping the corresponding src element so that it is in the range specified by the arguments low and high. */
+    static void JUCE_CALLTYPE clip (double* dest, const double* src, double low, double high, int num) noexcept;
+
     /** Finds the miniumum and maximum values in the given array. */
     static Range<float> JUCE_CALLTYPE findMinAndMax (const float* src, int numValues) noexcept;
 

source/modules/juce_audio_basics/effects/juce_FFT.cpp

@@ -0,0 +1,277 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+// (For the moment, we'll implement a few local operators for this complex class - one
+// day we'll probably either have a juce complex class, or use the C++11 one)
+static FFT::Complex operator+ (FFT::Complex a, FFT::Complex b) noexcept     { FFT::Complex c = { a.r + b.r, a.i + b.i }; return c; }
+static FFT::Complex operator- (FFT::Complex a, FFT::Complex b) noexcept     { FFT::Complex c = { a.r - b.r, a.i - b.i }; return c; }
+static FFT::Complex operator* (FFT::Complex a, FFT::Complex b) noexcept     { FFT::Complex c = { a.r * b.r - a.i * b.i, a.r * b.i + a.i * b.r }; return c; }
+static FFT::Complex& operator+= (FFT::Complex& a, FFT::Complex b) noexcept  { a.r += b.r; a.i += b.i; return a; }
+
+//==============================================================================
+struct FFT::FFTConfig
+{
+    FFTConfig (int sizeOfFFT, bool isInverse)
+        : fftSize (sizeOfFFT), inverse (isInverse), twiddleTable ((size_t) sizeOfFFT)
+    {
+        for (int i = 0; i < fftSize; ++i)
+        {
+            const double phase = (isInverse ? 2.0 : -2.0) * double_Pi * i / fftSize;
+            twiddleTable[i].r = (float) cos (phase);
+            twiddleTable[i].i = (float) sin (phase);
+        }
+
+        const int root = (int) std::sqrt ((double) fftSize);
+        int divisor = 4, n = fftSize;
+
+        for (int i = 0; i < numElementsInArray (factors); ++i)
+        {
+            while ((n % divisor) != 0)
+            {
+                if (divisor == 2)       divisor = 3;
+                else if (divisor == 4)  divisor = 2;
+                else                    divisor += 2;
+
+                if (divisor > root)
+                    divisor = n;
+            }
+
+            n /= divisor;
+
+            jassert (divisor == 1 || divisor == 2 || divisor == 4);
+            factors[i].radix = divisor;
+            factors[i].length = n;
+        }
+    }
+
+    void perform (const Complex* input, Complex* output) const noexcept
+    {
+        perform (input, output, 1, 1, factors);
+    }
+
+    const int fftSize;
+    const bool inverse;
+
+    struct Factor { int radix, length; };
+    Factor factors[32];
+    HeapBlock<Complex> twiddleTable;
+
+    void perform (const Complex* input, Complex* output, const int stride, const int strideIn, const Factor* facs) const noexcept
+    {
+        const Factor factor (*facs++);
+        Complex* const originalOutput = output;
+        const Complex* const outputEnd = output + factor.radix * factor.length;
+
+        if (stride == 1 && factor.radix <= 5)
+        {
+            for (int i = 0; i < factor.radix; ++i)
+                perform (input + stride * strideIn * i, output + i * factor.length, stride * factor.radix, strideIn, facs);
+
+            butterfly (factor, output, stride);
+            return;
+        }
+
+        if (factor.length == 1)
+        {
+            do
+            {
+                *output++ = *input;
+                input += stride * strideIn;
+            }
+            while (output < outputEnd);
+        }
+        else
+        {
+            do
+            {
+                perform (input, output, stride * factor.radix, strideIn, facs);
+                input += stride * strideIn;
+                output += factor.length;
+            }
+            while (output < outputEnd);
+        }
+
+        butterfly (factor, originalOutput, stride);
+    }
+
+    void butterfly (const Factor factor, Complex* data, const int stride) const noexcept
+    {
+        switch (factor.radix)
+        {
+            case 1:   break;
+            case 2:   butterfly2 (data, stride, factor.length); return;
+            case 4:   butterfly4 (data, stride, factor.length); return;
+            default:  jassertfalse; break;
+        }
+
+        Complex* scratch = static_cast<Complex*> (alloca (sizeof (Complex) * (size_t) factor.radix));
+
+        for (int i = 0; i < factor.length; ++i)
+        {
+            for (int k = i, q1 = 0; q1 < factor.radix; ++q1)
+            {
+                scratch[q1] = data[k];
+                k += factor.length;
+            }
+
+            for (int k = i, q1 = 0; q1 < factor.radix; ++q1)
+            {
+                int twiddleIndex = 0;
+                data[k] = scratch[0];
+
+                for (int q = 1; q < factor.radix; ++q)
+                {
+                    twiddleIndex += stride * k;
+
+                    if (twiddleIndex >= fftSize)
+                        twiddleIndex -= fftSize;
+
+                    data[k] += scratch[q] * twiddleTable[twiddleIndex];
+                }
+
+                k += factor.length;
+            }
+        }
+    }
+
+    void butterfly2 (Complex* data, const int stride, const int length) const noexcept
+    {
+        Complex* dataEnd = data + length;
+        const Complex* tw = twiddleTable;
+
+        for (int i = length; --i >= 0;)
+        {
+            const Complex s (*dataEnd * *tw);
+            tw += stride;
+            *dataEnd++ = *data - s;
+            *data++ += s;
+        }
+    }
+
+    void butterfly4 (Complex* data, const int stride, const int length) const noexcept
+    {
+        const int lengthX2 = length * 2;
+        const int lengthX3 = length * 3;
+
+        const Complex* twiddle1 = twiddleTable;
+        const Complex* twiddle2 = twiddle1;
+        const Complex* twiddle3 = twiddle1;
+
+        for (int i = length; --i >= 0;)
+        {
+            const Complex s0 = data[length]   * *twiddle1;
+            const Complex s1 = data[lengthX2] * *twiddle2;
+            const Complex s2 = data[lengthX3] * *twiddle3;
+            const Complex s3 = s0 + s2;
+            const Complex s4 = s0 - s2;
+            const Complex s5 = *data - s1;
+            *data += s1;
+            data[lengthX2] = *data - s3;
+            twiddle1 += stride;
+            twiddle2 += stride * 2;
+            twiddle3 += stride * 3;
+            *data += s3;
+
+            if (inverse)
+            {
+                data[length].r   = s5.r - s4.i;
+                data[length].i   = s5.i + s4.r;
+                data[lengthX3].r = s5.r + s4.i;
+                data[lengthX3].i = s5.i - s4.r;
+            }
+            else
+            {
+                data[length].r   = s5.r + s4.i;
+                data[length].i   = s5.i - s4.r;
+                data[lengthX3].r = s5.r - s4.i;
+                data[lengthX3].i = s5.i + s4.r;
+            }
+
+            ++data;
+        }
+    }
+
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (FFTConfig)
+};
+
+
+//==============================================================================
+FFT::FFT (int order, bool inverse)  : config (new FFTConfig (1 << order, inverse)), size (1 << order) {}
+FFT::~FFT() {}
+
+void FFT::perform (const Complex* const input, Complex* const output) const noexcept
+{
+    config->perform (input, output);
+}
+
+void FFT::performRealOnlyForwardTransform (float* d) const noexcept
+{
+    // This can only be called on an FFT object that was created to do forward transforms.
+    jassert (! config->inverse);
+
+    Complex* const scratch = static_cast<Complex*> (alloca (16 + sizeof (Complex) * (size_t) size));
+
+    for (int i = 0; i < size; ++i)
+    {
+        scratch[i].r = d[i];
+        scratch[i].i = 0;
+    }
+
+    perform (scratch, reinterpret_cast<Complex*> (d));
+}
+
+void FFT::performRealOnlyInverseTransform (float* d) const noexcept
+{
+    // This can only be called on an FFT object that was created to do inverse transforms.
+    jassert (config->inverse);
+
+    Complex* const scratch = static_cast<Complex*> (alloca (16 + sizeof (Complex) * (size_t) size));
+
+    perform (reinterpret_cast<const Complex*> (d), scratch);
+
+    const float scaleFactor = 1.0f / size;
+
+    for (int i = 0; i < size; ++i)
+    {
+        d[i]        = scratch[i].r * scaleFactor;
+        d[i + size] = scratch[i].i * scaleFactor;
+    }
+}
+
+void FFT::performFrequencyOnlyForwardTransform (float* d) const noexcept
+{
+    performRealOnlyForwardTransform (d);
+    const int twiceSize = size * 2;
+
+    for (int i = 0; i < twiceSize; i += 2)
+    {
+        d[i / 2] = juce_hypot (d[i], d[i + 1]);
+
+        if (i >= size)
+        {
+            d[i] = 0;
+            d[i + 1] = 0;
+        }
+    }
+}

source/modules/juce_audio_basics/effects/juce_FFT.h

@@ -0,0 +1,92 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+/**
+    A very minimal FFT class.
+
+    This is only a simple low-footprint implementation and isn't tuned for speed - it may
+    be useful for simple applications where one of the more complex FFT libraries would be
+    overkill. (But in the future it may end up becoming optimised of course...)
+
+    The FFT class itself contains lookup tables, so there's some overhead in creating
+    one, you should create and cache an FFT object for each size/direction of transform
+    that you need, and re-use them to perform the actual operation.
+*/
+class JUCE_API  FFT
+{
+public:
+    /** Initialises an object for performing either a forward or inverse FFT with the given size.
+        The the number of points the FFT will operate on will be 2 ^ order.
+    */
+    FFT (int order, bool isInverse);
+
+    /** Destructor. */
+    ~FFT();
+
+    /** A complex number, for the purposes of the FFT class. */
+    struct Complex
+    {
+        float r;  /**< Real part. */
+        float i;  /**< Imaginary part. */
+    };
+
+    /** Performs an out-of-place FFT, either forward or inverse depending on the mode
+        that was passed to this object's constructor.
+
+        The arrays must contain at least getSize() elements.
+    */
+    void perform (const Complex* input, Complex* output) const noexcept;
+
+    /** Performs an in-place forward transform on a block of real data.
+
+        The size of the array passed in must be 2 * getSize(), and the first half
+        should contain your raw input sample data. On return, the array will contain
+        complex frequency + phase data, and can be passed to performRealOnlyInverseTransform()
+        in order to convert it back to reals.
+    */
+    void performRealOnlyForwardTransform (float* inputOutputData) const noexcept;
+
+    /** Performs a reverse operation to data created in performRealOnlyForwardTransform().
+
+        The size of the array passed in must be 2 * getSize(), containing complex
+        frequency and phase data. On return, the first half of the array will contain
+        the reconstituted samples.
+    */
+    void performRealOnlyInverseTransform (float* inputOutputData) const noexcept;
+
+    /** Takes an array and simply transforms it to the frequency spectrum.
+        This may be handy for things like frequency displays or analysis.
+    */
+    void performFrequencyOnlyForwardTransform (float* inputOutputData) const noexcept;
+
+    /** Returns the number of data points that this FFT was created to work with. */
+    int getSize() const noexcept            { return size; }
+
+private:
+    struct FFTConfig;
+    ScopedPointer<FFTConfig> config;
+    const int size;
+
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (FFT)
+};

source/modules/juce_audio_basics/effects/juce_IIRFilter.cpp

@@ -23,7 +23,7 @@
 */
 
 #if JUCE_INTEL
- #define JUCE_SNAP_TO_ZERO(n)    if (! (n < -1.0e-8 || n > 1.0e-8)) n = 0;
+ #define JUCE_SNAP_TO_ZERO(n)    if (! (n < -1.0e-8f || n > 1.0e-8f)) n = 0;
 #else
  #define JUCE_SNAP_TO_ZERO(n)
 #endif
@@ -64,7 +64,7 @@ IIRCoefficients IIRCoefficients::makeLowPass (const double sampleRate,
 {
     jassert (sampleRate > 0);
 
-    const double n = 1.0 / tan (double_Pi * frequency / sampleRate);
+    const double n = 1.0 / std::tan (double_Pi * frequency / sampleRate);
     const double nSquared = n * n;
     const double c1 = 1.0 / (1.0 + std::sqrt (2.0) * n + nSquared);
 
@@ -79,7 +79,7 @@ IIRCoefficients IIRCoefficients::makeLowPass (const double sampleRate,
 IIRCoefficients IIRCoefficients::makeHighPass (const double sampleRate,
                                                const double frequency) noexcept
 {
-    const double n = tan (double_Pi * frequency / sampleRate);
+    const double n = std::tan (double_Pi * frequency / sampleRate);
     const double nSquared = n * n;
     const double c1 = 1.0 / (1.0 + std::sqrt (2.0) * n + nSquared);
 

source/modules/juce_audio_basics/effects/juce_LagrangeInterpolator.cpp

@@ -87,7 +87,10 @@ int LagrangeInterpolator::process (const double actualRatio, const float* in,
 
         if (numOut >= 4)
         {
-            memcpy (lastInputSamples, in + (numOut - 4), 4 * sizeof (float));
+            const float* end = in + numOut;
+
+            for (int i = 0; i < 4; ++i)
+                lastInputSamples[i] = *--end;
         }
         else
         {
@@ -152,7 +155,10 @@ int LagrangeInterpolator::processAdding (const double actualRatio, const float*
 
         if (numOut >= 4)
         {
-            memcpy (lastInputSamples, in + (numOut - 4), 4 * sizeof (float));
+            const float* end = in + numOut;
+
+            for (int i = 0; i < 4; ++i)
+                lastInputSamples[i] = *--end;
         }
         else
         {

source/modules/juce_audio_basics/effects/juce_Reverb.h

@@ -82,12 +82,13 @@ public:
         const float dryScaleFactor = 2.0f;
 
         const float wet = newParams.wetLevel * wetScaleFactor;
-        wet1 = wet * (newParams.width * 0.5f + 0.5f);
-        wet2 = wet * (1.0f - newParams.width) * 0.5f;
-        dry = newParams.dryLevel * dryScaleFactor;
+        dryGain.setValue (newParams.dryLevel * dryScaleFactor);
+        wetGain1.setValue (0.5f * wet * (1.0f + newParams.width));
+        wetGain2.setValue (0.5f * wet * (1.0f - newParams.width));
+
         gain = isFrozen (newParams.freezeMode) ? 0.0f : 0.015f;
         parameters = newParams;
-        shouldUpdateDamping = true;
+        updateDamping();
     }
 
     //==============================================================================
@@ -115,7 +116,12 @@ public:
             allPass[1][i].setSize ((intSampleRate * (allPassTunings[i] + stereoSpread)) / 44100);
         }
 
-        shouldUpdateDamping = true;
+        const double smoothTime = 0.01;
+        damping .reset (sampleRate, smoothTime);
+        feedback.reset (sampleRate, smoothTime);
+        dryGain .reset (sampleRate, smoothTime);
+        wetGain1.reset (sampleRate, smoothTime);
+        wetGain2.reset (sampleRate, smoothTime);
     }
 
     /** Clears the reverb's buffers. */
@@ -137,18 +143,18 @@ public:
     {
         jassert (left != nullptr && right != nullptr);
 
-        if (shouldUpdateDamping)
-            updateDamping();
-
         for (int i = 0; i < numSamples; ++i)
         {
             const float input = (left[i] + right[i]) * gain;
             float outL = 0, outR = 0;
 
+            const float damp    = damping.getNextValue();
+            const float feedbck = feedback.getNextValue();
+
             for (int j = 0; j < numCombs; ++j)  // accumulate the comb filters in parallel
             {
-                outL += comb[0][j].process (input);
-                outR += comb[1][j].process (input);
+                outL += comb[0][j].process (input, damp, feedbck);
+                outR += comb[1][j].process (input, damp, feedbck);
             }
 
             for (int j = 0; j < numAllPasses; ++j)  // run the allpass filters in series
@@ -157,6 +163,10 @@ public:
                 outR = allPass[1][j].process (outR);
             }
 
+            const float dry  = dryGain.getNextValue();
+            const float wet1 = wetGain1.getNextValue();
+            const float wet2 = wetGain2.getNextValue();
+
             left[i]  = outL * wet1 + outR * wet2 + left[i]  * dry;
             right[i] = outR * wet1 + outL * wet2 + right[i] * dry;
         }
@@ -167,32 +177,30 @@ public:
     {
         jassert (samples != nullptr);
 
-        if (shouldUpdateDamping)
-            updateDamping();
-
         for (int i = 0; i < numSamples; ++i)
         {
             const float input = samples[i] * gain;
             float output = 0;
 
+            const float damp    = damping.getNextValue();
+            const float feedbck = feedback.getNextValue();
+
             for (int j = 0; j < numCombs; ++j)  // accumulate the comb filters in parallel
-                output += comb[0][j].process (input);
+                output += comb[0][j].process (input, damp, feedbck);
 
             for (int j = 0; j < numAllPasses; ++j)  // run the allpass filters in series
                 output = allPass[0][j].process (output);
 
+            const float dry  = dryGain.getNextValue();
+            const float wet1 = wetGain1.getNextValue();
+
             samples[i] = output * wet1 + samples[i] * dry;
         }
     }
 
 private:
     //==============================================================================
-    Parameters parameters;
-
-    volatile bool shouldUpdateDamping;
-    float gain, wet1, wet2, dry;
-
-    inline static bool isFrozen (const float freezeMode) noexcept  { return freezeMode >= 0.5f; }
+    static bool isFrozen (const float freezeMode) noexcept  { return freezeMode >= 0.5f; }
 
     void updateDamping() noexcept
     {
@@ -200,8 +208,6 @@ private:
         const float roomOffset = 0.7f;
         const float dampScaleFactor = 0.4f;
 
-        shouldUpdateDamping = false;
-
         if (isFrozen (parameters.freezeMode))
             setDamping (0.0f, 1.0f);
         else
@@ -211,19 +217,15 @@ private:
 
     void setDamping (const float dampingToUse, const float roomSizeToUse) noexcept
     {
-        for (int j = 0; j < numChannels; ++j)
-            for (int i = numCombs; --i >= 0;)
-                comb[j][i].setFeedbackAndDamp (roomSizeToUse, dampingToUse);
+        damping.setValue (dampingToUse);
+        feedback.setValue (roomSizeToUse);
     }
 
     //==============================================================================
     class CombFilter
     {
     public:
-        CombFilter() noexcept
-            : bufferSize (0), bufferIndex (0),
-              feedback (0), last (0), damp1 (0), damp2 (0)
-        {}
+        CombFilter() noexcept   : bufferSize (0), bufferIndex (0), last (0)  {}
 
         void setSize (const int size)
         {
@@ -243,22 +245,15 @@ private:
             buffer.clear ((size_t) bufferSize);
         }
 
-        void setFeedbackAndDamp (const float f, const float d) noexcept
-        {
-            damp1 = d;
-            damp2 = 1.0f - d;
-            feedback = f;
-        }
-
-        inline float process (const float input) noexcept
+        float process (const float input, const float damp, const float feedbackLevel) noexcept
         {
-            const float output = buffer [bufferIndex];
-            last = (output * damp2) + (last * damp1);
+            const float output = buffer[bufferIndex];
+            last = (output * (1.0f - damp)) + (last * damp);
             JUCE_UNDENORMALISE (last);
 
-            float temp = input + (last * feedback);
+            float temp = input + (last * feedbackLevel);
             JUCE_UNDENORMALISE (temp);
-            buffer [bufferIndex] = temp;
+            buffer[bufferIndex] = temp;
             bufferIndex = (bufferIndex + 1) % bufferSize;
             return output;
         }
@@ -266,7 +261,7 @@ private:
     private:
         HeapBlock<float> buffer;
         int bufferSize, bufferIndex;
-        float feedback, last, damp1, damp2;
+        float last;
 
         JUCE_DECLARE_NON_COPYABLE (CombFilter)
     };
@@ -294,7 +289,7 @@ private:
             buffer.clear ((size_t) bufferSize);
         }
 
-        inline float process (const float input) noexcept
+        float process (const float input) noexcept
         {
             const float bufferedValue = buffer [bufferIndex];
             float temp = input + (bufferedValue * 0.5f);
@@ -311,11 +306,65 @@ private:
         JUCE_DECLARE_NON_COPYABLE (AllPassFilter)
     };
 
+    //==============================================================================
+    class LinearSmoothedValue
+    {
+    public:
+        LinearSmoothedValue() noexcept
+            : currentValue (0), target (0), step (0), countdown (0), stepsToTarget (0)
+        {
+        }
+
+        void reset (double sampleRate, double fadeLengthSeconds) noexcept
+        {
+            jassert (sampleRate > 0 && fadeLengthSeconds >= 0);
+            stepsToTarget = (int) std::floor (fadeLengthSeconds * sampleRate);
+            currentValue = target;
+            countdown = 0;
+        }
+
+        void setValue (float newValue) noexcept
+        {
+            if (target != newValue)
+            {
+                target = newValue;
+                countdown = stepsToTarget;
+
+                if (countdown <= 0)
+                    currentValue = target;
+                else
+                    step = (target - currentValue) / countdown;
+            }
+        }
+
+        float getNextValue() noexcept
+        {
+            if (countdown <= 0)
+                return target;
+
+            --countdown;
+            currentValue += step;
+            return currentValue;
+        }
+
+    private:
+        float currentValue, target, step;
+        int countdown, stepsToTarget;
+
+        JUCE_DECLARE_NON_COPYABLE (LinearSmoothedValue)
+    };
+
+    //==============================================================================
     enum { numCombs = 8, numAllPasses = 4, numChannels = 2 };
 
+    Parameters parameters;
+    float gain;
+
     CombFilter comb [numChannels][numCombs];
     AllPassFilter allPass [numChannels][numAllPasses];
 
+    LinearSmoothedValue damping, feedback, dryGain, wetGain1, wetGain2;
+
     JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (Reverb)
 };
 

source/modules/juce_audio_basics/juce_audio_basics.cpp

@@ -78,6 +78,7 @@ namespace juce
 #include "effects/juce_IIRFilter.cpp"
 #include "effects/juce_IIRFilterOld.cpp"
 #include "effects/juce_LagrangeInterpolator.cpp"
+#include "effects/juce_FFT.cpp"
 #include "midi/juce_MidiBuffer.cpp"
 #include "midi/juce_MidiFile.cpp"
 #include "midi/juce_MidiKeyboardState.cpp"

source/modules/juce_audio_basics/juce_audio_basics.h

@@ -31,6 +31,9 @@
 namespace juce
 {
 
+#undef Complex  // apparently some C libraries actually define these symbols (!)
+#undef Factor
+
 #include "buffers/juce_AudioDataConverters.h"
 #include "buffers/juce_AudioSampleBuffer.h"
 #include "buffers/juce_FloatVectorOperations.h"
@@ -38,6 +41,7 @@ namespace juce
 #include "effects/juce_IIRFilter.h"
 #include "effects/juce_IIRFilterOld.h"
 #include "effects/juce_LagrangeInterpolator.h"
+#include "effects/juce_FFT.h"
 #include "effects/juce_Reverb.h"
 #include "midi/juce_MidiMessage.h"
 #include "midi/juce_MidiBuffer.h"

source/modules/juce_audio_basics/midi/juce_MidiMessageSequence.cpp

@@ -276,14 +276,11 @@ void MidiMessageSequence::deleteSysExMessages()
 }
 
 //==============================================================================
-void MidiMessageSequence::createControllerUpdatesForTime (const int channelNumber,
-                                                          const double time,
-                                                          OwnedArray<MidiMessage>& dest)
+void MidiMessageSequence::createControllerUpdatesForTime (const int channelNumber, const double time, Array<MidiMessage>& dest)
 {
     bool doneProg = false;
     bool donePitchWheel = false;
-    Array<int> doneControllers;
-    doneControllers.ensureStorageAllocated (32);
+    bool doneControllers[128] = { 0 };
 
     for (int i = list.size(); --i >= 0;)
     {
@@ -291,28 +288,25 @@ void MidiMessageSequence::createControllerUpdatesForTime (const int channelNumbe
 
         if (mm.isForChannel (channelNumber) && mm.getTimeStamp() <= time)
         {
-            if (mm.isProgramChange())
+            if (mm.isProgramChange() && ! doneProg)
             {
-                if (! doneProg)
-                {
-                    dest.add (new MidiMessage (mm, 0.0));
-                    doneProg = true;
-                }
+                doneProg = true;
+                dest.add (MidiMessage (mm, 0.0));
             }
-            else if (mm.isController())
+            else if (mm.isPitchWheel() && ! donePitchWheel)
             {
-                if (! doneControllers.contains (mm.getControllerNumber()))
-                {
-                    dest.add (new MidiMessage (mm, 0.0));
-                    doneControllers.add (mm.getControllerNumber());
-                }
+                donePitchWheel = true;
+                dest.add (MidiMessage (mm, 0.0));
             }
-            else if (mm.isPitchWheel())
+            else if (mm.isController())
             {
-                if (! donePitchWheel)
+                const int controllerNumber = mm.getControllerNumber();
+                jassert (isPositiveAndBelow (controllerNumber, 128));
+
+                if (! doneControllers[controllerNumber])
                 {
-                    dest.add (new MidiMessage (mm, 0.0));
-                    donePitchWheel = true;
+                    doneControllers[controllerNumber] = true;
+                    dest.add (MidiMessage (mm, 0.0));
                 }
             }
         }

source/modules/juce_audio_basics/midi/juce_MidiMessageSequence.h

@@ -247,7 +247,7 @@ public:
                                 state at the required time.
     */
     void createControllerUpdatesForTime (int channelNumber, double time,
-                                         OwnedArray<MidiMessage>& resultMessages);
+                                         Array<MidiMessage>& resultMessages);
 
     //==============================================================================
     /** Swaps this sequence with another one. */

source/modules/juce_audio_basics/sources/juce_ResamplingAudioSource.cpp

@@ -24,14 +24,14 @@
 
 ResamplingAudioSource::ResamplingAudioSource (AudioSource* const inputSource,
                                               const bool deleteInputWhenDeleted,
-                                              const int numChannels_)
+                                              const int channels)
     : input (inputSource, deleteInputWhenDeleted),
       ratio (1.0),
       lastRatio (1.0),
       bufferPos (0),
       sampsInBuffer (0),
       subSampleOffset (0),
-      numChannels (numChannels_)
+      numChannels (channels)
 {
     jassert (input != nullptr);
     zeromem (coefficients, sizeof (coefficients));
@@ -51,9 +51,10 @@ void ResamplingAudioSource::prepareToPlay (int samplesPerBlockExpected, double s
 {
     const SpinLock::ScopedLockType sl (ratioLock);
 
-    input->prepareToPlay (samplesPerBlockExpected, sampleRate);
+    const int scaledBlockSize = roundToInt (samplesPerBlockExpected * ratio);
+    input->prepareToPlay (scaledBlockSize, sampleRate * ratio);
 
-    buffer.setSize (numChannels, roundToInt (samplesPerBlockExpected * ratio) + 32);
+    buffer.setSize (numChannels, scaledBlockSize + 32);
 
     filterStates.calloc ((size_t) numChannels);
     srcBuffers.calloc ((size_t) numChannels);
@@ -93,7 +94,7 @@ void ResamplingAudioSource::getNextAudioBlock (const AudioSourceChannelInfo& inf
         lastRatio = localRatio;
     }
 
-    const int sampsNeeded = roundToInt (info.numSamples * localRatio) + 2;
+    const int sampsNeeded = roundToInt (info.numSamples * localRatio) + 3;
 
     int bufferSize = buffer.getNumSamples();
 
@@ -138,8 +139,11 @@ void ResamplingAudioSource::getNextAudioBlock (const AudioSourceChannelInfo& inf
     }
 
     int nextPos = (bufferPos + 1) % bufferSize;
+
     for (int m = info.numSamples; --m >= 0;)
     {
+        jassert (sampsInBuffer > 0 && nextPos != endOfBufferPos);
+
         const float alpha = (float) subSampleOffset;
 
         for (int channel = 0; channel < channelsToProcess; ++channel)
@@ -148,8 +152,6 @@ void ResamplingAudioSource::getNextAudioBlock (const AudioSourceChannelInfo& inf
 
         subSampleOffset += localRatio;
 
-        jassert (sampsInBuffer > 0);
-
         while (subSampleOffset >= 1.0)
         {
             if (++bufferPos >= bufferSize)

source/modules/juce_audio_basics/synthesisers/juce_Synthesiser.cpp

@@ -29,6 +29,7 @@ SynthesiserSound::~SynthesiserSound() {}
 SynthesiserVoice::SynthesiserVoice()
     : currentSampleRate (44100.0),
       currentlyPlayingNote (-1),
+      currentPlayingMidiChannel (0),
       noteOnTime (0),
       keyIsDown (false),
       sostenutoPedalDown (false)
@@ -41,8 +42,7 @@ SynthesiserVoice::~SynthesiserVoice()
 
 bool SynthesiserVoice::isPlayingChannel (const int midiChannel) const
 {
-    return currentlyPlayingSound != nullptr
-            && currentlyPlayingSound->appliesToChannel (midiChannel);
+    return currentPlayingMidiChannel == midiChannel;
 }
 
 void SynthesiserVoice::setCurrentPlaybackSampleRate (const double newRate)
@@ -50,10 +50,16 @@ void SynthesiserVoice::setCurrentPlaybackSampleRate (const double newRate)
     currentSampleRate = newRate;
 }
 
+bool SynthesiserVoice::isVoiceActive() const
+{
+    return getCurrentlyPlayingNote() >= 0;
+}
+
 void SynthesiserVoice::clearCurrentNote()
 {
     currentlyPlayingNote = -1;
     currentlyPlayingSound = nullptr;
+    currentPlayingMidiChannel = 0;
 }
 
 void SynthesiserVoice::aftertouchChanged (int) {}
@@ -253,14 +259,15 @@ void Synthesiser::startVoice (SynthesiserVoice* const voice,
         if (voice->currentlyPlayingSound != nullptr)
             voice->stopNote (0.0f, false);
 
-        voice->startNote (midiNoteNumber, velocity, sound,
-                          lastPitchWheelValues [midiChannel - 1]);
-
         voice->currentlyPlayingNote = midiNoteNumber;
+        voice->currentPlayingMidiChannel = midiChannel;
         voice->noteOnTime = ++lastNoteOnCounter;
         voice->currentlyPlayingSound = sound;
         voice->keyIsDown = true;
         voice->sostenutoPedalDown = false;
+
+        voice->startNote (midiNoteNumber, velocity, sound,
+                          lastPitchWheelValues [midiChannel - 1]);
     }
 }
 
@@ -285,7 +292,8 @@ void Synthesiser::noteOff (const int midiChannel,
     {
         SynthesiserVoice* const voice = voices.getUnchecked (i);
 
-        if (voice->getCurrentlyPlayingNote() == midiNoteNumber)
+        if (voice->getCurrentlyPlayingNote() == midiNoteNumber
+              && voice->isPlayingChannel (midiChannel))
         {
             if (SynthesiserSound* const sound = voice->getCurrentlyPlayingSound())
             {
@@ -426,7 +434,7 @@ SynthesiserVoice* Synthesiser::findFreeVoice (SynthesiserSound* soundToPlay,
     {
         SynthesiserVoice* const voice = voices.getUnchecked (i);
 
-        if (voice->getCurrentlyPlayingNote() < 0  && voice->canPlaySound (soundToPlay))
+        if ((! voice->isVoiceActive()) && voice->canPlaySound (soundToPlay))
             return voice;
     }
 
@@ -440,7 +448,7 @@ struct VoiceAgeSorter
 {
     static int compareElements (SynthesiserVoice* v1, SynthesiserVoice* v2) noexcept
     {
-        return v1->wasStartedBefore (*v2) ? 1 : (v2->wasStartedBefore (*v1) ? -1 : 0);
+        return v1->wasStartedBefore (*v2) ? -1 : (v2->wasStartedBefore (*v1) ? 1 : 0);
     }
 };
 
@@ -473,10 +481,10 @@ SynthesiserVoice* Synthesiser::findVoiceToSteal (SynthesiserSound* soundToPlay,
         }
     }
 
-    jassert (bottom != nullptr && top != nullptr);
+    const int stealableVoiceRange = usableVoices.size() - 6;
 
     // The oldest note that's playing with the target pitch playing is ideal..
-    for (int i = 0; i < usableVoices.size(); ++i)
+    for (int i = 0; i < stealableVoiceRange; ++i)
     {
         SynthesiserVoice* const voice = usableVoices.getUnchecked (i);
 
@@ -485,7 +493,7 @@ SynthesiserVoice* Synthesiser::findVoiceToSteal (SynthesiserSound* soundToPlay,
     }
 
     // ..otherwise, look for the oldest note that isn't the top or bottom note..
-    for (int i = 0; i < usableVoices.size(); ++i)
+    for (int i = 0; i < stealableVoiceRange; ++i)
     {
         SynthesiserVoice* const voice = usableVoices.getUnchecked (i);
 
@@ -493,6 +501,6 @@ SynthesiserVoice* Synthesiser::findVoiceToSteal (SynthesiserSound* soundToPlay,
             return voice;
     }
 
-    // ..otherwise, there's only one or two voices to choose from - we'll return the top one..
-    return top;
+    // ..otherwise, there's only one or two voices to choose from - we'll return the oldest one..
+    return usableVoices.getFirst();
 }

source/modules/juce_audio_basics/synthesisers/juce_Synthesiser.h

@@ -140,6 +140,12 @@ public:
     */
     virtual void stopNote (float velocity, bool allowTailOff) = 0;
 
+    /** Returns true if this voice is currently busy playing a sound.
+        By default this just checks the getCurrentlyPlayingNote() value, but can
+        be overridden for more advanced checking.
+    */
+    virtual bool isVoiceActive() const;
+
     /** Called to let the voice know that the pitch wheel has been moved.
         This will be called during the rendering callback, so must be fast and thread-safe.
     */
@@ -185,17 +191,17 @@ public:
     */
     virtual void setCurrentPlaybackSampleRate (double newRate);
 
-    /** Returns the current target sample rate at which rendering is being done.
-        Subclasses may need to know this so that they can pitch things correctly.
-    */
-    double getSampleRate() const noexcept                       { return currentSampleRate; }
-
     /** Returns true if the voice is currently playing a sound which is mapped to the given
         midi channel.
 
         If it's not currently playing, this will return false.
     */
-    bool isPlayingChannel (int midiChannel) const;
+    virtual bool isPlayingChannel (int midiChannel) const;
+
+    /** Returns the current target sample rate at which rendering is being done.
+        Subclasses may need to know this so that they can pitch things correctly.
+    */
+    double getSampleRate() const noexcept                       { return currentSampleRate; }
 
     /** Returns true if the key that triggered this voice is still held down.
         Note that the voice may still be playing after the key was released (e.g because the
@@ -230,7 +236,7 @@ private:
     friend class Synthesiser;
 
     double currentSampleRate;
-    int currentlyPlayingNote;
+    int currentlyPlayingNote, currentPlayingMidiChannel;
     uint32 noteOnTime;
     SynthesiserSound::Ptr currentlyPlayingSound;
     bool keyIsDown, sostenutoPedalDown;

source/modules/juce_audio_devices/audio_io/juce_AudioDeviceManager.cpp

@@ -92,7 +92,6 @@ AudioDeviceManager::AudioDeviceManager()
     : numInputChansNeeded (0),
       numOutputChansNeeded (2),
       listNeedsScanning (true),
-      useInputNames (false),
       inputLevel (0),
       testSoundPosition (0),
       cpuUsageMs (0),
@@ -154,7 +153,8 @@ static void addIfNotNull (OwnedArray<AudioIODeviceType>& list, AudioIODeviceType
 
 void AudioDeviceManager::createAudioDeviceTypes (OwnedArray<AudioIODeviceType>& list)
 {
-    addIfNotNull (list, AudioIODeviceType::createAudioIODeviceType_WASAPI());
+    addIfNotNull (list, AudioIODeviceType::createAudioIODeviceType_WASAPI (false));
+    addIfNotNull (list, AudioIODeviceType::createAudioIODeviceType_WASAPI (true));
     addIfNotNull (list, AudioIODeviceType::createAudioIODeviceType_DirectSound());
     addIfNotNull (list, AudioIODeviceType::createAudioIODeviceType_ASIO());
     addIfNotNull (list, AudioIODeviceType::createAudioIODeviceType_CoreAudio());
@@ -177,6 +177,17 @@ void AudioDeviceManager::addAudioDeviceType (AudioIODeviceType* newDeviceType)
     }
 }
 
+static bool deviceListContains (AudioIODeviceType* type, bool isInput, const String& name)
+{
+    StringArray devices (type->getDeviceNames (isInput));
+
+    for (int i = devices.size(); --i >= 0;)
+        if (devices[i].trim().equalsIgnoreCase (name.trim()))
+            return true;
+
+    return false;
+}
+
 //==============================================================================
 String AudioDeviceManager::initialise (const int numInputChannelsNeeded,
                                        const int numOutputChannelsNeeded,
@@ -363,8 +374,8 @@ AudioIODeviceType* AudioDeviceManager::findType (const String& inputName, const
     {
         AudioIODeviceType* const type = availableDeviceTypes.getUnchecked(i);
 
-        if ((inputName.isNotEmpty() && type->getDeviceNames (true).contains (inputName, true))
-            || (outputName.isNotEmpty() && type->getDeviceNames (false).contains (outputName, true)))
+        if ((inputName.isNotEmpty() && deviceListContains (type, true, inputName))
+             || (outputName.isNotEmpty() && deviceListContains (type, false, outputName)))
         {
             return type;
         }
@@ -458,17 +469,11 @@ String AudioDeviceManager::setAudioDeviceSetup (const AudioDeviceSetup& newSetup
         deleteCurrentDevice();
         scanDevicesIfNeeded();
 
-        if (newOutputDeviceName.isNotEmpty()
-             && ! type->getDeviceNames (false).contains (newOutputDeviceName))
-        {
+        if (newOutputDeviceName.isNotEmpty() && ! deviceListContains (type, false, newOutputDeviceName))
             return "No such device: " + newOutputDeviceName;
-        }
 
-        if (newInputDeviceName.isNotEmpty()
-             && ! type->getDeviceNames (true).contains (newInputDeviceName))
-        {
+        if (newInputDeviceName.isNotEmpty() && ! deviceListContains (type, true, newInputDeviceName))
             return "No such device: " + newInputDeviceName;
-        }
 
         currentAudioDevice = type->createDevice (newOutputDeviceName, newInputDeviceName);
 

source/modules/juce_audio_devices/audio_io/juce_AudioDeviceManager.h

@@ -450,7 +450,6 @@ private:
     BigInteger inputChannels, outputChannels;
     ScopedPointer<XmlElement> lastExplicitSettings;
     mutable bool listNeedsScanning;
-    bool useInputNames;
     Atomic<int> inputLevelMeasurementEnabledCount;
     double inputLevel;
     ScopedPointer<AudioSampleBuffer> testSound;

source/modules/juce_audio_devices/audio_io/juce_AudioIODeviceType.cpp

@@ -50,7 +50,7 @@ AudioIODeviceType* AudioIODeviceType::createAudioIODeviceType_iOSAudio()
 #endif
 
 #if ! (JUCE_WINDOWS && JUCE_WASAPI)
-AudioIODeviceType* AudioIODeviceType::createAudioIODeviceType_WASAPI()          { return nullptr; }
+AudioIODeviceType* AudioIODeviceType::createAudioIODeviceType_WASAPI (bool)     { return nullptr; }
 #endif
 
 #if ! (JUCE_WINDOWS && JUCE_DIRECTSOUND)

source/modules/juce_audio_devices/audio_io/juce_AudioIODeviceType.h

@@ -151,7 +151,7 @@ public:
     /** Creates an iOS device type if it's available on this platform, or returns null. */
     static AudioIODeviceType* createAudioIODeviceType_iOSAudio();
     /** Creates a WASAPI device type if it's available on this platform, or returns null. */
-    static AudioIODeviceType* createAudioIODeviceType_WASAPI();
+    static AudioIODeviceType* createAudioIODeviceType_WASAPI (bool exclusiveMode);
     /** Creates a DirectSound device type if it's available on this platform, or returns null. */
     static AudioIODeviceType* createAudioIODeviceType_DirectSound();
     /** Creates an ASIO device type if it's available on this platform, or returns null. */

source/modules/juce_audio_devices/juce_audio_devices.cpp

@@ -72,7 +72,7 @@
      This means that anyone who wants to use JUCE's ASIO abilities will have to:
 
      1) Agree to Steinberg's licensing terms and download the ASIO SDK
-         (see www.steinberg.net/Steinberg/Developers.asp).
+         (see http://www.steinberg.net/en/company/developers.html).
 
      2) Enable this code with a global definition #define JUCE_ASIO 1.
 

source/modules/juce_audio_devices/juce_audio_devices.h

@@ -43,12 +43,21 @@
 #endif
 
 /** Config: JUCE_WASAPI
-    Enables WASAPI audio devices (Windows Vista and above).
+    Enables WASAPI audio devices (Windows Vista and above). See also the
+    JUCE_WASAPI_EXCLUSIVE flag.
 */
 #ifndef JUCE_WASAPI
  #define JUCE_WASAPI 1
 #endif
 
+/** Config: JUCE_WASAPI_EXCLUSIVE
+    Enables WASAPI audio devices in exclusive mode (Windows Vista and above).
+*/
+#ifndef JUCE_WASAPI_EXCLUSIVE
+ #define JUCE_WASAPI_EXCLUSIVE 0
+#endif
+
+
 /** Config: JUCE_DIRECTSOUND
     Enables DirectSound audio (MS Windows only).
 */

source/modules/juce_audio_devices/midi_io/juce_MidiInput.h

@@ -139,7 +139,7 @@ public:
 
     //==============================================================================
     /** Destructor. */
-    virtual ~MidiInput();
+    ~MidiInput();
 
     /** Returns the name of this device. */
     const String& getName() const noexcept                      { return name; }
@@ -158,23 +158,21 @@ public:
 
         @see stop
     */
-    virtual void start();
+    void start();
 
     /** Stops the device running.
-
         @see start
     */
-    virtual void stop();
+    void stop();
 
-protected:
+private:
     //==============================================================================
     String name;
     void* internal;
 
-    explicit MidiInput (const String& name);
+    // The input objects are created with the openDevice() method.
+    explicit MidiInput (const String&);
 
-private:
-    //==============================================================================
     JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (MidiInput)
 };
 

source/modules/juce_audio_devices/midi_io/juce_MidiOutput.h

@@ -82,13 +82,12 @@ public:
 
     //==============================================================================
     /** Destructor. */
-    virtual ~MidiOutput();
+    ~MidiOutput();
 
     /** Makes this device output a midi message.
-
         @see MidiMessage
     */
-    virtual void sendMessageNow (const MidiMessage& message);
+    void sendMessageNow (const MidiMessage& message);
 
     //==============================================================================
     /** This lets you supply a block of messages that will be sent out at some point
@@ -100,7 +99,7 @@ public:
 
         This will only work if you've already started the thread with startBackgroundThread().
 
-        A time is supplied, at which the block of messages should be sent. This time uses
+        A time is specified, at which the block of messages should be sent. This time uses
         the same time base as Time::getMillisecondCounter(), and must be in the future.
 
         The samplesPerSecondForBuffer parameter indicates the number of samples per second
@@ -108,38 +107,34 @@ public:
         samplesPerSecondForBuffer value is needed to convert this sample position to a
         real time.
     */
-    virtual void sendBlockOfMessages (const MidiBuffer& buffer,
-                                      double millisecondCounterToStartAt,
-                                      double samplesPerSecondForBuffer);
+    void sendBlockOfMessages (const MidiBuffer& buffer,
+                              double millisecondCounterToStartAt,
+                              double samplesPerSecondForBuffer);
 
-    /** Gets rid of any midi messages that had been added by sendBlockOfMessages().
-    */
-    virtual void clearAllPendingMessages();
+    /** Gets rid of any midi messages that had been added by sendBlockOfMessages(). */
+    void clearAllPendingMessages();
 
     /** Starts up a background thread so that the device can send blocks of data.
-
         Call this to get the device ready, before using sendBlockOfMessages().
     */
-    virtual void startBackgroundThread();
+    void startBackgroundThread();
 
     /** Stops the background thread, and clears any pending midi events.
-
         @see startBackgroundThread
     */
-    virtual void stopBackgroundThread();
+    void stopBackgroundThread();
 
 
-protected:
+private:
     //==============================================================================
     void* internal;
     CriticalSection lock;
     struct PendingMessage;
     PendingMessage* firstMessage;
 
-    MidiOutput();
+    MidiOutput(); // These objects are created with the openDevice() method.
     void run() override;
 
-private:
     JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (MidiOutput)
 };
 

source/modules/juce_audio_devices/native/juce_android_OpenSL.cpp

@@ -47,13 +47,13 @@ public:
 
         // this is a total guess about how to calculate the latency, but seems to vaguely agree
         // with the devices I've tested.. YMMV
-        inputLatency  = ((javaDevice.minBufferSizeIn  * 2) / 3);
-        outputLatency = ((javaDevice.minBufferSizeOut * 2) / 3);
+        inputLatency  = (javaDevice.minBufferSizeIn  * 2) / 3;
+        outputLatency = (javaDevice.minBufferSizeOut * 2) / 3;
 
-        const int longestLatency = jmax (inputLatency, outputLatency);
-        const int totalLatency = inputLatency + outputLatency;
-        inputLatency  = ((longestLatency * inputLatency)  / totalLatency) & ~15;
-        outputLatency = ((longestLatency * outputLatency) / totalLatency) & ~15;
+        const int64 longestLatency = jmax (inputLatency, outputLatency);
+        const int64 totalLatency = inputLatency + outputLatency;
+        inputLatency  = (int) ((longestLatency * inputLatency)  / totalLatency) & ~15;
+        outputLatency = (int) ((longestLatency * outputLatency) / totalLatency) & ~15;
     }
 
     ~OpenSLAudioIODevice()

source/modules/juce_audio_devices/native/juce_linux_ALSA.cpp

@@ -30,7 +30,7 @@ namespace
 #endif
 
 #if JUCE_ALSA_LOGGING
- #define JUCE_ALSA_LOG(dbgtext)   { juce::String tempDbgBuf ("ALSA: "); tempDbgBuf << dbgtext; Logger::writeToLog (tempDbgBuf); DBG (tempDbgBuf) }
+ #define JUCE_ALSA_LOG(dbgtext)   { juce::String tempDbgBuf ("ALSA: "); tempDbgBuf << dbgtext; Logger::writeToLog (tempDbgBuf); DBG (tempDbgBuf); }
  #define JUCE_CHECKED_RESULT(x)   (logErrorMessage (x, __LINE__))
 
  static int logErrorMessage (int err, int lineNum)
@@ -149,7 +149,7 @@ class ALSADevice
 {
 public:
     ALSADevice (const String& devID, bool forInput)
-        : handle (0),
+        : handle (nullptr),
           bitDepth (16),
           numChannelsRunning (0),
           latency (0),
@@ -183,16 +183,16 @@ public:
 
     void closeNow()
     {
-        if (handle != 0)
+        if (handle != nullptr)
         {
             snd_pcm_close (handle);
-            handle = 0;
+            handle = nullptr;
         }
     }
 
     bool setParameters (unsigned int sampleRate, int numChannels, int bufferSize)
     {
-        if (handle == 0)
+        if (handle == nullptr)
             return false;
 
         JUCE_ALSA_LOG ("ALSADevice::setParameters(" << deviceID << ", "
@@ -644,8 +644,21 @@ public:
     {
         while (! threadShouldExit())
         {
-            if (inputDevice != nullptr && inputDevice->handle)
+            if (inputDevice != nullptr && inputDevice->handle != nullptr)
             {
+                if (outputDevice == nullptr || outputDevice->handle == nullptr)
+                {
+                    JUCE_ALSA_FAILED (snd_pcm_wait (inputDevice->handle, 2000));
+
+                    if (threadShouldExit())
+                        break;
+
+                    snd_pcm_sframes_t avail = snd_pcm_avail_update (inputDevice->handle);
+
+                    if (avail < 0)
+                        JUCE_ALSA_FAILED (snd_pcm_recover (inputDevice->handle, avail, 0));
+                }
+
                 audioIoInProgress = true;
 
                 if (! inputDevice->readFromInputDevice (inputChannelBuffer, bufferSize))
@@ -679,7 +692,7 @@ public:
                 }
             }
 
-            if (outputDevice != nullptr && outputDevice->handle)
+            if (outputDevice != nullptr && outputDevice->handle != nullptr)
             {
                 JUCE_ALSA_FAILED (snd_pcm_wait (outputDevice->handle, 2000));
 
@@ -702,6 +715,7 @@ public:
                 audioIoInProgress = false;
             }
         }
+
         audioIoInProgress = false;
     }
 

source/modules/juce_audio_devices/native/juce_linux_Midi.cpp

@@ -419,7 +419,7 @@ public:
         port.deletePort();
     }
 
-    void sendMessageNow (const MidiMessage& message)
+    bool sendMessageNow (const MidiMessage& message)
     {
         if (message.getRawDataSize() > maxEventSize)
         {
@@ -435,12 +435,17 @@ public:
         const uint8* data = message.getRawData();
 
         snd_seq_t* seqHandle = port.client->get();
+        bool success = true;
 
         while (numBytes > 0)
         {
             const long numSent = snd_midi_event_encode (midiParser, data, numBytes, &event);
+
             if (numSent <= 0)
+            {
+                success = numSent == 0;
                 break;
+            }
 
             numBytes -= numSent;
             data += numSent;
@@ -449,11 +454,15 @@ public:
             snd_seq_ev_set_subs (&event);
             snd_seq_ev_set_direct (&event);
 
-            snd_seq_event_output (seqHandle, &event);
+            if (snd_seq_event_output_direct (seqHandle, &event) < 0)
+            {
+                success = false;
+                break;
+            }
         }
 
-        snd_seq_drain_output (seqHandle);
         snd_midi_event_reset_encode (midiParser);
+        return success;
     }
 
 private:

source/modules/juce_audio_devices/native/juce_mac_CoreAudio.cpp

@@ -543,7 +543,7 @@ public:
         // set sample rate
         AudioObjectPropertyAddress pa;
         pa.mSelector = kAudioDevicePropertyNominalSampleRate;
-        pa.mScope = kAudioObjectPropertyScopeWildcard;
+        pa.mScope = kAudioObjectPropertyScopeGlobal;
         pa.mElement = kAudioObjectPropertyElementMaster;
         Float64 sr = newSampleRate;
 

source/modules/juce_audio_devices/native/juce_win32_ASIO.cpp

@@ -123,9 +123,9 @@ struct ASIOSampleFormat
         {
             switch (bitDepth)
             {
-                case 16: convertInt16ToFloat (static_cast <const char*> (src), dst, byteStride, samps, littleEndian); break;
-                case 24: convertInt24ToFloat (static_cast <const char*> (src), dst, byteStride, samps, littleEndian); break;
-                case 32: convertInt32ToFloat (static_cast <const char*> (src), dst, byteStride, samps, littleEndian); break;
+                case 16: convertInt16ToFloat (static_cast<const char*> (src), dst, byteStride, samps, littleEndian); break;
+                case 24: convertInt24ToFloat (static_cast<const char*> (src), dst, byteStride, samps, littleEndian); break;
+                case 32: convertInt32ToFloat (static_cast<const char*> (src), dst, byteStride, samps, littleEndian); break;
                 default: jassertfalse; break;
             }
         }
@@ -141,9 +141,9 @@ struct ASIOSampleFormat
         {
             switch (bitDepth)
             {
-                case 16: convertFloatToInt16 (src, static_cast <char*> (dst), byteStride, samps, littleEndian); break;
-                case 24: convertFloatToInt24 (src, static_cast <char*> (dst), byteStride, samps, littleEndian); break;
-                case 32: convertFloatToInt32 (src, static_cast <char*> (dst), byteStride, samps, littleEndian); break;
+                case 16: convertFloatToInt16 (src, static_cast<char*> (dst), byteStride, samps, littleEndian); break;
+                case 24: convertFloatToInt24 (src, static_cast<char*> (dst), byteStride, samps, littleEndian); break;
+                case 32: convertFloatToInt32 (src, static_cast<char*> (dst), byteStride, samps, littleEndian); break;
                 default: jassertfalse; break;
             }
         }
@@ -300,7 +300,7 @@ private:
 
 //==============================================================================
 class ASIOAudioIODevice;
-static ASIOAudioIODevice* volatile currentASIODev[3] = { 0 };
+static ASIOAudioIODevice* volatile currentASIODev[16] = { 0 };
 
 extern HWND juce_messageWindowHandle;
 
@@ -438,8 +438,6 @@ public:
         currentBlockSizeSamples = bufferSizeSamples;
         currentChansOut.clear();
         currentChansIn.clear();
-        inBuffers.clear (totalNumInputChans + 1);
-        outBuffers.clear (totalNumOutputChans + 1);
 
         updateSampleRates();
 
@@ -458,6 +456,13 @@ public:
 
         setSampleRate (sampleRate);
 
+        // (need to get this again in case a sample rate change affected the channel count)
+        err = asioObject->getChannels (&totalNumInputChans, &totalNumOutputChans);
+        jassert (err == ASE_OK);
+
+        inBuffers.calloc (totalNumInputChans + 8);
+        outBuffers.calloc (totalNumOutputChans + 8);
+
         if (needToReset)
         {
             JUCE_ASIO_LOG (" Resetting");
@@ -1320,7 +1325,7 @@ private:
                     inputFormat[i].convertToFloat (infos[i].buffers[bi], inBuffers[i], samps);
                 }
 
-                currentCallback->audioDeviceIOCallback (const_cast <const float**> (inBuffers.getData()), numActiveInputChans,
+                currentCallback->audioDeviceIOCallback (const_cast<const float**> (inBuffers.getData()), numActiveInputChans,
                                                         outBuffers, numActiveOutputChans, samps);
 
                 for (int i = 0; i < numActiveOutputChans; ++i)
@@ -1340,6 +1345,29 @@ private:
             asioObject->outputReady();
     }
 
+    long asioMessagesCallback (long selector, long value)
+    {
+        switch (selector)
+        {
+            case kAsioSelectorSupported:
+                if (value == kAsioResetRequest || value == kAsioEngineVersion || value == kAsioResyncRequest
+                     || value == kAsioLatenciesChanged || value == kAsioSupportsInputMonitor)
+                    return 1;
+                break;
+
+            case kAsioBufferSizeChange: JUCE_ASIO_LOG ("kAsioBufferSizeChange"); resetRequest(); return 1;
+            case kAsioResetRequest:     JUCE_ASIO_LOG ("kAsioResetRequest");     resetRequest(); return 1;
+            case kAsioResyncRequest:    JUCE_ASIO_LOG ("kAsioResyncRequest");    resetRequest(); return 1;
+            case kAsioLatenciesChanged: JUCE_ASIO_LOG ("kAsioLatenciesChanged"); return 1;
+            case kAsioEngineVersion:    return 2;
+
+            case kAsioSupportsTimeInfo:
+            case kAsioSupportsTimeCode: return 0;
+        }
+
+        return 0;
+    }
+
     //==============================================================================
     template <int deviceIndex>
     struct ASIOCallbackFunctions
@@ -1360,56 +1388,43 @@ private:
 
         static long JUCE_ASIOCALLBACK asioMessagesCallback (long selector, long value, void*, double*)
         {
-            switch (selector)
-            {
-                case kAsioSelectorSupported:
-                    if (value == kAsioResetRequest || value == kAsioEngineVersion || value == kAsioResyncRequest
-                         || value == kAsioLatenciesChanged || value == kAsioSupportsInputMonitor)
-                        return 1;
-                    break;
-
-                case kAsioBufferSizeChange: JUCE_ASIO_LOG ("kAsioBufferSizeChange"); return sendResetRequest (deviceIndex);
-                case kAsioResetRequest:     JUCE_ASIO_LOG ("kAsioResetRequest");     return sendResetRequest (deviceIndex);
-                case kAsioResyncRequest:    JUCE_ASIO_LOG ("kAsioResyncRequest");    return sendResetRequest (deviceIndex);
-                case kAsioLatenciesChanged: JUCE_ASIO_LOG ("kAsioLatenciesChanged"); return 1;
-                case kAsioEngineVersion:    return 2;
-
-                case kAsioSupportsTimeInfo:
-                case kAsioSupportsTimeCode:
-                    return 0;
-            }
-
-            return 0;
+            return currentASIODev[deviceIndex] != nullptr
+                     ? currentASIODev[deviceIndex]->asioMessagesCallback (selector, value)
+                     : 0;
         }
 
         static void JUCE_ASIOCALLBACK sampleRateChangedCallback (ASIOSampleRate)
         {
-            sendResetRequest (deviceIndex);
+            if (currentASIODev[deviceIndex] != nullptr)
+                currentASIODev[deviceIndex]->resetRequest();
         }
 
-        static long sendResetRequest (int index)
+        static void setCallbacks (ASIOCallbacks& callbacks) noexcept
         {
-            if (currentASIODev[index] != nullptr)
-                currentASIODev[index]->resetRequest();
-
-            return 1;
+            callbacks.bufferSwitch          = &bufferSwitchCallback;
+            callbacks.asioMessage           = &asioMessagesCallback;
+            callbacks.bufferSwitchTimeInfo  = &bufferSwitchTimeInfoCallback;
+            callbacks.sampleRateDidChange   = &sampleRateChangedCallback;
         }
 
-        static void setCallbacks (ASIOCallbacks& callbacks)
+        static void setCallbacksForDevice (ASIOCallbacks& callbacks, ASIOAudioIODevice* device) noexcept
         {
-            callbacks.bufferSwitch = &bufferSwitchCallback;
-            callbacks.asioMessage = &asioMessagesCallback;
-            callbacks.bufferSwitchTimeInfo = &bufferSwitchTimeInfoCallback;
-            callbacks.sampleRateDidChange = &sampleRateChangedCallback;
+            if (currentASIODev[deviceIndex] == device)
+                setCallbacks (callbacks);
+            else
+                ASIOCallbackFunctions<deviceIndex + 1>::setCallbacksForDevice (callbacks, device);
         }
     };
 
-    void setCallbackFunctions()
+    template <>
+    struct ASIOCallbackFunctions <sizeof(currentASIODev) / sizeof(currentASIODev[0])>
+    {
+        static void setCallbacksForDevice (ASIOCallbacks&, ASIOAudioIODevice*) noexcept {}
+    };
+
+    void setCallbackFunctions() noexcept
     {
-        if      (currentASIODev[0] == this)  ASIOCallbackFunctions<0>::setCallbacks (callbacks);
-        else if (currentASIODev[1] == this)  ASIOCallbackFunctions<1>::setCallbacks (callbacks);
-        else if (currentASIODev[2] == this)  ASIOCallbackFunctions<2>::setCallbacks (callbacks);
-        else                                 jassertfalse;
+        ASIOCallbackFunctions<0>::setCallbacksForDevice (callbacks, this);
     }
 
     JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (ASIOAudioIODevice)
@@ -1503,15 +1518,16 @@ public:
         jassert (inputDeviceName == outputDeviceName || outputDeviceName.isEmpty() || inputDeviceName.isEmpty());
         jassert (hasScanned); // need to call scanForDevices() before doing this
 
-        const int index = deviceNames.indexOf (outputDeviceName.isNotEmpty() ? outputDeviceName
-                                                                             : inputDeviceName);
+        const String deviceName (outputDeviceName.isNotEmpty() ? outputDeviceName
+                                                               : inputDeviceName);
+        const int index = deviceNames.indexOf (deviceName);
 
         if (index >= 0)
         {
             const int freeSlot = findFreeSlot();
 
             if (freeSlot >= 0)
-                return new ASIOAudioIODevice (this, outputDeviceName,
+                return new ASIOAudioIODevice (this, deviceName,
                                               classIds.getReference (index), freeSlot);
         }
 

source/modules/juce_audio_devices/native/juce_win32_DirectSound.cpp

@@ -1255,7 +1255,7 @@ private:
     DSoundDeviceList deviceList;
     bool hasScanned;
 
-    void systemDeviceChanged()
+    void systemDeviceChanged() override
     {
         DSoundDeviceList newList;
         newList.scan();

source/modules/juce_audio_devices/native/juce_win32_WASAPI.cpp

@@ -45,6 +45,7 @@ void logFailure (HRESULT hr)
         {
             case E_POINTER:     m = "E_POINTER"; break;
             case E_INVALIDARG:  m = "E_INVALIDARG"; break;
+            case E_NOINTERFACE: m = "E_NOINTERFACE"; break;
 
             #define JUCE_WASAPI_ERR(desc, n) \
                 case MAKE_HRESULT(1, 0x889, n): m = #desc; break;
@@ -126,7 +127,11 @@ enum EDataFlow
     eAll = (eCapture + 1)
 };
 
-enum { DEVICE_STATE_ACTIVE = 1 };
+enum
+{
+    DEVICE_STATE_ACTIVE = 1,
+    AUDCLNT_BUFFERFLAGS_SILENT = 2
+};
 
 JUCE_IUNKNOWNCLASS (IPropertyStore, "886d8eeb-8cf2-4446-8d02-cdba1dbdcf99")
 {
@@ -328,6 +333,11 @@ int refTimeToSamples (const REFERENCE_TIME& t, const double sampleRate) noexcept
     return roundToInt (sampleRate * ((double) t) * 0.0000001);
 }
 
+REFERENCE_TIME samplesToRefTime (const int numSamples, const double sampleRate) noexcept
+{
+    return (REFERENCE_TIME) ((numSamples * 10000.0 * 1000.0 / sampleRate) + 0.5);
+}
+
 void copyWavFormat (WAVEFORMATEXTENSIBLE& dest, const WAVEFORMATEX* const src) noexcept
 {
     memcpy (&dest, src, src->wFormatTag == WAVE_FORMAT_EXTENSIBLE ? sizeof (WAVEFORMATEXTENSIBLE)
@@ -348,9 +358,12 @@ public:
           defaultBufferSize (0),
           latencySamples (0),
           useExclusiveMode (exclusiveMode),
+          actualBufferSize (0),
+          bytesPerSample (0),
+          bytesPerFrame (0),
           sampleRateHasChanged (false)
     {
-        clientEvent = CreateEvent (0, false, false, _T("JuceWASAPI"));
+        clientEvent = CreateEvent (nullptr, false, false, nullptr);
 
         ComSmartPtr<IAudioClient> tempClient (createClient());
         if (tempClient == nullptr)
@@ -376,16 +389,24 @@ public:
 
         rates.addUsingDefaultSort (defaultSampleRate);
 
-        static const int ratesToTest[] = { 44100, 48000, 88200, 96000, 176400, 192000 };
+        if (useExclusiveMode
+             && findSupportedFormat (tempClient, defaultSampleRate, format.dwChannelMask, format))
+        {
+            // Got a format that is supported by the device so we can ask what sample rates are supported (in whatever format)
+        }
+
+        static const int ratesToTest[] = { 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 };
 
         for (int i = 0; i < numElementsInArray (ratesToTest); ++i)
         {
-            if (ratesToTest[i] == defaultSampleRate)
+            if (rates.contains (ratesToTest[i]))
                 continue;
 
-            format.Format.nSamplesPerSec = (DWORD) ratesToTest[i];
+            format.Format.nSamplesPerSec  = (DWORD) ratesToTest[i];
+            format.Format.nAvgBytesPerSec = (DWORD) (format.Format.nSamplesPerSec * format.Format.nChannels * format.Format.wBitsPerSample / 8);
 
-            if (SUCCEEDED (tempClient->IsFormatSupported (useExclusiveMode ? AUDCLNT_SHAREMODE_EXCLUSIVE : AUDCLNT_SHAREMODE_SHARED,
+            if (SUCCEEDED (tempClient->IsFormatSupported (useExclusiveMode ? AUDCLNT_SHAREMODE_EXCLUSIVE
+                                                                           : AUDCLNT_SHAREMODE_SHARED,
                                                           (WAVEFORMATEX*) &format, 0)))
                 if (! rates.contains (ratesToTest[i]))
                     rates.addUsingDefaultSort (ratesToTest[i]);
@@ -400,7 +421,7 @@ public:
 
     bool isOk() const noexcept   { return defaultBufferSize > 0 && defaultSampleRate > 0; }
 
-    bool openClient (const double newSampleRate, const BigInteger& newChannels)
+    bool openClient (const double newSampleRate, const BigInteger& newChannels, const int bufferSizeSamples)
     {
         sampleRate = newSampleRate;
         channels = newChannels;
@@ -413,8 +434,7 @@ public:
         client = createClient();
 
         if (client != nullptr
-             && (tryInitialisingWithFormat (true, 4) || tryInitialisingWithFormat (false, 4)
-                  || tryInitialisingWithFormat (false, 3) || tryInitialisingWithFormat (false, 2)))
+             && tryInitialisingWithBufferSize (bufferSizeSamples))
         {
             sampleRateHasChanged = false;
 
@@ -465,7 +485,7 @@ public:
     BigInteger channels;
     Array<int> channelMaps;
     UINT32 actualBufferSize;
-    int bytesPerSample;
+    int bytesPerSample, bytesPerFrame;
     bool sampleRateHasChanged;
 
     virtual void updateFormat (bool isFloat) = 0;
@@ -536,12 +556,19 @@ private:
         return client;
     }
 
-    bool tryInitialisingWithFormat (const bool useFloat, const int bytesPerSampleToTry)
+    struct AudioSampleFormat
+    {
+        bool useFloat;
+        int  bitsPerSampleToTry;
+        int  bytesPerSampleContainer;
+    };
+
+    bool tryFormat (const AudioSampleFormat sampleFormat, IAudioClient* clientToUse, double sampleRate,
+                    DWORD mixFormatChannelMask, WAVEFORMATEXTENSIBLE& format) const
     {
-        WAVEFORMATEXTENSIBLE format;
         zerostruct (format);
 
-        if (numChannels <= 2 && bytesPerSampleToTry <= 2)
+        if (numChannels <= 2 && sampleFormat.bitsPerSampleToTry <= 16)
         {
             format.Format.wFormatTag = WAVE_FORMAT_PCM;
         }
@@ -553,45 +580,97 @@ private:
 
         format.Format.nSamplesPerSec       = (DWORD) sampleRate;
         format.Format.nChannels            = (WORD) numChannels;
-        format.Format.wBitsPerSample       = (WORD) (8 * bytesPerSampleToTry);
-        format.Format.nAvgBytesPerSec      = (DWORD) (format.Format.nSamplesPerSec * numChannels * bytesPerSampleToTry);
-        format.Format.nBlockAlign          = (WORD) (numChannels * bytesPerSampleToTry);
-        format.SubFormat                   = useFloat ? KSDATAFORMAT_SUBTYPE_IEEE_FLOAT : KSDATAFORMAT_SUBTYPE_PCM;
-        format.Samples.wValidBitsPerSample = format.Format.wBitsPerSample;
+        format.Format.wBitsPerSample       = (WORD) (8 * sampleFormat.bytesPerSampleContainer);
+        format.Samples.wValidBitsPerSample = (WORD) (sampleFormat.bitsPerSampleToTry);
+        format.Format.nBlockAlign          = (WORD) (format.Format.nChannels * format.Format.wBitsPerSample / 8);
+        format.Format.nAvgBytesPerSec      = (DWORD) (format.Format.nSamplesPerSec * format.Format.nBlockAlign);
+        format.SubFormat                   = sampleFormat.useFloat ? KSDATAFORMAT_SUBTYPE_IEEE_FLOAT : KSDATAFORMAT_SUBTYPE_PCM;
         format.dwChannelMask               = mixFormatChannelMask;
 
         WAVEFORMATEXTENSIBLE* nearestFormat = nullptr;
 
-        HRESULT hr = client->IsFormatSupported (useExclusiveMode ? AUDCLNT_SHAREMODE_EXCLUSIVE
-                                                                 : AUDCLNT_SHAREMODE_SHARED,
-                                                (WAVEFORMATEX*) &format,
-                                                useExclusiveMode ? nullptr : (WAVEFORMATEX**) &nearestFormat);
+        HRESULT hr = clientToUse->IsFormatSupported (useExclusiveMode ? AUDCLNT_SHAREMODE_EXCLUSIVE
+                                                                      : AUDCLNT_SHAREMODE_SHARED,
+                                                     (WAVEFORMATEX*) &format,
+                                                     useExclusiveMode ? nullptr : (WAVEFORMATEX**) &nearestFormat);
         logFailure (hr);
 
         if (hr == S_FALSE && format.Format.nSamplesPerSec == nearestFormat->Format.nSamplesPerSec)
         {
-            copyWavFormat (format, (WAVEFORMATEX*) nearestFormat);
+            copyWavFormat (format, (const WAVEFORMATEX*) nearestFormat);
             hr = S_OK;
         }
 
         CoTaskMemFree (nearestFormat);
+        return check (hr);
+    }
 
-        REFERENCE_TIME defaultPeriod = 0, minPeriod = 0;
-        if (useExclusiveMode)
-            check (client->GetDevicePeriod (&defaultPeriod, &minPeriod));
+    bool findSupportedFormat (IAudioClient* clientToUse, double sampleRate,
+                              DWORD mixFormatChannelMask, WAVEFORMATEXTENSIBLE& format) const
+    {
+        static const AudioSampleFormat formats[] =
+        {
+            { true,  32, 4 },
+            { false, 32, 4 },
+            { false, 24, 4 },
+            { false, 24, 3 },
+            { false, 20, 4 },
+            { false, 20, 3 },
+            { false, 16, 2 }
+        };
+
+        for (int i = 0; i < numElementsInArray (formats); ++i)
+            if (tryFormat (formats[i], clientToUse, sampleRate, mixFormatChannelMask, format))
+                return true;
+
+        return false;
+    }
 
-        GUID session;
-        if (hr == S_OK
-             && check (client->Initialize (useExclusiveMode ? AUDCLNT_SHAREMODE_EXCLUSIVE : AUDCLNT_SHAREMODE_SHARED,
-                                           0x40000 /*AUDCLNT_STREAMFLAGS_EVENTCALLBACK*/,
-                                           defaultPeriod, defaultPeriod, (WAVEFORMATEX*) &format, &session)))
+    bool tryInitialisingWithBufferSize (const int bufferSizeSamples)
+    {
+        WAVEFORMATEXTENSIBLE format;
+
+        if (findSupportedFormat (client, sampleRate, mixFormatChannelMask, format))
         {
-            actualNumChannels = format.Format.nChannels;
-            const bool isFloat = format.Format.wFormatTag == WAVE_FORMAT_EXTENSIBLE && format.SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT;
-            bytesPerSample = format.Format.wBitsPerSample / 8;
+            REFERENCE_TIME defaultPeriod = 0, minPeriod = 0;
 
-            updateFormat (isFloat);
-            return true;
+            check (client->GetDevicePeriod (&defaultPeriod, &minPeriod));
+
+            if (useExclusiveMode && bufferSizeSamples > 0)
+                defaultPeriod = jmax (minPeriod, samplesToRefTime (bufferSizeSamples, format.Format.nSamplesPerSec));
+
+            for (;;)
+            {
+                GUID session;
+                HRESULT hr = client->Initialize (useExclusiveMode ? AUDCLNT_SHAREMODE_EXCLUSIVE : AUDCLNT_SHAREMODE_SHARED,
+                                                 0x40000 /*AUDCLNT_STREAMFLAGS_EVENTCALLBACK*/,
+                                                 defaultPeriod, useExclusiveMode ? defaultPeriod : 0, (WAVEFORMATEX*) &format, &session);
+
+                if (check (hr))
+                {
+                    actualNumChannels  = format.Format.nChannels;
+                    const bool isFloat = format.Format.wFormatTag == WAVE_FORMAT_EXTENSIBLE && format.SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT;
+                    bytesPerSample     = format.Format.wBitsPerSample / 8;
+                    bytesPerFrame      = format.Format.nBlockAlign;
+
+                    updateFormat (isFloat);
+                    return true;
+                }
+
+                // Handle the "alignment dance" : http://msdn.microsoft.com/en-us/library/windows/desktop/dd370875(v=vs.85).aspx (see Remarks)
+                if (hr != MAKE_HRESULT (1, 0x889, 0x19)) // AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED
+                    break;
+
+                UINT32 numFrames = 0;
+                if (! check (client->GetBufferSize (&numFrames)))
+                    break;
+
+                // Recreate client
+                client = nullptr;
+                client = createClient();
+
+                defaultPeriod = samplesToRefTime (numFrames, format.Format.nSamplesPerSec);
+            }
         }
 
         return false;
@@ -615,12 +694,9 @@ public:
         close();
     }
 
-    bool open (const double newSampleRate, const BigInteger& newChannels)
+    bool open (const double newSampleRate, const BigInteger& newChannels, int bufferSizeSamples)
     {
-        reservoirSize = 0;
-        reservoirCapacity = 16384;
-        reservoir.setSize (actualNumChannels * reservoirCapacity * sizeof (float));
-        return openClient (newSampleRate, newChannels)
+        return openClient (newSampleRate, newChannels, bufferSizeSamples)
                 && (numChannels == 0 || check (client->GetService (__uuidof (IAudioCaptureClient),
                                                                    (void**) captureClient.resetAndGetPointerAddress())));
     }
@@ -630,88 +706,129 @@ public:
         closeClient();
         captureClient = nullptr;
         reservoir.reset();
+        reservoirReadPos = reservoirWritePos = 0;
     }
 
     template<class SourceType>
-    void updateFormatWithType (SourceType*)
+    void updateFormatWithType (SourceType*) noexcept
     {
         typedef AudioData::Pointer<AudioData::Float32, AudioData::NativeEndian, AudioData::NonInterleaved, AudioData::NonConst> NativeType;
         converter = new AudioData::ConverterInstance<AudioData::Pointer<SourceType, AudioData::LittleEndian, AudioData::Interleaved, AudioData::Const>, NativeType> (actualNumChannels, 1);
     }
 
-    void updateFormat (bool isFloat)
+    void updateFormat (bool isFloat) override
+    {
+        if (isFloat)                    updateFormatWithType ((AudioData::Float32*) nullptr);
+        else if (bytesPerSample == 4)   updateFormatWithType ((AudioData::Int32*)   nullptr);
+        else if (bytesPerSample == 3)   updateFormatWithType ((AudioData::Int24*)   nullptr);
+        else                            updateFormatWithType ((AudioData::Int16*)   nullptr);
+    }
+
+    bool start (const int userBufferSize)
+    {
+        reservoirSize = actualBufferSize + userBufferSize;
+        reservoirMask = nextPowerOfTwo (reservoirSize) - 1;
+        reservoir.setSize ((reservoirMask + 1) * bytesPerFrame, true);
+        reservoirReadPos = reservoirWritePos = 0;
+
+        if (! check (client->Start()))
+            return false;
+
+        purgeInputBuffers();
+        return true;
+    }
+
+    void purgeInputBuffers()
     {
-        if (isFloat)                    updateFormatWithType ((AudioData::Float32*) 0);
-        else if (bytesPerSample == 4)   updateFormatWithType ((AudioData::Int32*) 0);
-        else if (bytesPerSample == 3)   updateFormatWithType ((AudioData::Int24*) 0);
-        else                            updateFormatWithType ((AudioData::Int16*) 0);
+        uint8* inputData;
+        UINT32 numSamplesAvailable;
+        DWORD flags;
+
+        while (captureClient->GetBuffer (&inputData, &numSamplesAvailable, &flags, nullptr, nullptr)
+                  != MAKE_HRESULT (0, 0x889, 0x1) /* AUDCLNT_S_BUFFER_EMPTY */)
+            captureClient->ReleaseBuffer (numSamplesAvailable);
     }
 
-    void copyBuffers (float** destBuffers, int numDestBuffers, int bufferSize, Thread& thread)
+    int getNumSamplesInReservoir() const noexcept    { return reservoirWritePos - reservoirReadPos; }
+
+    void handleDeviceBuffer()
     {
         if (numChannels <= 0)
             return;
 
-        int offset = 0;
+        uint8* inputData;
+        UINT32 numSamplesAvailable;
+        DWORD flags;
 
-        while (bufferSize > 0)
+        while (check (captureClient->GetBuffer (&inputData, &numSamplesAvailable, &flags, nullptr, nullptr)) && numSamplesAvailable > 0)
         {
-            if (reservoirSize > 0)  // There's stuff in the reservoir, so use that...
+            int samplesLeft = (int) numSamplesAvailable;
+
+            while (samplesLeft > 0)
             {
-                const int samplesToDo = jmin (bufferSize, (int) reservoirSize);
+                const int localWrite = reservoirWritePos & reservoirMask;
+                const int samplesToDo = jmin (samplesLeft, reservoirMask + 1 - localWrite);
+                const int samplesToDoBytes = samplesToDo * bytesPerFrame;
 
-                for (int i = 0; i < numDestBuffers; ++i)
-                    converter->convertSamples (destBuffers[i] + offset, 0, reservoir.getData(), channelMaps.getUnchecked(i), samplesToDo);
+                void* reservoirPtr = addBytesToPointer (reservoir.getData(), localWrite * bytesPerFrame);
 
-                bufferSize -= samplesToDo;
-                offset += samplesToDo;
-                reservoirSize = 0;
+                if ((flags & AUDCLNT_BUFFERFLAGS_SILENT) != 0)
+                    zeromem (reservoirPtr, samplesToDoBytes);
+                else
+                    memcpy (reservoirPtr, inputData, samplesToDoBytes);
+
+                reservoirWritePos += samplesToDo;
+                inputData += samplesToDoBytes;
+                samplesLeft -= samplesToDo;
             }
-            else
-            {
-                UINT32 packetLength = 0;
-                if (! check (captureClient->GetNextPacketSize (&packetLength)))
-                    break;
 
-                if (packetLength == 0)
-                {
-                    if (thread.threadShouldExit()
-                         || WaitForSingleObject (clientEvent, 1000) == WAIT_TIMEOUT)
-                        break;
+            if (getNumSamplesInReservoir() > reservoirSize)
+                reservoirReadPos = reservoirWritePos - reservoirSize;
 
-                    continue;
-                }
+            captureClient->ReleaseBuffer (numSamplesAvailable);
+        }
+    }
 
-                uint8* inputData;
-                UINT32 numSamplesAvailable;
-                DWORD flags;
+    void copyBuffersFromReservoir (float** destBuffers, int numDestBuffers, int bufferSize)
+    {
+        if ((numChannels <= 0 && bufferSize == 0) || reservoir.getSize() == 0)
+            return;
 
-                if (check (captureClient->GetBuffer (&inputData, &numSamplesAvailable, &flags, 0, 0)))
-                {
-                    const int samplesToDo = jmin (bufferSize, (int) numSamplesAvailable);
+        int offset = jmax (0, bufferSize - getNumSamplesInReservoir());
 
-                    for (int i = 0; i < numDestBuffers; ++i)
-                        converter->convertSamples (destBuffers[i] + offset, 0, inputData, channelMaps.getUnchecked(i), samplesToDo);
+        if (offset > 0)
+        {
+            for (int i = 0; i < numDestBuffers; ++i)
+                zeromem (destBuffers[i], offset * sizeof (float));
 
-                    bufferSize -= samplesToDo;
-                    offset += samplesToDo;
+            bufferSize -= offset;
+            reservoirReadPos -= offset / 2;
+        }
 
-                    if (samplesToDo < (int) numSamplesAvailable)
-                    {
-                        reservoirSize = jmin ((int) (numSamplesAvailable - samplesToDo), reservoirCapacity);
-                        memcpy ((uint8*) reservoir.getData(), inputData + bytesPerSample * actualNumChannels * samplesToDo,
-                                (size_t) (bytesPerSample * actualNumChannels * reservoirSize));
-                    }
+        while (bufferSize > 0)
+        {
+            const int localRead = reservoirReadPos & reservoirMask;
 
-                    captureClient->ReleaseBuffer (numSamplesAvailable);
-                }
-            }
+            const int samplesToDo = jmin (bufferSize, getNumSamplesInReservoir(), reservoirMask + 1 - localRead);
+            if (samplesToDo <= 0)
+                break;
+
+            const int reservoirOffset = localRead * bytesPerFrame;
+
+            for (int i = 0; i < numDestBuffers; ++i)
+                converter->convertSamples (destBuffers[i] + offset, 0, addBytesToPointer (reservoir.getData(), reservoirOffset), channelMaps.getUnchecked(i), samplesToDo);
+
+            bufferSize -= samplesToDo;
+            offset += samplesToDo;
+            reservoirReadPos += samplesToDo;
         }
     }
 
     ComSmartPtr<IAudioCaptureClient> captureClient;
     MemoryBlock reservoir;
-    int reservoirSize, reservoirCapacity;
+    int reservoirSize, reservoirMask;
+    volatile int reservoirReadPos, reservoirWritePos;
+
     ScopedPointer<AudioData::Converter> converter;
 
 private:
@@ -732,10 +849,11 @@ public:
         close();
     }
 
-    bool open (const double newSampleRate, const BigInteger& newChannels)
+    bool open (const double newSampleRate, const BigInteger& newChannels, int bufferSizeSamples)
     {
-        return openClient (newSampleRate, newChannels)
-            && (numChannels == 0 || check (client->GetService (__uuidof (IAudioRenderClient), (void**) renderClient.resetAndGetPointerAddress())));
+        return openClient (newSampleRate, newChannels, bufferSizeSamples)
+                && (numChannels == 0 || check (client->GetService (__uuidof (IAudioRenderClient),
+                                                                   (void**) renderClient.resetAndGetPointerAddress())));
     }
 
     void close()
@@ -751,15 +869,42 @@ public:
         converter = new AudioData::ConverterInstance<NativeType, AudioData::Pointer<DestType, AudioData::LittleEndian, AudioData::Interleaved, AudioData::NonConst> > (1, actualNumChannels);
     }
 
-    void updateFormat (bool isFloat)
+    void updateFormat (bool isFloat) override
+    {
+        if (isFloat)                    updateFormatWithType ((AudioData::Float32*) nullptr);
+        else if (bytesPerSample == 4)   updateFormatWithType ((AudioData::Int32*)   nullptr);
+        else if (bytesPerSample == 3)   updateFormatWithType ((AudioData::Int24*)   nullptr);
+        else                            updateFormatWithType ((AudioData::Int16*)   nullptr);
+    }
+
+    bool start()
     {
-        if (isFloat)                    updateFormatWithType ((AudioData::Float32*) 0);
-        else if (bytesPerSample == 4)   updateFormatWithType ((AudioData::Int32*) 0);
-        else if (bytesPerSample == 3)   updateFormatWithType ((AudioData::Int24*) 0);
-        else                            updateFormatWithType ((AudioData::Int16*) 0);
+        int samplesToDo = getNumSamplesAvailableToCopy();
+        uint8* outputData;
+
+        if (check (renderClient->GetBuffer (samplesToDo, &outputData)))
+            renderClient->ReleaseBuffer (samplesToDo, AUDCLNT_BUFFERFLAGS_SILENT);
+
+        return check (client->Start());
+    }
+
+    int getNumSamplesAvailableToCopy() const
+    {
+        if (numChannels <= 0)
+            return 0;
+
+        if (! useExclusiveMode)
+        {
+            UINT32 padding = 0;
+            if (check (client->GetCurrentPadding (&padding)))
+                return actualBufferSize - (int) padding;
+        }
+
+        return actualBufferSize;
     }
 
-    void copyBuffers (const float** const srcBuffers, const int numSrcBuffers, int bufferSize, Thread& thread)
+    void copyBuffers (const float** const srcBuffers, const int numSrcBuffers, int bufferSize,
+                      WASAPIInputDevice* inputDevice, Thread& thread)
     {
         if (numChannels <= 0)
             return;
@@ -768,22 +913,25 @@ public:
 
         while (bufferSize > 0)
         {
-            UINT32 padding = 0;
-            if (! check (client->GetCurrentPadding (&padding)))
-                return;
+            // This is needed in order not to drop any input data if the output device endpoint buffer was full
+            if ((! useExclusiveMode) && inputDevice != nullptr
+                  && WaitForSingleObject (inputDevice->clientEvent, 0) == WAIT_OBJECT_0)
+                inputDevice->handleDeviceBuffer();
 
-            int samplesToDo = useExclusiveMode ? bufferSize
-                                               : jmin ((int) (actualBufferSize - padding), bufferSize);
+            int samplesToDo = jmin (getNumSamplesAvailableToCopy(), bufferSize);
 
-            if (samplesToDo <= 0)
+            if (samplesToDo == 0)
             {
-                if (thread.threadShouldExit()
-                     || WaitForSingleObject (clientEvent, 1000) == WAIT_TIMEOUT)
-                    break;
+                // This can ONLY occur in non-exclusive mode
+                if (! thread.threadShouldExit() && WaitForSingleObject (clientEvent, 1000) == WAIT_OBJECT_0)
+                    continue;
 
-                continue;
+                break;
             }
 
+            if (useExclusiveMode && WaitForSingleObject (clientEvent, 1000) == WAIT_TIMEOUT)
+                break;
+
             uint8* outputData = nullptr;
             if (check (renderClient->GetBuffer ((UINT32) samplesToDo, &outputData)))
             {
@@ -791,10 +939,10 @@ public:
                     converter->convertSamples (outputData, channelMaps.getUnchecked(i), srcBuffers[i] + offset, 0, samplesToDo);
 
                 renderClient->ReleaseBuffer ((UINT32) samplesToDo, 0);
-
-                offset += samplesToDo;
-                bufferSize -= samplesToDo;
             }
+
+            bufferSize -= samplesToDo;
+            offset += samplesToDo;
         }
     }
 
@@ -812,13 +960,14 @@ class WASAPIAudioIODevice  : public AudioIODevice,
 {
 public:
     WASAPIAudioIODevice (const String& deviceName,
-                         const String& outputDeviceId_,
-                         const String& inputDeviceId_,
+                         const String& typeName,
+                         const String& outputDeviceID,
+                         const String& inputDeviceID,
                          const bool exclusiveMode)
-        : AudioIODevice (deviceName, "Windows Audio"),
+        : AudioIODevice (deviceName, typeName),
           Thread ("Juce WASAPI"),
-          outputDeviceId (outputDeviceId_),
-          inputDeviceId (inputDeviceId_),
+          outputDeviceId (outputDeviceID),
+          inputDeviceId (inputDeviceID),
           useExclusiveMode (exclusiveMode),
           isOpen_ (false),
           isStarted (false),
@@ -932,19 +1081,33 @@ public:
         lastKnownInputChannels    = inputChannels;
         lastKnownOutputChannels   = outputChannels;
 
-        if (inputDevice != nullptr && ! inputDevice->open (currentSampleRate, inputChannels))
+        if (inputDevice != nullptr && ! inputDevice->open (currentSampleRate, inputChannels, bufferSizeSamples))
         {
             lastError = TRANS("Couldn't open the input device!");
             return lastError;
         }
 
-        if (outputDevice != nullptr && ! outputDevice->open (currentSampleRate, outputChannels))
+        if (outputDevice != nullptr && ! outputDevice->open (currentSampleRate, outputChannels, bufferSizeSamples))
         {
             close();
             lastError = TRANS("Couldn't open the output device!");
             return lastError;
         }
 
+        if (useExclusiveMode)
+        {
+            // This is to make sure that the callback uses actualBufferSize in case of exclusive mode
+            if (inputDevice != nullptr && outputDevice != nullptr && inputDevice->actualBufferSize != outputDevice->actualBufferSize)
+            {
+                close();
+                lastError = TRANS("Couldn't open the output device (buffer size mismatch)");
+                return lastError;
+            }
+
+            currentBufferSizeSamples = outputDevice != nullptr ? outputDevice->actualBufferSize
+                                                               : inputDevice->actualBufferSize;
+        }
+
         if (inputDevice != nullptr)   ResetEvent (inputDevice->clientEvent);
         if (outputDevice != nullptr)  ResetEvent (outputDevice->clientEvent);
 
@@ -955,7 +1118,7 @@ public:
         {
             latencyIn = (int) (inputDevice->latencySamples + currentBufferSizeSamples);
 
-            if (! check (inputDevice->client->Start()))
+            if (! inputDevice->start (currentBufferSizeSamples))
             {
                 close();
                 lastError = TRANS("Couldn't start the input device!");
@@ -967,7 +1130,7 @@ public:
         {
             latencyOut = (int) (outputDevice->latencySamples + currentBufferSizeSamples);
 
-            if (! check (outputDevice->client->Start()))
+            if (! outputDevice->start())
             {
                 close();
                 lastError = TRANS("Couldn't start the output device!");
@@ -1056,34 +1219,48 @@ public:
         const int bufferSize        = currentBufferSizeSamples;
         const int numInputBuffers   = getActiveInputChannels().countNumberOfSetBits();
         const int numOutputBuffers  = getActiveOutputChannels().countNumberOfSetBits();
-        bool sampleRateChanged      = false;
+        bool sampleRateHasChanged = false;
 
         AudioSampleBuffer ins  (jmax (1, numInputBuffers),  bufferSize + 32);
         AudioSampleBuffer outs (jmax (1, numOutputBuffers), bufferSize + 32);
         float** const inputBuffers  = ins.getArrayOfWritePointers();
         float** const outputBuffers = outs.getArrayOfWritePointers();
         ins.clear();
+        outs.clear();
 
         while (! threadShouldExit())
         {
             if (inputDevice != nullptr)
             {
-                inputDevice->copyBuffers (inputBuffers, numInputBuffers, bufferSize, *this);
+                if (outputDevice == nullptr)
+                {
+                    if (WaitForSingleObject (inputDevice->clientEvent, 1000) == WAIT_TIMEOUT)
+                        break;
 
-                if (threadShouldExit())
-                    break;
+                    inputDevice->handleDeviceBuffer();
+
+                    if (inputDevice->getNumSamplesInReservoir() < bufferSize)
+                        continue;
+                }
+                else
+                {
+                    if (useExclusiveMode && WaitForSingleObject (inputDevice->clientEvent, 0) == WAIT_OBJECT_0)
+                        inputDevice->handleDeviceBuffer();
+                }
+
+                inputDevice->copyBuffersFromReservoir (inputBuffers, numInputBuffers, bufferSize);
 
                 if (inputDevice->sampleRateHasChanged)
                 {
-                    sampleRateChanged = true;
+                    sampleRateHasChanged = true;
                     sampleRateChangedByOutput = false;
                 }
             }
 
             {
-                const ScopedLock sl (startStopLock);
+                const ScopedTryLock sl (startStopLock);
 
-                if (isStarted)
+                if (sl.isLocked() && isStarted)
                     callback->audioDeviceIOCallback (const_cast<const float**> (inputBuffers), numInputBuffers,
                                                      outputBuffers, numOutputBuffers, bufferSize);
                 else
@@ -1092,16 +1269,18 @@ public:
 
             if (outputDevice != nullptr)
             {
-                outputDevice->copyBuffers (const_cast<const float**> (outputBuffers), numOutputBuffers, bufferSize, *this);
+                // Note that this function is handed the input device so it can check for the event and make sure
+                // the input reservoir is filled up correctly even when bufferSize > device actualBufferSize
+                outputDevice->copyBuffers (const_cast<const float**> (outputBuffers), numOutputBuffers, bufferSize, inputDevice, *this);
 
                 if (outputDevice->sampleRateHasChanged)
                 {
-                    sampleRateChanged = true;
+                    sampleRateHasChanged = true;
                     sampleRateChangedByOutput = true;
                 }
             }
 
-            if (sampleRateChanged)
+            if (sampleRateHasChanged)
             {
                 triggerAsyncUpdate();
                 break; // Quit the thread... will restart it later!
@@ -1208,9 +1387,10 @@ class WASAPIAudioIODeviceType  : public AudioIODeviceType,
                                  private DeviceChangeDetector
 {
 public:
-    WASAPIAudioIODeviceType()
-        : AudioIODeviceType ("Windows Audio"),
+    WASAPIAudioIODeviceType (bool exclusive)
+        : AudioIODeviceType (exclusive ? "Windows Audio (Exclusive Mode)" : "Windows Audio"),
           DeviceChangeDetector (L"Windows Audio"),
+          exclusiveMode (exclusive),
           hasScanned (false)
     {
     }
@@ -1267,7 +1447,6 @@ public:
     {
         jassert (hasScanned); // need to call scanForDevices() before doing this
 
-        const bool useExclusiveMode = false;
         ScopedPointer<WASAPIAudioIODevice> device;
 
         const int outputIndex = outputDeviceNames.indexOf (outputDeviceName);
@@ -1277,9 +1456,10 @@ public:
         {
             device = new WASAPIAudioIODevice (outputDeviceName.isNotEmpty() ? outputDeviceName
                                                                             : inputDeviceName,
+                                              getTypeName(),
                                               outputDeviceIds [outputIndex],
                                               inputDeviceIds [inputIndex],
-                                              useExclusiveMode);
+                                              exclusiveMode);
 
             if (! device->initialise())
                 device = nullptr;
@@ -1293,7 +1473,7 @@ public:
     StringArray inputDeviceNames, inputDeviceIds;
 
 private:
-    bool hasScanned;
+    bool exclusiveMode, hasScanned;
     ComSmartPtr<IMMDeviceEnumerator> enumerator;
 
     //==============================================================================
@@ -1418,7 +1598,7 @@ private:
     }
 
     //==============================================================================
-    void systemDeviceChanged()
+    void systemDeviceChanged() override
     {
         StringArray newOutNames, newInNames, newOutIds, newInIds;
         scan (newOutNames, newInNames, newOutIds, newInIds);
@@ -1493,12 +1673,16 @@ struct MMDeviceMasterVolume
 }
 
 //==============================================================================
-AudioIODeviceType* AudioIODeviceType::createAudioIODeviceType_WASAPI()
+AudioIODeviceType* AudioIODeviceType::createAudioIODeviceType_WASAPI (bool exclusiveMode)
 {
-    if (SystemStats::getOperatingSystemType() >= SystemStats::WinVista)
-        return new WasapiClasses::WASAPIAudioIODeviceType();
+   #if ! JUCE_WASAPI_EXCLUSIVE
+    if (exclusiveMode)
+        return nullptr;
+   #endif
 
-    return nullptr;
+    return SystemStats::getOperatingSystemType() >= SystemStats::WinVista
+                ? new WasapiClasses::WASAPIAudioIODeviceType (exclusiveMode)
+                : nullptr;
 }
 
 //==============================================================================

source/modules/juce_audio_formats/codecs/flac/all.h

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2000-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

source/modules/juce_audio_formats/codecs/flac/alloc.h

@@ -1,6 +1,6 @@
 /* alloc - Convenience routines for safely allocating memory
  * Copyright (C) 2007-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -33,7 +33,7 @@
 #ifndef FLAC__SHARE__ALLOC_H
 #define FLAC__SHARE__ALLOC_H
 
-#if HAVE_CONFIG_H
+#ifdef HAVE_CONFIG_H
 #  include <config.h>
 #endif
 

source/modules/juce_audio_formats/codecs/flac/assert.h

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2001-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

source/modules/juce_audio_formats/codecs/flac/callback.h

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2004-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

source/modules/juce_audio_formats/codecs/flac/compat.h

@@ -1,5 +1,5 @@
 /* libFLAC - Free Lossless Audio Codec library
- * Copyright (C) 2012  Xiph.org Foundation
+ * Copyright (C) 2012-2014  Xiph.org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -73,23 +73,25 @@
 #endif
 
 #if defined(_MSC_VER)
-#if _MSC_VER < 1500
-/* Visual Studio 2008 has restrict. */
-#define restrict __restrict
-#endif
 #define inline __inline
 #endif
 
-/* adjust for compilers that can't understand using LLU suffix for uint64_t literals */
-#ifdef _MSC_VER
-#define FLAC__U64L(x) x
+#if defined __INTEL_COMPILER || (defined _MSC_VER && defined _WIN64)
+/* MSVS generates VERY slow 32-bit code with __restrict */
+#define flac_restrict __restrict
+#elif defined __GNUC__
+#define flac_restrict __restrict__
 #else
-#define FLAC__U64L(x) x##LLU
+#define flac_restrict
 #endif
 
+#define FLAC__U64L(x) x##ULL
+
 #if defined _MSC_VER || defined __BORLANDC__ || defined __MINGW32__
+#define FLAC__STRCASECMP stricmp
 #define FLAC__STRNCASECMP strnicmp
 #else
+#define FLAC__STRCASECMP strcasecmp
 #define FLAC__STRNCASECMP strncasecmp
 #endif
 
@@ -139,6 +141,7 @@
 
 #ifdef _WIN32
 /* All char* strings are in UTF-8 format. Added to support Unicode files on Windows */
+#include "win_utf8_io.h"
 
 #define flac_printf printf_utf8
 #define flac_fprintf fprintf_utf8
@@ -160,12 +163,7 @@
 #define flac_utime utime
 #define flac_unlink unlink
 #define flac_rename rename
-
-#ifdef _WIN32
-#define flac_stat _stat64
-#else
 #define flac_stat stat
-#endif
 
 #endif
 
@@ -177,8 +175,14 @@
 #define flac_fstat fstat
 #endif
 
+#ifndef M_LN2
+#define M_LN2 0.69314718055994530942
+#endif
+#ifndef M_PI
+#define M_PI 3.14159265358979323846
+#endif
 
-/* FLAC needs to compile and work correctly on systems with a norrmal ISO C99
+/* FLAC needs to compile and work correctly on systems with a normal ISO C99
  * snprintf as well as Microsoft Visual Studio which has an non-standards
  * conformant snprint_s function.
  *
@@ -188,6 +192,7 @@
 extern "C" {
 #endif
 int flac_snprintf(char *str, size_t size, const char *fmt, ...);
+int flac_vsnprintf(char *str, size_t size, const char *fmt, va_list va);
 #ifdef __cplusplus
 };
 #endif

source/modules/juce_audio_formats/codecs/flac/endswap.h

@@ -1,5 +1,5 @@
 /* libFLAC - Free Lossless Audio Codec library
- * Copyright (C) 2012  Xiph.org Foundation
+ * Copyright (C) 2012-2014  Xiph.org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -33,20 +33,46 @@
 
 #if HAVE_BSWAP32			/* GCC and Clang */
 
+/* GCC prior to 4.8 didn't provide bswap16 on x86_64 */
+#if ! HAVE_BSWAP16
+static inline unsigned short __builtin_bswap16(unsigned short a)
+{
+	return (a<<8)|(a>>8);
+}
+#endif
+
+#define	ENDSWAP_16(x)		(__builtin_bswap16 (x))
 #define	ENDSWAP_32(x)		(__builtin_bswap32 (x))
 
 #elif defined _MSC_VER		/* Windows. Apparently in <stdlib.h>. */
 
+#define	ENDSWAP_16(x)		(_byteswap_ushort (x))
 #define	ENDSWAP_32(x)		(_byteswap_ulong (x))
 
 #elif defined HAVE_BYTESWAP_H		/* Linux */
 
 #include <byteswap.h>
 
+#define	ENDSWAP_16(x)		(bswap_16 (x))
 #define	ENDSWAP_32(x)		(bswap_32 (x))
 
 #else
 
-#define	ENDSWAP_32(x)		((((x) >> 24) & 0xFF) + (((x) >> 8) & 0xFF00) + (((x) & 0xFF00) << 8) + (((x) & 0xFF) << 24))
+#define	ENDSWAP_16(x)		((((x) >> 8) & 0xFF) | (((x) & 0xFF) << 8))
+#define	ENDSWAP_32(x)		((((x) >> 24) & 0xFF) | (((x) >> 8) & 0xFF00) | (((x) & 0xFF00) << 8) | (((x) & 0xFF) << 24))
+
+#endif
+
+
+/* Host to little-endian byte swapping. */
+#if CPU_IS_BIG_ENDIAN
+
+#define H2LE_16(x)		ENDSWAP_16 (x)
+#define H2LE_32(x)		ENDSWAP_32 (x)
+
+#else
+
+#define H2LE_16(x)		(x)
+#define H2LE_32(x)		(x)
 
 #endif

source/modules/juce_audio_formats/codecs/flac/export.h

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2000-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -61,9 +61,9 @@
 
 #elif defined(_MSC_VER)
 #ifdef FLAC_API_EXPORTS
-#define	FLAC_API	_declspec(dllexport)
+#define	FLAC_API __declspec(dllexport)
 #else
-#define FLAC_API	_declspec(dllimport)
+#define FLAC_API __declspec(dllimport)
 #endif
 
 #elif defined(FLAC__USE_VISIBILITY_ATTR)

source/modules/juce_audio_formats/codecs/flac/format.h

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2000-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -509,9 +509,11 @@ typedef enum {
 	FLAC__METADATA_TYPE_PICTURE = 6,
 	/**< <A HREF="../format.html#metadata_block_picture">PICTURE</A> block */
 
-	FLAC__METADATA_TYPE_UNDEFINED = 7
+	FLAC__METADATA_TYPE_UNDEFINED = 7,
 	/**< marker to denote beginning of undefined type range; this number will increase as new metadata types are added */
 
+    FLAC__MAX_METADATA_TYPE = FLAC__MAX_METADATA_TYPE_CODE,
+    /**< No type will ever be greater than this. There is not enough room in the protocol block. */
 } FLAC__MetadataType;
 
 /** Maps a FLAC__MetadataType to a C string.

source/modules/juce_audio_formats/codecs/flac/libFLAC/bitmath.c

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2001-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -30,12 +30,11 @@
  * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
-#if HAVE_CONFIG_H
+#ifdef HAVE_CONFIG_H
 #  include <config.h>
 #endif
 
 #include "include/private/bitmath.h"
-#include "../assert.h"
 
 /* An example of what FLAC__bitmath_silog2() computes:
  *

source/modules/juce_audio_formats/codecs/flac/libFLAC/bitreader.c

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2000-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -30,7 +30,7 @@
  * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
-#if HAVE_CONFIG_H
+#ifdef HAVE_CONFIG_H
 #  include <config.h>
 #endif
 
@@ -73,7 +73,6 @@
  */
 static const unsigned FLAC__BITREADER_DEFAULT_CAPACITY = 65536u / FLAC__BITS_PER_WORD; /* in words */
 
-/* WATCHOUT: assembly routines rely on the order in which these fields are declared */
 struct FLAC__BitReader {
 	/* any partially-consumed word at the head will stay right-justified as bits are consumed from the left */
 	/* any incomplete word at the tail will be left-justified, and bytes from the read callback are added on the right */
@@ -87,7 +86,6 @@ struct FLAC__BitReader {
 	unsigned crc16_align; /* the number of bits in the current consumed word that should not be CRC'd */
 	FLAC__BitReaderReadCallback read_callback;
 	void *client_data;
-	FLAC__CPUInfo cpu_info;
 };
 
 static inline void crc16_update_word_(FLAC__BitReader *br, uint32_t word)
@@ -119,8 +117,7 @@ static inline void crc16_update_word_(FLAC__BitReader *br, uint32_t word)
 	br->crc16_align = 0;
 }
 
-/* would be static except it needs to be called by asm routines */
-FLAC__bool bitreader_read_from_client_(FLAC__BitReader *br)
+static FLAC__bool bitreader_read_from_client_(FLAC__BitReader *br)
 {
 	unsigned start, end;
 	size_t bytes;
@@ -231,7 +228,7 @@ void FLAC__bitreader_delete(FLAC__BitReader *br)
  *
  ***********************************************************************/
 
-FLAC__bool FLAC__bitreader_init(FLAC__BitReader *br, FLAC__CPUInfo cpu, FLAC__BitReaderReadCallback rcb, void *cd)
+FLAC__bool FLAC__bitreader_init(FLAC__BitReader *br, FLAC__BitReaderReadCallback rcb, void *cd)
 {
 	FLAC__ASSERT(0 != br);
 
@@ -243,7 +240,6 @@ FLAC__bool FLAC__bitreader_init(FLAC__BitReader *br, FLAC__CPUInfo cpu, FLAC__Bi
 		return false;
 	br->read_callback = rcb;
 	br->client_data = cd;
-	br->cpu_info = cpu;
 
 	return true;
 }
@@ -1048,9 +1044,9 @@ FLAC__bool FLAC__bitreader_read_utf8_uint64(FLAC__BitReader *br, FLAC__uint64 *v
 	return true;
 }
 
-/* These functions a declared inline in this file but are also callable as
+/* These functions are declared inline in this file but are also callable as
  * externs from elsewhere.
- * According to the C99 sepc, section 6.7.4, simply providing a function
+ * According to the C99 spec, section 6.7.4, simply providing a function
  * prototype in a header file without 'inline' and making the function inline
  * in this file should be sufficient.
  * Unfortunately, the Microsoft VS compiler doesn't pick them up externally. To

source/modules/juce_audio_formats/codecs/flac/libFLAC/bitwriter.c

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2000-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -30,7 +30,7 @@
  * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
-#if HAVE_CONFIG_H
+#ifdef HAVE_CONFIG_H
 #  include <config.h>
 #endif
 
@@ -47,8 +47,7 @@
 /* WATCHOUT: if you change this you must also change the following #defines down to SWAP_BE_WORD_TO_HOST below to match */
 /* WATCHOUT: there are a few places where the code will not work unless uint32_t is >= 32 bits wide */
 #define FLAC__BYTES_PER_WORD 4
-#undef FLAC__BITS_PER_WORD
-#define FLAC__BITS_PER_WORD 32
+#define FLAC__BITS_PER_WORD (8 * FLAC__BYTES_PER_WORD)
 #define FLAC__WORD_ALL_ONES ((FLAC__uint32)0xffffffff)
 /* SWAP_BE_WORD_TO_HOST swaps bytes in a uint32_t (which is always big-endian) if necessary to match host byte order */
 #if WORDS_BIGENDIAN
@@ -524,28 +523,6 @@ FLAC__bool FLAC__bitwriter_write_rice_signed_block(FLAC__BitWriter *bw, const FL
 
 		msbits = uval >> parameter;
 
-#if 0 /* OPT: can remove this special case if it doesn't make up for the extra compare (doesn't make a statistically significant difference with msvc or gcc/x86) */
-		if(bw->bits && bw->bits + msbits + lsbits <= FLAC__BITS_PER_WORD) { /* i.e. if the whole thing fits in the current uint32_t */
-			/* ^^^ if bw->bits is 0 then we may have filled the buffer and have no free uint32_t to work in */
-			bw->bits = bw->bits + msbits + lsbits;
-			uval |= mask1; /* set stop bit */
-			uval &= mask2; /* mask off unused top bits */
-			/* NOT: bw->accum <<= msbits + lsbits because msbits+lsbits could be 32, then the shift would be a NOP */
-			bw->accum <<= msbits;
-			bw->accum <<= lsbits;
-			bw->accum |= uval;
-			if(bw->bits == FLAC__BITS_PER_WORD) {
-				bw->buffer[bw->words++] = SWAP_BE_WORD_TO_HOST(bw->accum);
-				bw->bits = 0;
-				/* burying the capacity check down here means we have to grow the buffer a little if there are more vals to do */
-				if(bw->capacity <= bw->words && nvals > 1 && !bitwriter_grow_(bw, 1)) {
-					FLAC__ASSERT(bw->capacity == bw->words);
-					return false;
-				}
-			}
-		}
-		else {
-#elif 1 /*@@@@@@ OPT: try this version with MSVC6 to see if better, not much difference for gcc-4 */
 		if(bw->bits && bw->bits + msbits + lsbits < FLAC__BITS_PER_WORD) { /* i.e. if the whole thing fits in the current uint32_t */
 			/* ^^^ if bw->bits is 0 then we may have filled the buffer and have no free uint32_t to work in */
 			bw->bits = bw->bits + msbits + lsbits;
@@ -555,7 +532,6 @@ FLAC__bool FLAC__bitwriter_write_rice_signed_block(FLAC__BitWriter *bw, const FL
 			bw->accum |= uval;
 		}
 		else {
-#endif
 			/* slightly pessimistic size check but faster than "<= bw->words + (bw->bits+msbits+lsbits+FLAC__BITS_PER_WORD-1)/FLAC__BITS_PER_WORD" */
 			/* OPT: pessimism may cause flurry of false calls to grow_ which eat up all savings before it */
 			if(bw->capacity <= bw->words + bw->bits + msbits + 1/*lsbits always fit in 1 uint32_t*/ && !bitwriter_grow_(bw, msbits+lsbits))
@@ -610,9 +586,7 @@ break1:
 				bw->buffer[bw->words++] = SWAP_BE_WORD_TO_HOST(bw->accum);
 				bw->accum = uval;
 			}
-#if 1
 		}
-#endif
 		vals++;
 		nvals--;
 	}
@@ -853,9 +827,9 @@ FLAC__bool FLAC__bitwriter_zero_pad_to_byte_boundary(FLAC__BitWriter *bw)
 		return true;
 }
 
-/* These functions a declared inline in this file but are also callable as
+/* These functions are declared inline in this file but are also callable as
  * externs from elsewhere.
- * According to the C99 sepc, section 6.7.4, simply providing a function
+ * According to the C99 spec, section 6.7.4, simply providing a function
  * prototype in a header file without 'inline' and making the function inline
  * in this file should be sufficient.
  * Unfortunately, the Microsoft VS compiler doesn't pick them up externally. To

source/modules/juce_audio_formats/codecs/flac/libFLAC/cpu.c

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2001-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -30,47 +30,47 @@
  * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
-#if HAVE_CONFIG_H
+#ifdef HAVE_CONFIG_H
 #  include <config.h>
 #endif
 
 #include "include/private/cpu.h"
-#include <stdlib.h>
-#include <stdio.h>
+
+#if 0
+ #include <stdlib.h>
+ #include <memory.h>
+ #include <stdio.h>
+#endif
 
 #if defined FLAC__CPU_IA32
 # include <signal.h>
-#elif defined FLAC__CPU_PPC
-# if !defined FLAC__NO_ASM
-#  if defined FLAC__SYS_DARWIN
-#   include <sys/sysctl.h>
-#   include <mach/mach.h>
-#   include <mach/mach_host.h>
-#   include <mach/host_info.h>
-#   include <mach/machine.h>
-#   ifndef CPU_SUBTYPE_POWERPC_970
-#    define CPU_SUBTYPE_POWERPC_970 ((cpu_subtype_t) 100)
-#   endif
-#  else /* FLAC__SYS_DARWIN */
-
-#   include <signal.h>
-#   include <setjmp.h>
-
-static sigjmp_buf jmpbuf;
-static volatile sig_atomic_t canjump = 0;
-
-static void sigill_handler (int sig)
+
+static void disable_sse(FLAC__CPUInfo *info)
 {
-	if (!canjump) {
-		signal (sig, SIG_DFL);
-		raise (sig);
-	}
-	canjump = 0;
-	siglongjmp (jmpbuf, 1);
+	info->ia32.sse   = false;
+	info->ia32.sse2  = false;
+	info->ia32.sse3  = false;
+	info->ia32.ssse3 = false;
+	info->ia32.sse41 = false;
+	info->ia32.sse42 = false;
+}
+
+static void disable_avx(FLAC__CPUInfo *info)
+{
+	info->ia32.avx     = false;
+	info->ia32.avx2    = false;
+	info->ia32.fma     = false;
 }
-#  endif /* FLAC__SYS_DARWIN */
-# endif /* FLAC__NO_ASM */
-#endif /* FLAC__CPU_PPC */
+
+#elif defined FLAC__CPU_X86_64
+
+static void disable_avx(FLAC__CPUInfo *info)
+{
+	info->x86.avx     = false;
+	info->x86.avx2    = false;
+	info->x86.fma     = false;
+}
+#endif
 
 #if defined (__NetBSD__) || defined(__OpenBSD__)
 #include <sys/param.h>
@@ -87,25 +87,34 @@ static void sigill_handler (int sig)
 /* how to get sysctlbyname()? */
 #endif
 
+#ifdef FLAC__CPU_IA32
 /* these are flags in EDX of CPUID AX=00000001 */
 static const unsigned FLAC__CPUINFO_IA32_CPUID_CMOV = 0x00008000;
 static const unsigned FLAC__CPUINFO_IA32_CPUID_MMX = 0x00800000;
 static const unsigned FLAC__CPUINFO_IA32_CPUID_FXSR = 0x01000000;
 static const unsigned FLAC__CPUINFO_IA32_CPUID_SSE = 0x02000000;
 static const unsigned FLAC__CPUINFO_IA32_CPUID_SSE2 = 0x04000000;
+#endif
+
 /* these are flags in ECX of CPUID AX=00000001 */
 static const unsigned FLAC__CPUINFO_IA32_CPUID_SSE3 = 0x00000001;
 static const unsigned FLAC__CPUINFO_IA32_CPUID_SSSE3 = 0x00000200;
-/* these are flags in EDX of CPUID AX=80000001 */
-static const unsigned FLAC__CPUINFO_IA32_CPUID_EXTENDED_AMD_3DNOW = 0x80000000;
-static const unsigned FLAC__CPUINFO_IA32_CPUID_EXTENDED_AMD_EXT3DNOW = 0x40000000;
-static const unsigned FLAC__CPUINFO_IA32_CPUID_EXTENDED_AMD_EXTMMX = 0x00400000;
+static const unsigned FLAC__CPUINFO_IA32_CPUID_SSE41 = 0x00080000;
+static const unsigned FLAC__CPUINFO_IA32_CPUID_SSE42 = 0x00100000;
 
+#if defined FLAC__AVX_SUPPORTED
+/* these are flags in ECX of CPUID AX=00000001 */
+static const unsigned FLAC__CPUINFO_IA32_CPUID_OSXSAVE = 0x08000000;
+static const unsigned FLAC__CPUINFO_IA32_CPUID_AVX = 0x10000000;
+static const unsigned FLAC__CPUINFO_IA32_CPUID_FMA = 0x00001000;
+/* these are flags in EBX of CPUID AX=00000007 */
+static const unsigned FLAC__CPUINFO_IA32_CPUID_AVX2 = 0x00000020;
+#endif
 
 /*
  * Extra stuff needed for detection of OS support for SSE on IA-32
  */
-#if defined(FLAC__CPU_IA32) && !defined FLAC__NO_ASM && defined FLAC__HAS_NASM && !defined FLAC__NO_SSE_OS && !defined FLAC__SSE_OS
+#if defined(FLAC__CPU_IA32) && !defined FLAC__NO_ASM && (defined FLAC__HAS_NASM || defined FLAC__HAS_X86INTRIN) && !defined FLAC__NO_SSE_OS && !defined FLAC__SSE_OS
 # if defined(__linux__)
 /*
  * If the OS doesn't support SSE, we will get here with a SIGILL.  We
@@ -120,35 +129,14 @@ static const unsigned FLAC__CPUINFO_IA32_CPUID_EXTENDED_AMD_EXTMMX = 0x00400000;
  *   6 bytes extra in case our estimate is wrong
  * 12 bytes puts us in the NOP "landing zone"
  */
-#  undef USE_OBSOLETE_SIGCONTEXT_FLAVOR /* #define this to use the older signal handler method */
-#  ifdef USE_OBSOLETE_SIGCONTEXT_FLAVOR
-	static void sigill_handler_sse_os(int signal, struct sigcontext sc)
-	{
-		(void)signal;
-		sc.eip += 3 + 3 + 6;
-	}
-#  else
 #   include <sys/ucontext.h>
 	static void sigill_handler_sse_os(int signal, siginfo_t *si, void *uc)
 	{
 		(void)signal, (void)si;
 		((ucontext_t*)uc)->uc_mcontext.gregs[14/*REG_EIP*/] += 3 + 3 + 6;
 	}
-#  endif
 # elif defined(_MSC_VER)
 #  include <windows.h>
-#  define USE_TRY_CATCH_FLAVOR /* sigill_handler flavor resulted in several crash reports on win32 */
-#  ifdef USE_TRY_CATCH_FLAVOR
-#  else
-	LONG CALLBACK sigill_handler_sse_os(EXCEPTION_POINTERS *ep)
-	{
-		if(ep->ExceptionRecord->ExceptionCode == EXCEPTION_ILLEGAL_INSTRUCTION) {
-			ep->ContextRecord->Eip += 3 + 3 + 6;
-			return EXCEPTION_CONTINUE_EXECUTION;
-		}
-		return EXCEPTION_CONTINUE_SEARCH;
-	}
-#  endif
 # endif
 #endif
 
@@ -159,261 +147,344 @@ void FLAC__cpu_info(FLAC__CPUInfo *info)
  * IA32-specific
  */
 #ifdef FLAC__CPU_IA32
+	FLAC__bool ia32_fxsr = false;
+	FLAC__bool ia32_osxsave = false;
+	(void) ia32_fxsr; (void) ia32_osxsave; /* to avoid warnings about unused variables */
+	memset(info, 0, sizeof(*info));
 	info->type = FLAC__CPUINFO_TYPE_IA32;
-#if !defined FLAC__NO_ASM && defined FLAC__HAS_NASM
+#if !defined FLAC__NO_ASM && (defined FLAC__HAS_NASM || defined FLAC__HAS_X86INTRIN)
 	info->use_asm = true; /* we assume a minimum of 80386 with FLAC__CPU_IA32 */
-	info->data.ia32.cpuid = FLAC__cpu_have_cpuid_asm_ia32()? true : false;
-	info->data.ia32.bswap = info->data.ia32.cpuid; /* CPUID => BSWAP since it came after */
-	info->data.ia32.cmov = false;
-	info->data.ia32.mmx = false;
-	info->data.ia32.fxsr = false;
-	info->data.ia32.sse = false;
-	info->data.ia32.sse2 = false;
-	info->data.ia32.sse3 = false;
-	info->data.ia32.ssse3 = false;
-	info->data.ia32._3dnow = false;
-	info->data.ia32.ext3dnow = false;
-	info->data.ia32.extmmx = false;
-	if(info->data.ia32.cpuid) {
-		/* http://www.sandpile.org/ia32/cpuid.htm */
-		FLAC__uint32 flags_edx, flags_ecx;
-		FLAC__cpu_info_asm_ia32(&flags_edx, &flags_ecx);
-		info->data.ia32.cmov  = (flags_edx & FLAC__CPUINFO_IA32_CPUID_CMOV )? true : false;
-		info->data.ia32.mmx   = (flags_edx & FLAC__CPUINFO_IA32_CPUID_MMX  )? true : false;
-		info->data.ia32.fxsr  = (flags_edx & FLAC__CPUINFO_IA32_CPUID_FXSR )? true : false;
-		info->data.ia32.sse   = (flags_edx & FLAC__CPUINFO_IA32_CPUID_SSE  )? true : false;
-		info->data.ia32.sse2  = (flags_edx & FLAC__CPUINFO_IA32_CPUID_SSE2 )? true : false;
-		info->data.ia32.sse3  = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_SSE3 )? true : false;
-		info->data.ia32.ssse3 = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_SSSE3)? true : false;
-
-#ifdef FLAC__USE_3DNOW
-		flags_edx = FLAC__cpu_info_extended_amd_asm_ia32();
-		info->data.ia32._3dnow   = (flags_edx & FLAC__CPUINFO_IA32_CPUID_EXTENDED_AMD_3DNOW   )? true : false;
-		info->data.ia32.ext3dnow = (flags_edx & FLAC__CPUINFO_IA32_CPUID_EXTENDED_AMD_EXT3DNOW)? true : false;
-		info->data.ia32.extmmx   = (flags_edx & FLAC__CPUINFO_IA32_CPUID_EXTENDED_AMD_EXTMMX  )? true : false;
+#ifdef FLAC__HAS_X86INTRIN
+	if(!FLAC__cpu_have_cpuid_x86())
+		return;
+#else
+	if(!FLAC__cpu_have_cpuid_asm_ia32())
+		return;
+#endif
+	{
+		/* http://www.sandpile.org/x86/cpuid.htm */
+#ifdef FLAC__HAS_X86INTRIN
+		FLAC__uint32 flags_eax, flags_ebx, flags_ecx, flags_edx;
+		FLAC__cpu_info_x86(1, &flags_eax, &flags_ebx, &flags_ecx, &flags_edx);
 #else
-		info->data.ia32._3dnow = info->data.ia32.ext3dnow = info->data.ia32.extmmx = false;
+		FLAC__uint32 flags_ecx, flags_edx;
+		FLAC__cpu_info_asm_ia32(&flags_edx, &flags_ecx);
 #endif
+		info->ia32.cmov  = (flags_edx & FLAC__CPUINFO_IA32_CPUID_CMOV )? true : false;
+		info->ia32.mmx   = (flags_edx & FLAC__CPUINFO_IA32_CPUID_MMX  )? true : false;
+		      ia32_fxsr  = (flags_edx & FLAC__CPUINFO_IA32_CPUID_FXSR )? true : false;
+		info->ia32.sse   = (flags_edx & FLAC__CPUINFO_IA32_CPUID_SSE  )? true : false;
+		info->ia32.sse2  = (flags_edx & FLAC__CPUINFO_IA32_CPUID_SSE2 )? true : false;
+		info->ia32.sse3  = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_SSE3 )? true : false;
+		info->ia32.ssse3 = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_SSSE3)? true : false;
+		info->ia32.sse41 = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_SSE41)? true : false;
+		info->ia32.sse42 = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_SSE42)? true : false;
+#if defined FLAC__HAS_X86INTRIN && defined FLAC__AVX_SUPPORTED
+		    ia32_osxsave = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_OSXSAVE)? true : false;
+		info->ia32.avx   = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_AVX    )? true : false;
+		info->ia32.fma   = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_FMA    )? true : false;
+		FLAC__cpu_info_x86(7, &flags_eax, &flags_ebx, &flags_ecx, &flags_edx);
+		info->ia32.avx2  = (flags_ebx & FLAC__CPUINFO_IA32_CPUID_AVX2   )? true : false;
+#endif
+	}
 
 #ifdef DEBUG
-		fprintf(stderr, "CPU info (IA-32):\n");
-		fprintf(stderr, "  CPUID ...... %c\n", info->data.ia32.cpuid   ? 'Y' : 'n');
-		fprintf(stderr, "  BSWAP ...... %c\n", info->data.ia32.bswap   ? 'Y' : 'n');
-		fprintf(stderr, "  CMOV ....... %c\n", info->data.ia32.cmov    ? 'Y' : 'n');
-		fprintf(stderr, "  MMX ........ %c\n", info->data.ia32.mmx     ? 'Y' : 'n');
-		fprintf(stderr, "  FXSR ....... %c\n", info->data.ia32.fxsr    ? 'Y' : 'n');
-		fprintf(stderr, "  SSE ........ %c\n", info->data.ia32.sse     ? 'Y' : 'n');
-		fprintf(stderr, "  SSE2 ....... %c\n", info->data.ia32.sse2    ? 'Y' : 'n');
-		fprintf(stderr, "  SSE3 ....... %c\n", info->data.ia32.sse3    ? 'Y' : 'n');
-		fprintf(stderr, "  SSSE3 ...... %c\n", info->data.ia32.ssse3   ? 'Y' : 'n');
-		fprintf(stderr, "  3DNow! ..... %c\n", info->data.ia32._3dnow  ? 'Y' : 'n');
-		fprintf(stderr, "  3DNow!-ext . %c\n", info->data.ia32.ext3dnow? 'Y' : 'n');
-		fprintf(stderr, "  3DNow!-MMX . %c\n", info->data.ia32.extmmx  ? 'Y' : 'n');
+	fprintf(stderr, "CPU info (IA-32):\n");
+	fprintf(stderr, "  CMOV ....... %c\n", info->ia32.cmov    ? 'Y' : 'n');
+	fprintf(stderr, "  MMX ........ %c\n", info->ia32.mmx     ? 'Y' : 'n');
+	fprintf(stderr, "  SSE ........ %c\n", info->ia32.sse     ? 'Y' : 'n');
+	fprintf(stderr, "  SSE2 ....... %c\n", info->ia32.sse2    ? 'Y' : 'n');
+	fprintf(stderr, "  SSE3 ....... %c\n", info->ia32.sse3    ? 'Y' : 'n');
+	fprintf(stderr, "  SSSE3 ...... %c\n", info->ia32.ssse3   ? 'Y' : 'n');
+	fprintf(stderr, "  SSE41 ...... %c\n", info->ia32.sse41   ? 'Y' : 'n');
+	fprintf(stderr, "  SSE42 ...... %c\n", info->ia32.sse42   ? 'Y' : 'n');
+# if defined FLAC__HAS_X86INTRIN && defined FLAC__AVX_SUPPORTED
+	fprintf(stderr, "  AVX ........ %c\n", info->ia32.avx     ? 'Y' : 'n');
+	fprintf(stderr, "  FMA ........ %c\n", info->ia32.fma     ? 'Y' : 'n');
+	fprintf(stderr, "  AVX2 ....... %c\n", info->ia32.avx2    ? 'Y' : 'n');
+# endif
 #endif
 
-		/*
-		 * now have to check for OS support of SSE/SSE2
-		 */
-		if(info->data.ia32.fxsr || info->data.ia32.sse || info->data.ia32.sse2) {
+	/*
+	 * now have to check for OS support of SSE instructions
+	 */
+	if(info->ia32.sse) {
 #if defined FLAC__NO_SSE_OS
-			/* assume user knows better than us; turn it off */
-			info->data.ia32.fxsr = info->data.ia32.sse = info->data.ia32.sse2 = info->data.ia32.sse3 = info->data.ia32.ssse3 = false;
+		/* assume user knows better than us; turn it off */
+		disable_sse(info);
 #elif defined FLAC__SSE_OS
-			/* assume user knows better than us; leave as detected above */
+		/* assume user knows better than us; leave as detected above */
 #elif defined(__FreeBSD__) || defined(__FreeBSD_kernel__) || defined(__DragonFly__) || defined(__APPLE__)
-			int sse = 0;
-			size_t len;
-			/* at least one of these must work: */
-			len = sizeof(sse); sse = sse || (sysctlbyname("hw.instruction_sse", &sse, &len, NULL, 0) == 0 && sse);
-			len = sizeof(sse); sse = sse || (sysctlbyname("hw.optional.sse"   , &sse, &len, NULL, 0) == 0 && sse); /* __APPLE__ ? */
-			if(!sse)
-				info->data.ia32.fxsr = info->data.ia32.sse = info->data.ia32.sse2 = info->data.ia32.sse3 = info->data.ia32.ssse3 = false;
+		int sse = 0;
+		size_t len;
+		/* at least one of these must work: */
+		len = sizeof(sse); sse = sse || (sysctlbyname("hw.instruction_sse", &sse, &len, NULL, 0) == 0 && sse);
+		len = sizeof(sse); sse = sse || (sysctlbyname("hw.optional.sse"   , &sse, &len, NULL, 0) == 0 && sse); /* __APPLE__ ? */
+		if(!sse)
+			disable_sse(info);
 #elif defined(__NetBSD__) || defined (__OpenBSD__)
 # if __NetBSD_Version__ >= 105250000 || (defined __OpenBSD__)
-			int val = 0, mib[2] = { CTL_MACHDEP, CPU_SSE };
-			size_t len = sizeof(val);
-			if(sysctl(mib, 2, &val, &len, NULL, 0) < 0 || !val)
-				info->data.ia32.fxsr = info->data.ia32.sse = info->data.ia32.sse2 = info->data.ia32.sse3 = info->data.ia32.ssse3 = false;
-			else { /* double-check SSE2 */
-				mib[1] = CPU_SSE2;
-				len = sizeof(val);
-				if(sysctl(mib, 2, &val, &len, NULL, 0) < 0 || !val)
-					info->data.ia32.sse2 = info->data.ia32.sse3 = info->data.ia32.ssse3 = false;
+		int val = 0, mib[2] = { CTL_MACHDEP, CPU_SSE };
+		size_t len = sizeof(val);
+		if(sysctl(mib, 2, &val, &len, NULL, 0) < 0 || !val)
+			disable_sse(info);
+		else { /* double-check SSE2 */
+			mib[1] = CPU_SSE2;
+			len = sizeof(val);
+			if(sysctl(mib, 2, &val, &len, NULL, 0) < 0 || !val) {
+				disable_sse(info);
+				info->ia32.sse = true;
 			}
+		}
 # else
-			info->data.ia32.fxsr = info->data.ia32.sse = info->data.ia32.sse2 = info->data.ia32.sse3 = info->data.ia32.ssse3 = false;
+		disable_sse(info);
 # endif
 #elif defined(__linux__)
-			int sse = 0;
-			struct sigaction sigill_save;
-#ifdef USE_OBSOLETE_SIGCONTEXT_FLAVOR
-			if(0 == sigaction(SIGILL, NULL, &sigill_save) && signal(SIGILL, (void (*)(int))sigill_handler_sse_os) != SIG_ERR)
-#else
-			struct sigaction sigill_sse;
-			sigill_sse.sa_sigaction = sigill_handler_sse_os;
-			__sigemptyset(&sigill_sse.sa_mask);
-			sigill_sse.sa_flags = SA_SIGINFO | SA_RESETHAND; /* SA_RESETHAND just in case our SIGILL return jump breaks, so we don't get stuck in a loop */
-			if(0 == sigaction(SIGILL, &sigill_sse, &sigill_save))
-#endif
-			{
-				/* http://www.ibiblio.org/gferg/ldp/GCC-Inline-Assembly-HOWTO.html */
-				/* see sigill_handler_sse_os() for an explanation of the following: */
-				asm volatile (
-					"xorl %0,%0\n\t"          /* for some reason, still need to do this to clear 'sse' var */
-					"xorps %%xmm0,%%xmm0\n\t" /* will cause SIGILL if unsupported by OS */
-					"incl %0\n\t"             /* SIGILL handler will jump over this */
-					/* landing zone */
-					"nop\n\t" /* SIGILL jump lands here if "inc" is 9 bytes */
-					"nop\n\t"
-					"nop\n\t"
-					"nop\n\t"
-					"nop\n\t"
-					"nop\n\t"
-					"nop\n\t" /* SIGILL jump lands here if "inc" is 3 bytes (expected) */
-					"nop\n\t"
-					"nop"     /* SIGILL jump lands here if "inc" is 1 byte */
-					: "=r"(sse)
-					: "r"(sse)
-				);
-
-				sigaction(SIGILL, &sigill_save, NULL);
-			}
+		int sse = 0;
+		struct sigaction sigill_save;
+		struct sigaction sigill_sse;
+		sigill_sse.sa_sigaction = sigill_handler_sse_os;
+		__sigemptyset(&sigill_sse.sa_mask);
+		sigill_sse.sa_flags = SA_SIGINFO | SA_RESETHAND; /* SA_RESETHAND just in case our SIGILL return jump breaks, so we don't get stuck in a loop */
+		if(0 == sigaction(SIGILL, &sigill_sse, &sigill_save))
+		{
+			/* http://www.ibiblio.org/gferg/ldp/GCC-Inline-Assembly-HOWTO.html */
+			/* see sigill_handler_sse_os() for an explanation of the following: */
+			asm volatile (
+				"xorps %%xmm0,%%xmm0\n\t" /* will cause SIGILL if unsupported by OS */
+				"incl %0\n\t"             /* SIGILL handler will jump over this */
+				/* landing zone */
+				"nop\n\t" /* SIGILL jump lands here if "inc" is 9 bytes */
+				"nop\n\t"
+				"nop\n\t"
+				"nop\n\t"
+				"nop\n\t"
+				"nop\n\t"
+				"nop\n\t" /* SIGILL jump lands here if "inc" is 3 bytes (expected) */
+				"nop\n\t"
+				"nop"     /* SIGILL jump lands here if "inc" is 1 byte */
+				: "=r"(sse)
+				: "0"(sse)
+			);
 
-			if(!sse)
-				info->data.ia32.fxsr = info->data.ia32.sse = info->data.ia32.sse2 = info->data.ia32.sse3 = info->data.ia32.ssse3 = false;
+			sigaction(SIGILL, &sigill_save, NULL);
+		}
+
+		if(!sse)
+			disable_sse(info);
 #elif defined(_MSC_VER)
-# ifdef USE_TRY_CATCH_FLAVOR
-			_try {
-				__asm {
-#  if _MSC_VER <= 1200
-					/* VC6 assembler doesn't know SSE, have to emit bytecode instead */
-					_emit 0x0F
-					_emit 0x57
-					_emit 0xC0
-#  else
-					xorps xmm0,xmm0
-#  endif
-				}
-			}
-			_except(EXCEPTION_EXECUTE_HANDLER) {
-				if (_exception_code() == STATUS_ILLEGAL_INSTRUCTION)
-					info->data.ia32.fxsr = info->data.ia32.sse = info->data.ia32.sse2 = info->data.ia32.sse3 = info->data.ia32.ssse3 = false;
-			}
-# else
-			int sse = 0;
-			LPTOP_LEVEL_EXCEPTION_FILTER save = SetUnhandledExceptionFilter(sigill_handler_sse_os);
-			/* see GCC version above for explanation */
-			/*  http://msdn2.microsoft.com/en-us/library/4ks26t93.aspx */
-			/*  http://www.codeproject.com/cpp/gccasm.asp */
-			/*  http://www.hick.org/~mmiller/msvc_inline_asm.html */
+		__try {
 			__asm {
-#  if _MSC_VER <= 1200
-				/* VC6 assembler doesn't know SSE, have to emit bytecode instead */
-				_emit 0x0F
-				_emit 0x57
-				_emit 0xC0
-#  else
 				xorps xmm0,xmm0
-#  endif
-				inc sse
-				nop
-				nop
-				nop
-				nop
-				nop
-				nop
-				nop
-				nop
-				nop
 			}
-			SetUnhandledExceptionFilter(save);
-			if(!sse)
-				info->data.ia32.fxsr = info->data.ia32.sse = info->data.ia32.sse2 = info->data.ia32.sse3 = info->data.ia32.ssse3 = false;
-# endif
+		}
+		__except(EXCEPTION_EXECUTE_HANDLER) {
+			if (_exception_code() == STATUS_ILLEGAL_INSTRUCTION)
+				disable_sse(info);
+		}
+#elif defined(__GNUC__) /* MinGW goes here */
+		int sse = 0;
+		/* Based on the idea described in Agner Fog's manual "Optimizing subroutines in assembly language" */
+		/* In theory, not guaranteed to detect lack of OS SSE support on some future Intel CPUs, but in practice works (see the aforementioned manual) */
+		if (ia32_fxsr) {
+			struct {
+				FLAC__uint32 buff[128];
+			} __attribute__((aligned(16))) fxsr;
+			FLAC__uint32 old_val, new_val;
+
+			asm volatile ("fxsave %0"  : "=m" (fxsr) : "m" (fxsr));
+			old_val = fxsr.buff[50];
+			fxsr.buff[50] ^= 0x0013c0de;                             /* change value in the buffer */
+			asm volatile ("fxrstor %0" : "=m" (fxsr) : "m" (fxsr));  /* try to change SSE register */
+			fxsr.buff[50] = old_val;                                 /* restore old value in the buffer */
+			asm volatile ("fxsave %0 " : "=m" (fxsr) : "m" (fxsr));  /* old value will be overwritten if SSE register was changed */
+			new_val = fxsr.buff[50];                                 /* == old_val if FXRSTOR didn't change SSE register and (old_val ^ 0x0013c0de) otherwise */
+			fxsr.buff[50] = old_val;                                 /* again restore old value in the buffer */
+			asm volatile ("fxrstor %0" : "=m" (fxsr) : "m" (fxsr));  /* restore old values of registers */
+
+			if ((old_val^new_val) == 0x0013c0de)
+				sse = 1;
+		}
+		if(!sse)
+			disable_sse(info);
 #else
-			/* no way to test, disable to be safe */
-			info->data.ia32.fxsr = info->data.ia32.sse = info->data.ia32.sse2 = info->data.ia32.sse3 = info->data.ia32.ssse3 = false;
+		/* no way to test, disable to be safe */
+		disable_sse(info);
 #endif
 #ifdef DEBUG
-		fprintf(stderr, "  SSE OS sup . %c\n", info->data.ia32.sse     ? 'Y' : 'n');
+		fprintf(stderr, "  SSE OS sup . %c\n", info->ia32.sse ? 'Y' : 'n');
 #endif
+	}
+	else /* info->ia32.sse == false */
+		disable_sse(info);
 
-		}
+	/*
+	 * now have to check for OS support of AVX instructions
+	 */
+	if(info->ia32.avx && ia32_osxsave) {
+		FLAC__uint32 ecr = FLAC__cpu_xgetbv_x86();
+		if ((ecr & 0x6) != 0x6)
+			disable_avx(info);
+#ifdef DEBUG
+		fprintf(stderr, "  AVX OS sup . %c\n", info->ia32.avx ? 'Y' : 'n');
+#endif
 	}
+	else /* no OS AVX support*/
+		disable_avx(info);
 #else
 	info->use_asm = false;
 #endif
 
 /*
- * PPC-specific
+ * x86-64-specific
  */
-#elif defined FLAC__CPU_PPC
-	info->type = FLAC__CPUINFO_TYPE_PPC;
-# if !defined FLAC__NO_ASM
+#elif defined FLAC__CPU_X86_64
+	FLAC__bool x86_osxsave = false;
+	(void) x86_osxsave; /* to avoid warnings about unused variables */
+	memset(info, 0, sizeof(*info));
+	info->type = FLAC__CPUINFO_TYPE_X86_64;
+#if !defined FLAC__NO_ASM && defined FLAC__HAS_X86INTRIN
 	info->use_asm = true;
-#  ifdef FLAC__USE_ALTIVEC
-#   if defined FLAC__SYS_DARWIN
 	{
-		int val = 0, mib[2] = { CTL_HW, HW_VECTORUNIT };
-		size_t len = sizeof(val);
-		info->data.ppc.altivec = !(sysctl(mib, 2, &val, &len, NULL, 0) || !val);
+		/* http://www.sandpile.org/x86/cpuid.htm */
+		FLAC__uint32 flags_eax, flags_ebx, flags_ecx, flags_edx;
+		FLAC__cpu_info_x86(1, &flags_eax, &flags_ebx, &flags_ecx, &flags_edx);
+		info->x86.sse3  = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_SSE3 )? true : false;
+		info->x86.ssse3 = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_SSSE3)? true : false;
+		info->x86.sse41 = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_SSE41)? true : false;
+		info->x86.sse42 = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_SSE42)? true : false;
+#if defined FLAC__AVX_SUPPORTED
+		    x86_osxsave = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_OSXSAVE)? true : false;
+		info->x86.avx   = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_AVX    )? true : false;
+		info->x86.fma   = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_FMA    )? true : false;
+		FLAC__cpu_info_x86(7, &flags_eax, &flags_ebx, &flags_ecx, &flags_edx);
+		info->x86.avx2  = (flags_ebx & FLAC__CPUINFO_IA32_CPUID_AVX2   )? true : false;
+#endif
 	}
-	{
-		host_basic_info_data_t hostInfo;
-		mach_msg_type_number_t infoCount;
-
-		infoCount = HOST_BASIC_INFO_COUNT;
-		host_info(mach_host_self(), HOST_BASIC_INFO, (host_info_t)&hostInfo, &infoCount);
+#ifdef DEBUG
+	fprintf(stderr, "CPU info (x86-64):\n");
+	fprintf(stderr, "  SSE3 ....... %c\n", info->x86.sse3  ? 'Y' : 'n');
+	fprintf(stderr, "  SSSE3 ...... %c\n", info->x86.ssse3 ? 'Y' : 'n');
+	fprintf(stderr, "  SSE41 ...... %c\n", info->x86.sse41 ? 'Y' : 'n');
+	fprintf(stderr, "  SSE42 ...... %c\n", info->x86.sse42 ? 'Y' : 'n');
+# if defined FLAC__AVX_SUPPORTED
+	fprintf(stderr, "  AVX ........ %c\n", info->x86.avx   ? 'Y' : 'n');
+	fprintf(stderr, "  FMA ........ %c\n", info->x86.fma   ? 'Y' : 'n');
+	fprintf(stderr, "  AVX2 ....... %c\n", info->x86.avx2  ? 'Y' : 'n');
+# endif
+#endif
 
-		info->data.ppc.ppc64 = (hostInfo.cpu_type == CPU_TYPE_POWERPC) && (hostInfo.cpu_subtype == CPU_SUBTYPE_POWERPC_970);
+	/*
+	 * now have to check for OS support of AVX instructions
+	 */
+	if(info->x86.avx && x86_osxsave) {
+		FLAC__uint32 ecr = FLAC__cpu_xgetbv_x86();
+		if ((ecr & 0x6) != 0x6)
+			disable_avx(info);
+#ifdef DEBUG
+		fprintf(stderr, "  AVX OS sup . %c\n", info->x86.avx ? 'Y' : 'n');
+#endif
 	}
-#   else /* FLAC__USE_ALTIVEC && !FLAC__SYS_DARWIN */
-	{
-		/* no Darwin, do it the brute-force way */
-		/* @@@@@@ this is not thread-safe; replace with SSE OS method above or remove */
-		info->data.ppc.altivec = 0;
-		info->data.ppc.ppc64 = 0;
+	else /* no OS AVX support*/
+		disable_avx(info);
+#else
+	info->use_asm = false;
+#endif
 
-		signal (SIGILL, sigill_handler);
-		canjump = 0;
-		if (!sigsetjmp (jmpbuf, 1)) {
-			canjump = 1;
+/*
+ * unknown CPU
+ */
+#else
+	info->type = FLAC__CPUINFO_TYPE_UNKNOWN;
+	info->use_asm = false;
+#endif
+}
 
-			asm volatile (
-				"mtspr 256, %0\n\t"
-				"vand %%v0, %%v0, %%v0"
-				:
-				: "r" (-1)
-			);
+#if (defined FLAC__CPU_IA32 || defined FLAC__CPU_X86_64) && defined FLAC__HAS_X86INTRIN
 
-			info->data.ppc.altivec = 1;
-		}
-		canjump = 0;
-		if (!sigsetjmp (jmpbuf, 1)) {
-			int x = 0;
-			canjump = 1;
-
-			/* PPC64 hardware implements the cntlzd instruction */
-			asm volatile ("cntlzd %0, %1" : "=r" (x) : "r" (x) );
+#if defined _MSC_VER
+#include <intrin.h> /* for __cpuid() and _xgetbv() */
+#elif defined __GNUC__ && defined HAVE_CPUID_H
+#include <cpuid.h> /* for __get_cpuid() and __get_cpuid_max() */
+#endif
 
-			info->data.ppc.ppc64 = 1;
-		}
-		signal (SIGILL, SIG_DFL); /*@@@@@@ should save and restore old signal */
+FLAC__uint32 FLAC__cpu_have_cpuid_x86(void)
+{
+#ifdef FLAC__CPU_X86_64
+	return 1;
+#else
+# if defined _MSC_VER || defined __INTEL_COMPILER /* Do they support CPUs w/o CPUID support (or OSes that work on those CPUs)? */
+	FLAC__uint32 flags1, flags2;
+	__asm {
+		pushfd
+		pushfd
+		pop		eax
+		mov		flags1, eax
+		xor		eax, 0x200000
+		push	eax
+		popfd
+		pushfd
+		pop		eax
+		mov		flags2, eax
+		popfd
 	}
-#   endif
-#  else /* !FLAC__USE_ALTIVEC */
-	info->data.ppc.altivec = 0;
-	info->data.ppc.ppc64 = 0;
-#  endif
+	if (((flags1^flags2) & 0x200000) != 0)
+		return 1;
+	else
+		return 0;
+# elif defined __GNUC__ && defined HAVE_CPUID_H
+	if (__get_cpuid_max(0, 0) != 0)
+		return 1;
+	else
+		return 0;
 # else
-	info->use_asm = false;
+	return 0;
 # endif
+#endif
+}
 
-/*
- * unknown CPI
- */
+void FLAC__cpu_info_x86(FLAC__uint32 level, FLAC__uint32 *eax, FLAC__uint32 *ebx, FLAC__uint32 *ecx, FLAC__uint32 *edx)
+{
+    (void) level;
+
+#if defined _MSC_VER || defined __INTEL_COMPILER
+	int cpuinfo[4];
+	int ext = level & 0x80000000;
+	__cpuid(cpuinfo, ext);
+	if((unsigned)cpuinfo[0] < level) {
+		*eax = *ebx = *ecx = *edx = 0;
+		return;
+	}
+#if defined FLAC__AVX_SUPPORTED
+	__cpuidex(cpuinfo, level, 0); /* for AVX2 detection */
 #else
-	info->type = FLAC__CPUINFO_TYPE_UNKNOWN;
-	info->use_asm = false;
+	__cpuid(cpuinfo, level); /* some old compilers don't support __cpuidex */
+#endif
+	*eax = cpuinfo[0]; *ebx = cpuinfo[1]; *ecx = cpuinfo[2]; *edx = cpuinfo[3];
+#elif defined __GNUC__ && defined HAVE_CPUID_H
+	FLAC__uint32 ext = level & 0x80000000;
+	__cpuid(ext, *eax, *ebx, *ecx, *edx);
+	if (*eax < level) {
+		*eax = *ebx = *ecx = *edx = 0;
+		return;
+	}
+	__cpuid_count(level, 0, *eax, *ebx, *ecx, *edx);
+#else
+	*eax = *ebx = *ecx = *edx = 0;
 #endif
 }
+
+FLAC__uint32 FLAC__cpu_xgetbv_x86(void)
+{
+#if (defined _MSC_VER || defined __INTEL_COMPILER) && defined FLAC__AVX_SUPPORTED
+	return (FLAC__uint32)_xgetbv(0);
+#elif defined __GNUC__
+	FLAC__uint32 lo, hi;
+	asm volatile (".byte 0x0f, 0x01, 0xd0" : "=a"(lo), "=d"(hi) : "c" (0));
+	return lo;
+#else
+	return 0;
+#endif
+}
+
+#endif /* (FLAC__CPU_IA32 || FLAC__CPU_X86_64) && FLAC__HAS_X86INTRIN */

source/modules/juce_audio_formats/codecs/flac/libFLAC/crc.c

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2000-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -30,7 +30,7 @@
  * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
-#if HAVE_CONFIG_H
+#ifdef HAVE_CONFIG_H
 #  include <config.h>
 #endif
 

source/modules/juce_audio_formats/codecs/flac/libFLAC/fixed.c

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2000-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -30,21 +30,17 @@
  * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
-#if HAVE_CONFIG_H
+#ifdef HAVE_CONFIG_H
 #  include <config.h>
 #endif
 
 #include <math.h>
 #include <string.h>
+#include "../compat.h"
 #include "include/private/bitmath.h"
 #include "include/private/fixed.h"
 #include "../assert.h"
 
-#ifndef M_LN2
-/* math.h in VC++ doesn't seem to have this (how Microsoft is that?) */
-#define M_LN2 0.69314718055994530942
-#endif
-
 #ifdef local_abs
 #undef local_abs
 #endif
@@ -320,20 +316,11 @@ unsigned FLAC__fixed_compute_best_predictor_wide(const FLAC__int32 data[], unsig
 	FLAC__ASSERT(data_len > 0 || total_error_3 == 0);
 	FLAC__ASSERT(data_len > 0 || total_error_4 == 0);
 #ifndef FLAC__INTEGER_ONLY_LIBRARY
-#if defined _MSC_VER || defined __MINGW32__
-	/* with MSVC you have to spoon feed it the casting */
-	residual_bits_per_sample[0] = (FLAC__float)((total_error_0 > 0) ? log(M_LN2 * (FLAC__double)(FLAC__int64)total_error_0 / (FLAC__double)data_len) / M_LN2 : 0.0);
-	residual_bits_per_sample[1] = (FLAC__float)((total_error_1 > 0) ? log(M_LN2 * (FLAC__double)(FLAC__int64)total_error_1 / (FLAC__double)data_len) / M_LN2 : 0.0);
-	residual_bits_per_sample[2] = (FLAC__float)((total_error_2 > 0) ? log(M_LN2 * (FLAC__double)(FLAC__int64)total_error_2 / (FLAC__double)data_len) / M_LN2 : 0.0);
-	residual_bits_per_sample[3] = (FLAC__float)((total_error_3 > 0) ? log(M_LN2 * (FLAC__double)(FLAC__int64)total_error_3 / (FLAC__double)data_len) / M_LN2 : 0.0);
-	residual_bits_per_sample[4] = (FLAC__float)((total_error_4 > 0) ? log(M_LN2 * (FLAC__double)(FLAC__int64)total_error_4 / (FLAC__double)data_len) / M_LN2 : 0.0);
-#else
 	residual_bits_per_sample[0] = (FLAC__float)((total_error_0 > 0) ? log(M_LN2 * (FLAC__double)total_error_0 / (FLAC__double)data_len) / M_LN2 : 0.0);
 	residual_bits_per_sample[1] = (FLAC__float)((total_error_1 > 0) ? log(M_LN2 * (FLAC__double)total_error_1 / (FLAC__double)data_len) / M_LN2 : 0.0);
 	residual_bits_per_sample[2] = (FLAC__float)((total_error_2 > 0) ? log(M_LN2 * (FLAC__double)total_error_2 / (FLAC__double)data_len) / M_LN2 : 0.0);
 	residual_bits_per_sample[3] = (FLAC__float)((total_error_3 > 0) ? log(M_LN2 * (FLAC__double)total_error_3 / (FLAC__double)data_len) / M_LN2 : 0.0);
 	residual_bits_per_sample[4] = (FLAC__float)((total_error_4 > 0) ? log(M_LN2 * (FLAC__double)total_error_4 / (FLAC__double)data_len) / M_LN2 : 0.0);
-#endif
 #else
 	residual_bits_per_sample[0] = (total_error_0 > 0) ? local__compute_rbps_wide_integerized(total_error_0, data_len) : 0;
 	residual_bits_per_sample[1] = (total_error_1 > 0) ? local__compute_rbps_wide_integerized(total_error_1, data_len) : 0;

source/modules/juce_audio_formats/codecs/flac/libFLAC/float.c

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2004-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -30,7 +30,7 @@
  * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
-#if HAVE_CONFIG_H
+#ifdef HAVE_CONFIG_H
 #  include <config.h>
 #endif
 

source/modules/juce_audio_formats/codecs/flac/libFLAC/format.c

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2000-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -30,7 +30,7 @@
  * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
-#if HAVE_CONFIG_H
+#ifdef HAVE_CONFIG_H
 #  include <config.h>
 #endif
 
@@ -45,12 +45,7 @@
 /* VERSION should come from configure */
 FLAC_API const char *FLAC__VERSION_STRING = VERSION;
 
-#if defined _MSC_VER || defined __BORLANDC__ || defined __MINW32__
-/* yet one more hack because of MSVC6: */
-FLAC_API const char *FLAC__VENDOR_STRING = "reference libFLAC 1.3.0 20130526";
-#else
-FLAC_API const char *FLAC__VENDOR_STRING = "reference libFLAC " VERSION " 20130526";
-#endif
+FLAC_API const char *FLAC__VENDOR_STRING = "reference libFLAC " VERSION " 20141125";
 
 FLAC_API const FLAC__byte FLAC__STREAM_SYNC_STRING[4] = { 'f','L','a','C' };
 FLAC_API const unsigned FLAC__STREAM_SYNC = 0x664C6143;

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/all.h

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2000-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/bitmath.h

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2001-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -34,6 +34,7 @@
 #define FLAC__PRIVATE__BITMATH_H
 
 #include "../../../ordinals.h"
+#include "../../../assert.h"
 
 /* for CHAR_BIT */
 #include <limits.h>
@@ -74,16 +75,19 @@ static inline unsigned int FLAC__clz_soft_uint32(unsigned int word)
 static inline unsigned int FLAC__clz_uint32(FLAC__uint32 v)
 {
 /* Never used with input 0 */
+    FLAC__ASSERT(v > 0);
 #if defined(__INTEL_COMPILER)
     return _bit_scan_reverse(v) ^ 31U;
 #elif defined(__GNUC__) && (__GNUC__ >= 4 || (__GNUC__ == 3 && __GNUC_MINOR__ >= 4))
 /* This will translate either to (bsr ^ 31U), clz , ctlz, cntlz, lzcnt depending on
- * -march= setting or to a software rutine in exotic machines. */
+ * -march= setting or to a software routine in exotic machines. */
     return __builtin_clz(v);
 #elif defined(_MSC_VER) && (_MSC_VER >= 1400)
-    FLAC__uint32 idx;
-    _BitScanReverse((DWORD*) &idx, v);
-    return idx ^ 31U;
+    {
+        unsigned long idx;
+        _BitScanReverse(&idx, v);
+        return idx ^ 31U;
+    }
 #else
     return FLAC__clz_soft_uint32(v);
 #endif
@@ -99,7 +103,7 @@ static inline unsigned int FLAC__clz2_uint32(FLAC__uint32 v)
 
 /* An example of what FLAC__bitmath_ilog2() computes:
  *
- * ilog2( 0) = undefined
+ * ilog2( 0) = assertion failure
  * ilog2( 1) = 0
  * ilog2( 2) = 1
  * ilog2( 3) = 1
@@ -122,45 +126,56 @@ static inline unsigned int FLAC__clz2_uint32(FLAC__uint32 v)
 
 static inline unsigned FLAC__bitmath_ilog2(FLAC__uint32 v)
 {
+    FLAC__ASSERT(v > 0);
+#if defined(__INTEL_COMPILER)
+    return _bit_scan_reverse(v);
+#elif defined(_MSC_VER) && (_MSC_VER >= 1400)
+    {
+        unsigned long idx;
+        _BitScanReverse(&idx, v);
+        return idx;
+    }
+#else
     return sizeof(FLAC__uint32) * CHAR_BIT  - 1 - FLAC__clz_uint32(v);
+#endif
 }
 
 
-#ifdef FLAC__INTEGER_ONLY_LIBRARY /*Unused otherwise */
+#ifdef FLAC__INTEGER_ONLY_LIBRARY /* Unused otherwise */
 
 static inline unsigned FLAC__bitmath_ilog2_wide(FLAC__uint64 v)
 {
-    if (v == 0)
-		return 0;
+    FLAC__ASSERT(v > 0);
 #if defined(__GNUC__) && (__GNUC__ >= 4 || (__GNUC__ == 3 && __GNUC_MINOR__ >= 4))
     return sizeof(FLAC__uint64) * CHAR_BIT - 1 - __builtin_clzll(v);
-/* Sorry, only supported in win64/Itanium.. */
-#elif (defined(_MSC_VER) && (_MSC_VER >= 1400)) && (defined(_M_IA64) || defined(_WIN64))
-    FLAC__uint64 idx;
-    _BitScanReverse64(&idx, v);
-    return idx ^ 63U;
+/* Sorry, only supported in x64/Itanium.. and both have fast FPU which makes integer-only encoder pointless */
+#elif (defined(_MSC_VER) && (_MSC_VER >= 1400)) && (defined(_M_IA64) || defined(_M_X64))
+    {
+        unsigned long idx;
+        _BitScanReverse64(&idx, v);
+        return idx;
+    }
 #else
-/* Brain-damaged compilers will use the fastest possible way that is,
+/*  Brain-damaged compilers will use the fastest possible way that is,
     de Bruijn sequences (http://supertech.csail.mit.edu/papers/debruijn.pdf)
-    (C) Timothy B. Terriberry (tterribe@xiph.org) 2001-2009 LGPL (v2 or later).
+    (C) Timothy B. Terriberry (tterribe@xiph.org) 2001-2009 CC0 (Public domain).
 */
-    static const unsigned char DEBRUIJN_IDX64[64]={
-        0, 1, 2, 7, 3,13, 8,19, 4,25,14,28, 9,34,20,40,
-        5,17,26,38,15,46,29,48,10,31,35,54,21,50,41,57,
-        63, 6,12,18,24,27,33,39,16,37,45,47,30,53,49,56,
-        62,11,23,32,36,44,52,55,61,22,43,51,60,42,59,58
-    };
-    int ret;
-    ret= v>0;
-    v|= v>>1;
-    v|= v>>2;
-    v|= v>>4;
-    v|= v>>8;
-    v|= v>>16;
-    v|= v>>32;
-    v= (v>>1)+1;
-    ret+=DEBRUIJN_IDX64[v*0x218A392CD3D5DBF>>58&0x3F];
-    return ret;
+    {
+        static const unsigned char DEBRUIJN_IDX64[64]={
+            0, 1, 2, 7, 3,13, 8,19, 4,25,14,28, 9,34,20,40,
+            5,17,26,38,15,46,29,48,10,31,35,54,21,50,41,57,
+            63, 6,12,18,24,27,33,39,16,37,45,47,30,53,49,56,
+            62,11,23,32,36,44,52,55,61,22,43,51,60,42,59,58
+        };
+        v|= v>>1;
+        v|= v>>2;
+        v|= v>>4;
+        v|= v>>8;
+        v|= v>>16;
+        v|= v>>32;
+        v= (v>>1)+1;
+        return DEBRUIJN_IDX64[v*0x218A392CD3D5DBF>>58&0x3F];
+    }
 #endif
 }
 #endif

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/bitreader.h

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2000-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -50,7 +50,7 @@ typedef FLAC__bool (*FLAC__BitReaderReadCallback)(FLAC__byte buffer[], size_t *b
  */
 FLAC__BitReader *FLAC__bitreader_new(void);
 void FLAC__bitreader_delete(FLAC__BitReader *br);
-FLAC__bool FLAC__bitreader_init(FLAC__BitReader *br, FLAC__CPUInfo cpu, FLAC__BitReaderReadCallback rcb, void *cd);
+FLAC__bool FLAC__bitreader_init(FLAC__BitReader *br, FLAC__BitReaderReadCallback rcb, void *cd);
 void FLAC__bitreader_free(FLAC__BitReader *br); /* does not 'free(br)' */
 FLAC__bool FLAC__bitreader_clear(FLAC__BitReader *br);
 void FLAC__bitreader_dump(const FLAC__BitReader *br, FILE *out);
@@ -82,19 +82,10 @@ FLAC__bool FLAC__bitreader_read_byte_block_aligned_no_crc(FLAC__BitReader *br, F
 FLAC__bool FLAC__bitreader_read_unary_unsigned(FLAC__BitReader *br, unsigned *val);
 FLAC__bool FLAC__bitreader_read_rice_signed(FLAC__BitReader *br, int *val, unsigned parameter);
 FLAC__bool FLAC__bitreader_read_rice_signed_block(FLAC__BitReader *br, int vals[], unsigned nvals, unsigned parameter);
-#ifndef FLAC__NO_ASM
-#  ifdef FLAC__CPU_IA32
-#    ifdef FLAC__HAS_NASM
-FLAC__bool FLAC__bitreader_read_rice_signed_block_asm_ia32_bswap(FLAC__BitReader *br, int vals[], unsigned nvals, unsigned parameter);
-#    endif
-#  endif
-#endif
 #if 0 /* UNUSED */
 FLAC__bool FLAC__bitreader_read_golomb_signed(FLAC__BitReader *br, int *val, unsigned parameter);
 FLAC__bool FLAC__bitreader_read_golomb_unsigned(FLAC__BitReader *br, unsigned *val, unsigned parameter);
 #endif
 FLAC__bool FLAC__bitreader_read_utf8_uint32(FLAC__BitReader *br, FLAC__uint32 *val, FLAC__byte *raw, unsigned *rawlen);
 FLAC__bool FLAC__bitreader_read_utf8_uint64(FLAC__BitReader *br, FLAC__uint64 *val, FLAC__byte *raw, unsigned *rawlen);
-
-FLAC__bool bitreader_read_from_client_(FLAC__BitReader *br);
 #endif

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/bitwriter.h

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2000-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/cpu.h

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2001-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -41,49 +41,59 @@
 
 typedef enum {
 	FLAC__CPUINFO_TYPE_IA32,
-	FLAC__CPUINFO_TYPE_PPC,
+	FLAC__CPUINFO_TYPE_X86_64,
 	FLAC__CPUINFO_TYPE_UNKNOWN
 } FLAC__CPUInfo_Type;
 
+#if defined FLAC__CPU_IA32
 typedef struct {
-	FLAC__bool cpuid;
-	FLAC__bool bswap;
 	FLAC__bool cmov;
 	FLAC__bool mmx;
-	FLAC__bool fxsr;
 	FLAC__bool sse;
 	FLAC__bool sse2;
+
 	FLAC__bool sse3;
 	FLAC__bool ssse3;
-	FLAC__bool _3dnow;
-	FLAC__bool ext3dnow;
-	FLAC__bool extmmx;
+	FLAC__bool sse41;
+	FLAC__bool sse42;
+	FLAC__bool avx;
+	FLAC__bool avx2;
+	FLAC__bool fma;
 } FLAC__CPUInfo_IA32;
-
+#elif defined FLAC__CPU_X86_64
 typedef struct {
-	FLAC__bool altivec;
-	FLAC__bool ppc64;
-} FLAC__CPUInfo_PPC;
+	FLAC__bool sse3;
+	FLAC__bool ssse3;
+	FLAC__bool sse41;
+	FLAC__bool sse42;
+	FLAC__bool avx;
+	FLAC__bool avx2;
+	FLAC__bool fma;
+} FLAC__CPUInfo_x86;
+#endif
 
 typedef struct {
 	FLAC__bool use_asm;
 	FLAC__CPUInfo_Type type;
-	union {
-		FLAC__CPUInfo_IA32 ia32;
-		FLAC__CPUInfo_PPC ppc;
-	} data;
+#if defined FLAC__CPU_IA32
+	FLAC__CPUInfo_IA32 ia32;
+#elif defined FLAC__CPU_X86_64
+	FLAC__CPUInfo_x86 x86;
+#endif
 } FLAC__CPUInfo;
 
 void FLAC__cpu_info(FLAC__CPUInfo *info);
 
 #ifndef FLAC__NO_ASM
-#ifdef FLAC__CPU_IA32
-#ifdef FLAC__HAS_NASM
+# if defined FLAC__CPU_IA32 && defined FLAC__HAS_NASM
 FLAC__uint32 FLAC__cpu_have_cpuid_asm_ia32(void);
 void         FLAC__cpu_info_asm_ia32(FLAC__uint32 *flags_edx, FLAC__uint32 *flags_ecx);
-FLAC__uint32 FLAC__cpu_info_extended_amd_asm_ia32(void);
-#endif
-#endif
+# endif
+# if (defined FLAC__CPU_IA32 || defined FLAC__CPU_X86_64) && defined FLAC__HAS_X86INTRIN
+FLAC__uint32 FLAC__cpu_have_cpuid_x86(void);
+void         FLAC__cpu_info_x86(FLAC__uint32 level, FLAC__uint32 *eax, FLAC__uint32 *ebx, FLAC__uint32 *ecx, FLAC__uint32 *edx);
+FLAC__uint32 FLAC__cpu_xgetbv_x86(void);
+# endif
 #endif
 
 #endif

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/crc.h

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2000-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -51,7 +51,7 @@ FLAC__uint8 FLAC__crc8(const FLAC__byte *data, unsigned len);
 */
 extern unsigned const FLAC__crc16_table[256];
 
-#define FLAC__CRC16_UPDATE(data, crc) (((((crc)<<8) & 0xffff) ^ FLAC__crc16_table[((crc)>>8) ^ (data)]))
+#define FLAC__CRC16_UPDATE(data, crc) ((((crc)<<8) & 0xffff) ^ FLAC__crc16_table[((crc)>>8) ^ (data)])
 /* this alternate may be faster on some systems/compilers */
 #if 0
 #define FLAC__CRC16_UPDATE(data, crc) ((((crc)<<8) ^ FLAC__crc16_table[((crc)>>8) ^ (data)]) & 0xffff)

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/fixed.h

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2000-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -37,6 +37,7 @@
 #include <config.h>
 #endif
 
+#include "cpu.h"
 #include "float.h"
 #include "../../../format.h"
 
@@ -54,14 +55,22 @@
  */
 #ifndef FLAC__INTEGER_ONLY_LIBRARY
 unsigned FLAC__fixed_compute_best_predictor(const FLAC__int32 data[], unsigned data_len, FLAC__float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1]);
+unsigned FLAC__fixed_compute_best_predictor_wide(const FLAC__int32 data[], unsigned data_len, FLAC__float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1]);
 # ifndef FLAC__NO_ASM
-#  ifdef FLAC__CPU_IA32
-#   ifdef FLAC__HAS_NASM
-unsigned FLAC__fixed_compute_best_predictor_asm_ia32_mmx_cmov(const FLAC__int32 data[], unsigned data_len, FLAC__float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1]);
+#  if (defined FLAC__CPU_IA32 || defined FLAC__CPU_X86_64) && defined FLAC__HAS_X86INTRIN
+#   ifdef FLAC__SSE2_SUPPORTED
+unsigned FLAC__fixed_compute_best_predictor_intrin_sse2(const FLAC__int32 data[], unsigned data_len, FLAC__float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER + 1]);
+unsigned FLAC__fixed_compute_best_predictor_wide_intrin_sse2(const FLAC__int32 data[], unsigned data_len, FLAC__float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER + 1]);
+#   endif
+#   ifdef FLAC__SSSE3_SUPPORTED
+unsigned FLAC__fixed_compute_best_predictor_intrin_ssse3(const FLAC__int32 data[], unsigned data_len, FLAC__float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1]);
+unsigned FLAC__fixed_compute_best_predictor_wide_intrin_ssse3(const FLAC__int32 data[], unsigned data_len, FLAC__float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER + 1]);
 #   endif
 #  endif
+#  if defined FLAC__CPU_IA32 && defined FLAC__HAS_NASM
+unsigned FLAC__fixed_compute_best_predictor_asm_ia32_mmx_cmov(const FLAC__int32 data[], unsigned data_len, FLAC__float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1]);
+#  endif
 # endif
-unsigned FLAC__fixed_compute_best_predictor_wide(const FLAC__int32 data[], unsigned data_len, FLAC__float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1]);
 #else
 unsigned FLAC__fixed_compute_best_predictor(const FLAC__int32 data[], unsigned data_len, FLAC__fixedpoint residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1]);
 unsigned FLAC__fixed_compute_best_predictor_wide(const FLAC__int32 data[], unsigned data_len, FLAC__fixedpoint residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1]);

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/float.h

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2004-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/format.h

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2000-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/lpc.h

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2000-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -37,6 +37,7 @@
 #include <config.h>
 #endif
 
+#include "cpu.h"
 #include "float.h"
 #include "../../../format.h"
 
@@ -75,7 +76,15 @@ void FLAC__lpc_compute_autocorrelation_asm_ia32(const FLAC__real data[], unsigne
 void FLAC__lpc_compute_autocorrelation_asm_ia32_sse_lag_4(const FLAC__real data[], unsigned data_len, unsigned lag, FLAC__real autoc[]);
 void FLAC__lpc_compute_autocorrelation_asm_ia32_sse_lag_8(const FLAC__real data[], unsigned data_len, unsigned lag, FLAC__real autoc[]);
 void FLAC__lpc_compute_autocorrelation_asm_ia32_sse_lag_12(const FLAC__real data[], unsigned data_len, unsigned lag, FLAC__real autoc[]);
-void FLAC__lpc_compute_autocorrelation_asm_ia32_3dnow(const FLAC__real data[], unsigned data_len, unsigned lag, FLAC__real autoc[]);
+void FLAC__lpc_compute_autocorrelation_asm_ia32_sse_lag_16(const FLAC__real data[], unsigned data_len, unsigned lag, FLAC__real autoc[]);
+#    endif
+#  endif
+#  if (defined FLAC__CPU_IA32 || defined FLAC__CPU_X86_64) && defined FLAC__HAS_X86INTRIN
+#    ifdef FLAC__SSE_SUPPORTED
+void FLAC__lpc_compute_autocorrelation_intrin_sse_lag_4(const FLAC__real data[], unsigned data_len, unsigned lag, FLAC__real autoc[]);
+void FLAC__lpc_compute_autocorrelation_intrin_sse_lag_8(const FLAC__real data[], unsigned data_len, unsigned lag, FLAC__real autoc[]);
+void FLAC__lpc_compute_autocorrelation_intrin_sse_lag_12(const FLAC__real data[], unsigned data_len, unsigned lag, FLAC__real autoc[]);
+void FLAC__lpc_compute_autocorrelation_intrin_sse_lag_16(const FLAC__real data[], unsigned data_len, unsigned lag, FLAC__real autoc[]);
 #    endif
 #  endif
 #endif
@@ -145,6 +154,22 @@ void FLAC__lpc_compute_residual_from_qlp_coefficients_wide(const FLAC__int32 *da
 #    ifdef FLAC__HAS_NASM
 void FLAC__lpc_compute_residual_from_qlp_coefficients_asm_ia32(const FLAC__int32 *data, unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 residual[]);
 void FLAC__lpc_compute_residual_from_qlp_coefficients_asm_ia32_mmx(const FLAC__int32 *data, unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 residual[]);
+void FLAC__lpc_compute_residual_from_qlp_coefficients_wide_asm_ia32(const FLAC__int32 *data, unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 residual[]);
+#    endif
+#  endif
+#  if (defined FLAC__CPU_IA32 || defined FLAC__CPU_X86_64) && defined FLAC__HAS_X86INTRIN
+#    ifdef FLAC__SSE2_SUPPORTED
+void FLAC__lpc_compute_residual_from_qlp_coefficients_16_intrin_sse2(const FLAC__int32 *data, unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 residual[]);
+void FLAC__lpc_compute_residual_from_qlp_coefficients_intrin_sse2(const FLAC__int32 *data, unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 residual[]);
+#    endif
+#    ifdef FLAC__SSE4_1_SUPPORTED
+void FLAC__lpc_compute_residual_from_qlp_coefficients_intrin_sse41(const FLAC__int32 *data, unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 residual[]);
+void FLAC__lpc_compute_residual_from_qlp_coefficients_wide_intrin_sse41(const FLAC__int32 *data, unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 residual[]);
+#    endif
+#    ifdef FLAC__AVX2_SUPPORTED
+void FLAC__lpc_compute_residual_from_qlp_coefficients_16_intrin_avx2(const FLAC__int32 *data, unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 residual[]);
+void FLAC__lpc_compute_residual_from_qlp_coefficients_intrin_avx2(const FLAC__int32 *data, unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 residual[]);
+void FLAC__lpc_compute_residual_from_qlp_coefficients_wide_intrin_avx2(const FLAC__int32 *data, unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 residual[]);
 #    endif
 #  endif
 #endif
@@ -173,11 +198,17 @@ void FLAC__lpc_restore_signal_wide(const FLAC__int32 residual[], unsigned data_l
 #    ifdef FLAC__HAS_NASM
 void FLAC__lpc_restore_signal_asm_ia32(const FLAC__int32 residual[], unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 data[]);
 void FLAC__lpc_restore_signal_asm_ia32_mmx(const FLAC__int32 residual[], unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 data[]);
+void FLAC__lpc_restore_signal_wide_asm_ia32(const FLAC__int32 residual[], unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 data[]);
 #    endif /* FLAC__HAS_NASM */
-#  elif defined FLAC__CPU_PPC
-void FLAC__lpc_restore_signal_asm_ppc_altivec_16(const FLAC__int32 residual[], unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 data[]);
-void FLAC__lpc_restore_signal_asm_ppc_altivec_16_order8(const FLAC__int32 residual[], unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 data[]);
-#  endif/* FLAC__CPU_IA32 || FLAC__CPU_PPC */
+#  endif /* FLAC__CPU_IA32 */
+#  if (defined FLAC__CPU_IA32 || defined FLAC__CPU_X86_64) && defined FLAC__HAS_X86INTRIN
+#    ifdef FLAC__SSE2_SUPPORTED
+void FLAC__lpc_restore_signal_16_intrin_sse2(const FLAC__int32 residual[], unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 data[]);
+#    endif
+#    ifdef FLAC__SSE4_1_SUPPORTED
+void FLAC__lpc_restore_signal_wide_intrin_sse41(const FLAC__int32 residual[], unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 data[]);
+#    endif
+#  endif
 #endif /* FLAC__NO_ASM */
 
 #ifndef FLAC__INTEGER_ONLY_LIBRARY

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/md5.h

@@ -28,11 +28,17 @@
 
 #include "../../../ordinals.h"
 
+typedef union {
+	FLAC__byte *p8;
+	FLAC__int16 *p16;
+	FLAC__int32 *p32;
+} FLAC__multibyte;
+
 typedef struct {
 	FLAC__uint32 in[16];
 	FLAC__uint32 buf[4];
 	FLAC__uint32 bytes[2];
-	FLAC__byte *internal_buf;
+	FLAC__multibyte internal_buf;
 	size_t capacity;
 } FLAC__MD5Context;
 

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/memory.h

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2001-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/metadata.h

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2002-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/stream_encoder.h

@@ -0,0 +1,67 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef FLAC__PRIVATE__STREAM_ENCODER_H
+#define FLAC__PRIVATE__STREAM_ENCODER_H
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+/*
+ * This is used to avoid overflow with unusual signals in 32-bit
+ * accumulator in the *precompute_partition_info_sums_* functions.
+ */
+#define FLAC__MAX_EXTRA_RESIDUAL_BPS 4
+
+#if (defined FLAC__CPU_IA32 || defined FLAC__CPU_X86_64) && defined FLAC__HAS_X86INTRIN
+#include "cpu.h"
+#include "../../../format.h"
+
+#ifdef FLAC__SSE2_SUPPORTED
+extern void FLAC__precompute_partition_info_sums_intrin_sse2(const FLAC__int32 residual[], FLAC__uint64 abs_residual_partition_sums[],
+			unsigned residual_samples, unsigned predictor_order, unsigned min_partition_order, unsigned max_partition_order, unsigned bps);
+#endif
+
+#ifdef FLAC__SSSE3_SUPPORTED
+extern void FLAC__precompute_partition_info_sums_intrin_ssse3(const FLAC__int32 residual[], FLAC__uint64 abs_residual_partition_sums[],
+			unsigned residual_samples, unsigned predictor_order, unsigned min_partition_order, unsigned max_partition_order, unsigned bps);
+#endif
+
+#ifdef FLAC__AVX2_SUPPORTED
+extern void FLAC__precompute_partition_info_sums_intrin_avx2(const FLAC__int32 residual[], FLAC__uint64 abs_residual_partition_sums[],
+			unsigned residual_samples, unsigned predictor_order, unsigned min_partition_order, unsigned max_partition_order, unsigned bps);
+#endif
+
+#endif
+
+#endif

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/stream_encoder_framing.h

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2000-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/window.h

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2006-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -65,6 +65,8 @@ void FLAC__window_nuttall(FLAC__real *window, const FLAC__int32 L);
 void FLAC__window_rectangle(FLAC__real *window, const FLAC__int32 L);
 void FLAC__window_triangle(FLAC__real *window, const FLAC__int32 L);
 void FLAC__window_tukey(FLAC__real *window, const FLAC__int32 L, const FLAC__real p);
+void FLAC__window_partial_tukey(FLAC__real *window, const FLAC__int32 L, const FLAC__real p, const FLAC__real start, const FLAC__real end);
+void FLAC__window_punchout_tukey(FLAC__real *window, const FLAC__int32 L, const FLAC__real p, const FLAC__real start, const FLAC__real end);
 void FLAC__window_welch(FLAC__real *window, const FLAC__int32 L);
 
 #endif /* !defined FLAC__INTEGER_ONLY_LIBRARY */

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/protected/all.h

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2001-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/protected/stream_decoder.h

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2000-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -40,6 +40,7 @@
 
 typedef struct FLAC__StreamDecoderProtected {
 	FLAC__StreamDecoderState state;
+	FLAC__StreamDecoderInitStatus initstate;
 	unsigned channels;
 	FLAC__ChannelAssignment channel_assignment;
 	unsigned bits_per_sample;

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/protected/stream_encoder.h

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2001-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -59,6 +59,8 @@ typedef enum {
 	FLAC__APODIZATION_RECTANGLE,
 	FLAC__APODIZATION_TRIANGLE,
 	FLAC__APODIZATION_TUKEY,
+	FLAC__APODIZATION_PARTIAL_TUKEY,
+	FLAC__APODIZATION_PUNCHOUT_TUKEY,
 	FLAC__APODIZATION_WELCH
 } FLAC__ApodizationFunction;
 
@@ -71,6 +73,11 @@ typedef struct {
 		struct {
 			FLAC__real p;
 		} tukey;
+		struct {
+			FLAC__real p;
+			FLAC__real start;
+			FLAC__real end;
+		} multiple_tukey;
 	} parameters;
 } FLAC__ApodizationSpecification;
 

source/modules/juce_audio_formats/codecs/flac/libFLAC/lpc_flac.c

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2000-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -30,7 +30,7 @@
  * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
-#if HAVE_CONFIG_H
+#ifdef HAVE_CONFIG_H
 #  include <config.h>
 #endif
 
@@ -50,11 +50,6 @@
 
 #ifndef FLAC__INTEGER_ONLY_LIBRARY
 
-#ifndef M_LN2
-/* math.h in VC++ doesn't seem to have this (how Microsoft is that?) */
-#define M_LN2 0.69314718055994530942
-#endif
-
 #if !defined(HAVE_LROUND)
 #if defined(_MSC_VER)
 #include <float.h>
@@ -65,7 +60,7 @@
 static inline long int lround(double x) {
     return (long)(x + copysign (0.5, x));
 }
-//If this fails, we are in the precence of a mid 90's compiler..move along...
+/* If this fails, we are in the presence of a mid 90's compiler, move along... */
 #endif
 
 void FLAC__lpc_window_data(const FLAC__int32 in[], const FLAC__real window[], FLAC__real out[], unsigned data_len)
@@ -160,7 +155,7 @@ void FLAC__lpc_compute_lp_coefficients(const FLAC__real autoc[], unsigned *max_o
 			lp_coeff[i][j] = (FLAC__real)(-lpc[j]); /* negate FIR filter coeff to get predictor coeff */
 		error[i] = err;
 
-		/* see SF bug #1601812 http://sourceforge.net/tracker/index.php?func=detail&aid=1601812&group_id=13478&atid=113478 */
+		/* see SF bug https://sourceforge.net/p/flac/bugs/234/ */
 		if(err == 0.0) {
 			*max_order = i+1;
 			return;
@@ -264,7 +259,12 @@ int FLAC__lpc_quantize_coefficients(const FLAC__real lp_coeff[], unsigned order,
 	return 0;
 }
 
-void FLAC__lpc_compute_residual_from_qlp_coefficients(const FLAC__int32 *data, unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 residual[])
+#if defined(_MSC_VER)
+// silence MSVC warnings about __restrict modifier
+#pragma warning ( disable : 4028 )
+#endif
+
+void FLAC__lpc_compute_residual_from_qlp_coefficients(const FLAC__int32 * flac_restrict data, unsigned data_len, const FLAC__int32 * flac_restrict qlp_coeff, unsigned order, int lp_quantization, FLAC__int32 * flac_restrict residual)
 #if defined(FLAC__OVERFLOW_DETECT) || !defined(FLAC__LPC_UNROLLED_FILTER_LOOPS)
 {
 	FLAC__int64 sumo;
@@ -524,7 +524,7 @@ void FLAC__lpc_compute_residual_from_qlp_coefficients(const FLAC__int32 *data, u
 }
 #endif
 
-void FLAC__lpc_compute_residual_from_qlp_coefficients_wide(const FLAC__int32 *data, unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 residual[])
+void FLAC__lpc_compute_residual_from_qlp_coefficients_wide(const FLAC__int32 * flac_restrict data, unsigned data_len, const FLAC__int32 * flac_restrict qlp_coeff, unsigned order, int lp_quantization, FLAC__int32 * flac_restrict residual)
 #if defined(FLAC__OVERFLOW_DETECT) || !defined(FLAC__LPC_UNROLLED_FILTER_LOOPS)
 {
 	unsigned i, j;
@@ -780,7 +780,7 @@ void FLAC__lpc_compute_residual_from_qlp_coefficients_wide(const FLAC__int32 *da
 
 #endif /* !defined FLAC__INTEGER_ONLY_LIBRARY */
 
-void FLAC__lpc_restore_signal(const FLAC__int32 residual[], unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 data[])
+void FLAC__lpc_restore_signal(const FLAC__int32 * flac_restrict residual, unsigned data_len, const FLAC__int32 * flac_restrict qlp_coeff, unsigned order, int lp_quantization, FLAC__int32 * flac_restrict data)
 #if defined(FLAC__OVERFLOW_DETECT) || !defined(FLAC__LPC_UNROLLED_FILTER_LOOPS)
 {
 	FLAC__int64 sumo;
@@ -1041,7 +1041,7 @@ void FLAC__lpc_restore_signal(const FLAC__int32 residual[], unsigned data_len, c
 }
 #endif
 
-void FLAC__lpc_restore_signal_wide(const FLAC__int32 residual[], unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 data[])
+void FLAC__lpc_restore_signal_wide(const FLAC__int32 * flac_restrict residual, unsigned data_len, const FLAC__int32 * flac_restrict qlp_coeff, unsigned order, int lp_quantization, FLAC__int32 * flac_restrict data)
 #if defined(FLAC__OVERFLOW_DETECT) || !defined(FLAC__LPC_UNROLLED_FILTER_LOOPS)
 {
 	unsigned i, j;
@@ -1295,6 +1295,10 @@ void FLAC__lpc_restore_signal_wide(const FLAC__int32 residual[], unsigned data_l
 }
 #endif
 
+#if defined(_MSC_VER)
+#pragma warning ( default : 4028 )
+#endif
+
 #ifndef FLAC__INTEGER_ONLY_LIBRARY
 
 FLAC__double FLAC__lpc_compute_expected_bits_per_residual_sample(FLAC__double lpc_error, unsigned total_samples)

source/modules/juce_audio_formats/codecs/flac/libFLAC/md5.c

@@ -1,4 +1,4 @@
-#if HAVE_CONFIG_H
+#ifdef HAVE_CONFIG_H
 #  include <config.h>
 #endif
 
@@ -7,6 +7,7 @@
 
 #include "include/private/md5.h"
 #include "../alloc.h"
+#include "../endswap.h"
 
 /*
  * This code implements the MD5 message-digest algorithm.
@@ -223,7 +224,7 @@ void FLAC__MD5Init(FLAC__MD5Context *ctx)
 	ctx->bytes[0] = 0;
 	ctx->bytes[1] = 0;
 
-	ctx->internal_buf = 0;
+	ctx->internal_buf.p8= 0;
 	ctx->capacity = 0;
 }
 
@@ -259,9 +260,9 @@ void FLAC__MD5Final(FLAC__byte digest[16], FLAC__MD5Context *ctx)
 
 	byteSwap(ctx->buf, 4);
 	memcpy(digest, ctx->buf, 16);
-	if(0 != ctx->internal_buf) {
-		free(ctx->internal_buf);
-		ctx->internal_buf = 0;
+	if (0 != ctx->internal_buf.p8) {
+		free(ctx->internal_buf.p8);
+		ctx->internal_buf.p8= 0;
 		ctx->capacity = 0;
 	}
 	memset(ctx, 0, sizeof(*ctx));	/* In case it's sensitive */
@@ -270,58 +271,124 @@ void FLAC__MD5Final(FLAC__byte digest[16], FLAC__MD5Context *ctx)
 /*
  * Convert the incoming audio signal to a byte stream
  */
-static void format_input_(FLAC__byte *buf, const FLAC__int32 * const signal[], unsigned channels, unsigned samples, unsigned bytes_per_sample)
+static void format_input_(FLAC__multibyte *mbuf, const FLAC__int32 * const signal[], unsigned channels, unsigned samples, unsigned bytes_per_sample)
 {
+	FLAC__byte *buf_ = mbuf->p8;
+	FLAC__int16 *buf16 = mbuf->p16;
+	FLAC__int32 *buf32 = mbuf->p32;
+	FLAC__int32 a_word;
 	unsigned channel, sample;
-	register FLAC__int32 a_word;
-	register FLAC__byte *buf_ = buf;
 
-#if WORDS_BIGENDIAN
-#else
-	if(channels == 2 && bytes_per_sample == 2) {
-		FLAC__int16 *buf1_ = ((FLAC__int16*)buf_) + 1;
-		memcpy(buf_, signal[0], sizeof(FLAC__int32) * samples);
-		for(sample = 0; sample < samples; sample++, buf1_+=2)
-			*buf1_ = (FLAC__int16)signal[1][sample];
-	}
-	else if(channels == 1 && bytes_per_sample == 2) {
-		FLAC__int16 *buf1_ = (FLAC__int16*)buf_;
-		for(sample = 0; sample < samples; sample++)
-			*buf1_++ = (FLAC__int16)signal[0][sample];
-	}
-	else
-#endif
-	if(bytes_per_sample == 2) {
-		if(channels == 2) {
-			for(sample = 0; sample < samples; sample++) {
-				a_word = signal[0][sample];
-				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
-				*buf_++ = (FLAC__byte)a_word;
-				a_word = signal[1][sample];
-				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
-				*buf_++ = (FLAC__byte)a_word;
+	/* Storage in the output buffer, buf, is little endian. */
+
+#define BYTES_CHANNEL_SELECTOR(bytes, channels)   (bytes * 100 + channels)
+
+	/* First do the most commonly used combinations. */
+	switch (BYTES_CHANNEL_SELECTOR (bytes_per_sample, channels)) {
+		/* One byte per sample. */
+		case (BYTES_CHANNEL_SELECTOR (1, 1)):
+			for (sample = 0; sample < samples; sample++)
+				*buf_++ = signal[0][sample];
+			return;
+
+		case (BYTES_CHANNEL_SELECTOR (1, 2)):
+			for (sample = 0; sample < samples; sample++) {
+				*buf_++ = signal[0][sample];
+				*buf_++ = signal[1][sample];
 			}
-		}
-		else if(channels == 1) {
-			for(sample = 0; sample < samples; sample++) {
+			return;
+
+		case (BYTES_CHANNEL_SELECTOR (1, 4)):
+			for (sample = 0; sample < samples; sample++) {
+				*buf_++ = signal[0][sample];
+				*buf_++ = signal[1][sample];
+				*buf_++ = signal[2][sample];
+				*buf_++ = signal[3][sample];
+			}
+			return;
+
+		case (BYTES_CHANNEL_SELECTOR (1, 6)):
+			for (sample = 0; sample < samples; sample++) {
+				*buf_++ = signal[0][sample];
+				*buf_++ = signal[1][sample];
+				*buf_++ = signal[2][sample];
+				*buf_++ = signal[3][sample];
+				*buf_++ = signal[4][sample];
+				*buf_++ = signal[5][sample];
+			}
+			return;
+
+		case (BYTES_CHANNEL_SELECTOR (1, 8)):
+			for (sample = 0; sample < samples; sample++) {
+				*buf_++ = signal[0][sample];
+				*buf_++ = signal[1][sample];
+				*buf_++ = signal[2][sample];
+				*buf_++ = signal[3][sample];
+				*buf_++ = signal[4][sample];
+				*buf_++ = signal[5][sample];
+				*buf_++ = signal[6][sample];
+				*buf_++ = signal[7][sample];
+			}
+			return;
+
+		/* Two bytes per sample. */
+		case (BYTES_CHANNEL_SELECTOR (2, 1)):
+			for (sample = 0; sample < samples; sample++)
+				*buf16++ = H2LE_16(signal[0][sample]);
+			return;
+
+		case (BYTES_CHANNEL_SELECTOR (2, 2)):
+			for (sample = 0; sample < samples; sample++) {
+				*buf16++ = H2LE_16(signal[0][sample]);
+				*buf16++ = H2LE_16(signal[1][sample]);
+			}
+			return;
+
+		case (BYTES_CHANNEL_SELECTOR (2, 4)):
+			for (sample = 0; sample < samples; sample++) {
+				*buf16++ = H2LE_16(signal[0][sample]);
+				*buf16++ = H2LE_16(signal[1][sample]);
+				*buf16++ = H2LE_16(signal[2][sample]);
+				*buf16++ = H2LE_16(signal[3][sample]);
+			}
+			return;
+
+		case (BYTES_CHANNEL_SELECTOR (2, 6)):
+			for (sample = 0; sample < samples; sample++) {
+				*buf16++ = H2LE_16(signal[0][sample]);
+				*buf16++ = H2LE_16(signal[1][sample]);
+				*buf16++ = H2LE_16(signal[2][sample]);
+				*buf16++ = H2LE_16(signal[3][sample]);
+				*buf16++ = H2LE_16(signal[4][sample]);
+				*buf16++ = H2LE_16(signal[5][sample]);
+			}
+			return;
+
+		case (BYTES_CHANNEL_SELECTOR (2, 8)):
+			for (sample = 0; sample < samples; sample++) {
+				*buf16++ = H2LE_16(signal[0][sample]);
+				*buf16++ = H2LE_16(signal[1][sample]);
+				*buf16++ = H2LE_16(signal[2][sample]);
+				*buf16++ = H2LE_16(signal[3][sample]);
+				*buf16++ = H2LE_16(signal[4][sample]);
+				*buf16++ = H2LE_16(signal[5][sample]);
+				*buf16++ = H2LE_16(signal[6][sample]);
+				*buf16++ = H2LE_16(signal[7][sample]);
+			}
+			return;
+
+		/* Three bytes per sample. */
+		case (BYTES_CHANNEL_SELECTOR (3, 1)):
+			for (sample = 0; sample < samples; sample++) {
 				a_word = signal[0][sample];
 				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
 				*buf_++ = (FLAC__byte)a_word;
 			}
-		}
-		else {
-			for(sample = 0; sample < samples; sample++) {
-				for(channel = 0; channel < channels; channel++) {
-					a_word = signal[channel][sample];
-					*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
-					*buf_++ = (FLAC__byte)a_word;
-				}
-			}
-		}
-	}
-	else if(bytes_per_sample == 3) {
-		if(channels == 2) {
-			for(sample = 0; sample < samples; sample++) {
+			return;
+
+		case (BYTES_CHANNEL_SELECTOR (3, 2)):
+			for (sample = 0; sample < samples; sample++) {
 				a_word = signal[0][sample];
 				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
 				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
@@ -331,60 +398,90 @@ static void format_input_(FLAC__byte *buf, const FLAC__int32 * const signal[], u
 				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
 				*buf_++ = (FLAC__byte)a_word;
 			}
-		}
-		else if(channels == 1) {
-			for(sample = 0; sample < samples; sample++) {
-				a_word = signal[0][sample];
-				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
-				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
-				*buf_++ = (FLAC__byte)a_word;
+			return;
+
+		/* Four bytes per sample. */
+		case (BYTES_CHANNEL_SELECTOR (4, 1)):
+			for (sample = 0; sample < samples; sample++)
+				*buf32++ = H2LE_32(signal[0][sample]);
+			return;
+
+		case (BYTES_CHANNEL_SELECTOR (4, 2)):
+			for (sample = 0; sample < samples; sample++) {
+				*buf32++ = H2LE_32(signal[0][sample]);
+				*buf32++ = H2LE_32(signal[1][sample]);
 			}
-		}
-		else {
-			for(sample = 0; sample < samples; sample++) {
-				for(channel = 0; channel < channels; channel++) {
-					a_word = signal[channel][sample];
-					*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
-					*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
-					*buf_++ = (FLAC__byte)a_word;
-				}
+			return;
+
+		case (BYTES_CHANNEL_SELECTOR (4, 4)):
+			for (sample = 0; sample < samples; sample++) {
+				*buf32++ = H2LE_32(signal[0][sample]);
+				*buf32++ = H2LE_32(signal[1][sample]);
+				*buf32++ = H2LE_32(signal[2][sample]);
+				*buf32++ = H2LE_32(signal[3][sample]);
 			}
-		}
-	}
-	else if(bytes_per_sample == 1) {
-		if(channels == 2) {
-			for(sample = 0; sample < samples; sample++) {
-				a_word = signal[0][sample];
-				*buf_++ = (FLAC__byte)a_word;
-				a_word = signal[1][sample];
-				*buf_++ = (FLAC__byte)a_word;
+			return;
+
+		case (BYTES_CHANNEL_SELECTOR (4, 6)):
+			for (sample = 0; sample < samples; sample++) {
+				*buf32++ = H2LE_32(signal[0][sample]);
+				*buf32++ = H2LE_32(signal[1][sample]);
+				*buf32++ = H2LE_32(signal[2][sample]);
+				*buf32++ = H2LE_32(signal[3][sample]);
+				*buf32++ = H2LE_32(signal[4][sample]);
+				*buf32++ = H2LE_32(signal[5][sample]);
 			}
-		}
-		else if(channels == 1) {
-			for(sample = 0; sample < samples; sample++) {
-				a_word = signal[0][sample];
-				*buf_++ = (FLAC__byte)a_word;
+			return;
+
+		case (BYTES_CHANNEL_SELECTOR (4, 8)):
+			for (sample = 0; sample < samples; sample++) {
+				*buf32++ = H2LE_32(signal[0][sample]);
+				*buf32++ = H2LE_32(signal[1][sample]);
+				*buf32++ = H2LE_32(signal[2][sample]);
+				*buf32++ = H2LE_32(signal[3][sample]);
+				*buf32++ = H2LE_32(signal[4][sample]);
+				*buf32++ = H2LE_32(signal[5][sample]);
+				*buf32++ = H2LE_32(signal[6][sample]);
+				*buf32++ = H2LE_32(signal[7][sample]);
 			}
-		}
-		else {
-			for(sample = 0; sample < samples; sample++) {
-				for(channel = 0; channel < channels; channel++) {
+			return;
+
+		default:
+			break;
+	}
+
+	/* General version. */
+	switch (bytes_per_sample) {
+		case 1:
+			for (sample = 0; sample < samples; sample++)
+				for (channel = 0; channel < channels; channel++)
+					*buf_++ = signal[channel][sample];
+			return;
+
+		case 2:
+			for (sample = 0; sample < samples; sample++)
+				for (channel = 0; channel < channels; channel++)
+					*buf16++ = H2LE_16(signal[channel][sample]);
+			return;
+
+		case 3:
+			for (sample = 0; sample < samples; sample++)
+				for (channel = 0; channel < channels; channel++) {
 					a_word = signal[channel][sample];
+					*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+					*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
 					*buf_++ = (FLAC__byte)a_word;
 				}
-			}
-		}
-	}
-	else { /* bytes_per_sample == 4, maybe optimize more later */
-		for(sample = 0; sample < samples; sample++) {
-			for(channel = 0; channel < channels; channel++) {
-				a_word = signal[channel][sample];
-				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
-				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
-				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
-				*buf_++ = (FLAC__byte)a_word;
-			}
-		}
+			return;
+
+		case 4:
+			for (sample = 0; sample < samples; sample++)
+				for (channel = 0; channel < channels; channel++)
+					*buf32++ = H2LE_32(signal[channel][sample]);
+			return;
+
+		default:
+			break;
 	}
 }
 
@@ -396,26 +493,26 @@ FLAC__bool FLAC__MD5Accumulate(FLAC__MD5Context *ctx, const FLAC__int32 * const
 	const size_t bytes_needed = (size_t)channels * (size_t)samples * (size_t)bytes_per_sample;
 
 	/* overflow check */
-	if((size_t)channels > SIZE_MAX / (size_t)bytes_per_sample)
+	if ((size_t)channels > SIZE_MAX / (size_t)bytes_per_sample)
 		return false;
-	if((size_t)channels * (size_t)bytes_per_sample > SIZE_MAX / (size_t)samples)
+	if ((size_t)channels * (size_t)bytes_per_sample > SIZE_MAX / (size_t)samples)
 		return false;
 
-	if(ctx->capacity < bytes_needed) {
-		FLAC__byte *tmp = (FLAC__byte*) realloc(ctx->internal_buf, bytes_needed);
-		if(0 == tmp) {
-			free(ctx->internal_buf);
-			if(0 == (ctx->internal_buf = (FLAC__byte*) safe_malloc_(bytes_needed)))
+	if (ctx->capacity < bytes_needed) {
+		FLAC__byte *tmp = (FLAC__byte*) realloc(ctx->internal_buf.p8, bytes_needed);
+		if (0 == tmp) {
+			free(ctx->internal_buf.p8);
+			if (0 == (ctx->internal_buf.p8= (FLAC__byte*) safe_malloc_(bytes_needed)))
 				return false;
 		}
 		else
-			ctx->internal_buf = tmp;
+			ctx->internal_buf.p8= tmp;
 		ctx->capacity = bytes_needed;
 	}
 
-	format_input_(ctx->internal_buf, signal, channels, samples, bytes_per_sample);
+	format_input_(&ctx->internal_buf, signal, channels, samples, bytes_per_sample);
 
-	FLAC__MD5Update(ctx, ctx->internal_buf, bytes_needed);
+	FLAC__MD5Update(ctx, ctx->internal_buf.p8, bytes_needed);
 
 	return true;
 }

source/modules/juce_audio_formats/codecs/flac/libFLAC/memory.c

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2001-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -30,10 +30,14 @@
  * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
-#if HAVE_CONFIG_H
+#ifdef HAVE_CONFIG_H
 #  include <config.h>
 #endif
 
+#ifdef HAVE_STDINT_H
+#include <stdint.h>
+#endif
+
 #include "include/private/memory.h"
 #include "../assert.h"
 #include "../alloc.h"
@@ -46,25 +50,8 @@ void *FLAC__memory_alloc_aligned(size_t bytes, void **aligned_address)
 
 #ifdef FLAC__ALIGN_MALLOC_DATA
 	/* align on 32-byte (256-bit) boundary */
-	x = safe_malloc_add_2op_(bytes, /*+*/31);
-#ifdef SIZEOF_VOIDP
-#if SIZEOF_VOIDP == 4
-		/* could do  *aligned_address = x + ((unsigned) (32 - (((unsigned)x) & 31))) & 31; */
-		*aligned_address = (void*)(((unsigned)x + 31) & -32);
-#elif SIZEOF_VOIDP == 8
-		*aligned_address = (void*)(((FLAC__uint64)x + 31) & (FLAC__uint64)(-((FLAC__int64)32)));
-#else
-# error  Unsupported sizeof(void*)
-#endif
-#else
-	/* there's got to be a better way to do this right for all archs */
-	if(sizeof(void*) == sizeof(unsigned))
-		*aligned_address = (void*)(((unsigned)x + 31) & -32);
-	else if(sizeof(void*) == sizeof(FLAC__uint64))
-		*aligned_address = (void*)(((FLAC__uint64)x + 31) & (FLAC__uint64)(-((FLAC__int64)32)));
-	else
-		return 0;
-#endif
+	x = safe_malloc_add_2op_(bytes, /*+*/31L);
+	*aligned_address = (void*)(((uintptr_t)x + 31L) & -32L);
 #else
 	x = safe_malloc_(bytes);
 	*aligned_address = x;

source/modules/juce_audio_formats/codecs/flac/libFLAC/stream_decoder.c

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2000-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -30,7 +30,7 @@
  * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
-#if HAVE_CONFIG_H
+#ifdef HAVE_CONFIG_H
 #  include <config.h>
 #endif
 
@@ -70,7 +70,7 @@ FLAC_API int FLAC_API_SUPPORTS_OGG_FLAC =
  *
  ***********************************************************************/
 
-static FLAC__byte ID3V2_TAG_[3] = { 'I', 'D', '3' };
+static const FLAC__byte ID3V2_TAG_[3] = { 'I', 'D', '3' };
 
 /***********************************************************************
  *
@@ -86,7 +86,7 @@ static FLAC__bool find_metadata_(FLAC__StreamDecoder *decoder);
 static FLAC__bool read_metadata_(FLAC__StreamDecoder *decoder);
 static FLAC__bool read_metadata_streaminfo_(FLAC__StreamDecoder *decoder, FLAC__bool is_last, unsigned length);
 static FLAC__bool read_metadata_seektable_(FLAC__StreamDecoder *decoder, FLAC__bool is_last, unsigned length);
-static FLAC__bool read_metadata_vorbiscomment_(FLAC__StreamDecoder *decoder, FLAC__StreamMetadata_VorbisComment *obj);
+static FLAC__bool read_metadata_vorbiscomment_(FLAC__StreamDecoder *decoder, FLAC__StreamMetadata_VorbisComment *obj, unsigned length);
 static FLAC__bool read_metadata_cuesheet_(FLAC__StreamDecoder *decoder, FLAC__StreamMetadata_CueSheet *obj);
 static FLAC__bool read_metadata_picture_(FLAC__StreamDecoder *decoder, FLAC__StreamMetadata_Picture *obj);
 static FLAC__bool skip_id3v2_tag_(FLAC__StreamDecoder *decoder);
@@ -141,9 +141,6 @@ typedef struct FLAC__StreamDecoderPrivate {
 	void (*local_lpc_restore_signal_64bit)(const FLAC__int32 residual[], unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 data[]);
 	/* for use when the signal is <= 16 bits-per-sample, or <= 15 bits-per-sample on a side channel (which requires 1 extra bit): */
 	void (*local_lpc_restore_signal_16bit)(const FLAC__int32 residual[], unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 data[]);
-	/* for use when the signal is <= 16 bits-per-sample, or <= 15 bits-per-sample on a side channel (which requires 1 extra bit), AND order <= 8: */
-	void (*local_lpc_restore_signal_16bit_order8)(const FLAC__int32 residual[], unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 data[]);
-	FLAC__bool (*local_bitreader_read_rice_signed_block)(FLAC__BitReader *br, int vals[], unsigned nvals, unsigned parameter);
 	void *client_data;
 	FILE *file; /* only used if FLAC__stream_decoder_init_file()/FLAC__stream_decoder_init_file() called, else NULL */
 	FLAC__BitReader *input;
@@ -380,7 +377,7 @@ static FLAC__StreamDecoderInitStatus init_stream_internal_(
 #if FLAC__HAS_OGG
 	decoder->private_->is_ogg = is_ogg;
 	if(is_ogg && !FLAC__ogg_decoder_aspect_init(&decoder->protected_->ogg_decoder_aspect))
-		return decoder->protected_->state = FLAC__STREAM_DECODER_OGG_ERROR;
+		return decoder->protected_->initstate = FLAC__STREAM_DECODER_INIT_STATUS_ERROR_OPENING_FILE;
 #endif
 
 	/*
@@ -391,42 +388,44 @@ static FLAC__StreamDecoderInitStatus init_stream_internal_(
 	decoder->private_->local_lpc_restore_signal = FLAC__lpc_restore_signal;
 	decoder->private_->local_lpc_restore_signal_64bit = FLAC__lpc_restore_signal_wide;
 	decoder->private_->local_lpc_restore_signal_16bit = FLAC__lpc_restore_signal;
-	decoder->private_->local_lpc_restore_signal_16bit_order8 = FLAC__lpc_restore_signal;
-	decoder->private_->local_bitreader_read_rice_signed_block = FLAC__bitreader_read_rice_signed_block;
 	/* now override with asm where appropriate */
 #ifndef FLAC__NO_ASM
 	if(decoder->private_->cpuinfo.use_asm) {
 #ifdef FLAC__CPU_IA32
 		FLAC__ASSERT(decoder->private_->cpuinfo.type == FLAC__CPUINFO_TYPE_IA32);
 #ifdef FLAC__HAS_NASM
-#if 1 /*@@@@@@ OPT: not clearly faster, needs more testing */
-		if(decoder->private_->cpuinfo.data.ia32.bswap)
-			decoder->private_->local_bitreader_read_rice_signed_block = FLAC__bitreader_read_rice_signed_block_asm_ia32_bswap;
-#endif
-		if(decoder->private_->cpuinfo.data.ia32.mmx) {
+		decoder->private_->local_lpc_restore_signal_64bit = FLAC__lpc_restore_signal_wide_asm_ia32; /* OPT_IA32: was really necessary for GCC < 4.9 */
+		if(decoder->private_->cpuinfo.ia32.mmx) {
 			decoder->private_->local_lpc_restore_signal = FLAC__lpc_restore_signal_asm_ia32;
 			decoder->private_->local_lpc_restore_signal_16bit = FLAC__lpc_restore_signal_asm_ia32_mmx;
-			decoder->private_->local_lpc_restore_signal_16bit_order8 = FLAC__lpc_restore_signal_asm_ia32_mmx;
 		}
 		else {
 			decoder->private_->local_lpc_restore_signal = FLAC__lpc_restore_signal_asm_ia32;
 			decoder->private_->local_lpc_restore_signal_16bit = FLAC__lpc_restore_signal_asm_ia32;
-			decoder->private_->local_lpc_restore_signal_16bit_order8 = FLAC__lpc_restore_signal_asm_ia32;
 		}
 #endif
-#elif defined FLAC__CPU_PPC
-		FLAC__ASSERT(decoder->private_->cpuinfo.type == FLAC__CPUINFO_TYPE_PPC);
-		if(decoder->private_->cpuinfo.data.ppc.altivec) {
-			decoder->private_->local_lpc_restore_signal_16bit = FLAC__lpc_restore_signal_asm_ppc_altivec_16;
-			decoder->private_->local_lpc_restore_signal_16bit_order8 = FLAC__lpc_restore_signal_asm_ppc_altivec_16_order8;
+#ifdef FLAC__HAS_X86INTRIN
+# if defined FLAC__SSE2_SUPPORTED && !defined FLAC__HAS_NASM /* OPT_SSE: not better than MMX asm */
+		if(decoder->private_->cpuinfo.ia32.sse2) {
+			decoder->private_->local_lpc_restore_signal_16bit = FLAC__lpc_restore_signal_16_intrin_sse2;
+		}
+# endif
+# if defined FLAC__SSE4_1_SUPPORTED
+		if(decoder->private_->cpuinfo.ia32.sse41) {
+			decoder->private_->local_lpc_restore_signal_64bit = FLAC__lpc_restore_signal_wide_intrin_sse41;
 		}
+# endif
+#endif
+#elif defined FLAC__CPU_X86_64
+		FLAC__ASSERT(decoder->private_->cpuinfo.type == FLAC__CPUINFO_TYPE_X86_64);
+		/* No useful SSE optimizations yet */
 #endif
 	}
 #endif
 
 	/* from here on, errors are fatal */
 
-	if(!FLAC__bitreader_init(decoder->private_->input, decoder->private_->cpuinfo, read_callback_, decoder)) {
+	if(!FLAC__bitreader_init(decoder->private_->input, read_callback_, decoder)) {
 		decoder->protected_->state = FLAC__STREAM_DECODER_MEMORY_ALLOCATION_ERROR;
 		return FLAC__STREAM_DECODER_INIT_STATUS_MEMORY_ALLOCATION_ERROR;
 	}
@@ -528,10 +527,10 @@ static FLAC__StreamDecoderInitStatus init_FILE_internal_(
 	FLAC__ASSERT(0 != file);
 
 	if(decoder->protected_->state != FLAC__STREAM_DECODER_UNINITIALIZED)
-		return (FLAC__StreamDecoderInitStatus) (decoder->protected_->state = (FLAC__StreamDecoderState) FLAC__STREAM_DECODER_INIT_STATUS_ALREADY_INITIALIZED);
+		return decoder->protected_->initstate = FLAC__STREAM_DECODER_INIT_STATUS_ALREADY_INITIALIZED;
 
 	if(0 == write_callback || 0 == error_callback)
-		return (FLAC__StreamDecoderInitStatus) (decoder->protected_->state = (FLAC__StreamDecoderState) FLAC__STREAM_DECODER_INIT_STATUS_INVALID_CALLBACKS);
+		return decoder->protected_->initstate = FLAC__STREAM_DECODER_INIT_STATUS_INVALID_CALLBACKS;
 
 	/*
 	 * To make sure that our file does not go unclosed after an error, we
@@ -602,10 +601,10 @@ static FLAC__StreamDecoderInitStatus init_file_internal_(
 	 * in FLAC__stream_decoder_init_FILE() before the FILE* is assigned.
 	 */
 	if(decoder->protected_->state != FLAC__STREAM_DECODER_UNINITIALIZED)
-		return (FLAC__StreamDecoderInitStatus) (decoder->protected_->state = (FLAC__StreamDecoderState) FLAC__STREAM_DECODER_INIT_STATUS_ALREADY_INITIALIZED);
+		return decoder->protected_->initstate = FLAC__STREAM_DECODER_INIT_STATUS_ALREADY_INITIALIZED;
 
 	if(0 == write_callback || 0 == error_callback)
-		return (FLAC__StreamDecoderInitStatus) (decoder->protected_->state = (FLAC__StreamDecoderState) FLAC__STREAM_DECODER_INIT_STATUS_INVALID_CALLBACKS);
+		return decoder->protected_->initstate = FLAC__STREAM_DECODER_INIT_STATUS_INVALID_CALLBACKS;
 
 	file = filename? flac_fopen(filename, "rb") : stdin;
 
@@ -652,7 +651,7 @@ FLAC_API FLAC__bool FLAC__stream_decoder_finish(FLAC__StreamDecoder *decoder)
 	if(decoder->protected_->state == FLAC__STREAM_DECODER_UNINITIALIZED)
 		return true;
 
-	/* see the comment in FLAC__seekable_stream_decoder_reset() as to why we
+	/* see the comment in FLAC__stream_decoder_reset() as to why we
 	 * always call FLAC__MD5Final()
 	 */
 	FLAC__MD5Final(decoder->private_->computed_md5sum, &decoder->private_->md5context);
@@ -1316,9 +1315,6 @@ FLAC__bool allocate_output_(FLAC__StreamDecoder *decoder, unsigned size, unsigne
 		memset(tmp, 0, sizeof(FLAC__int32)*4);
 		decoder->private_->output[i] = tmp + 4;
 
-		/* WATCHOUT:
-		 * minimum of quadword alignment for PPC vector optimizations is REQUIRED:
-		 */
 		if(!FLAC__memory_alloc_aligned_int32_array(size, &decoder->private_->residual_unaligned[i], &decoder->private_->residual[i])) {
 			decoder->protected_->state = FLAC__STREAM_DECODER_MEMORY_ALLOCATION_ERROR;
 			return false;
@@ -1368,6 +1364,10 @@ FLAC__bool find_metadata_(FLAC__StreamDecoder *decoder)
 			id_ = 0;
 			continue;
 		}
+
+		if(id_ >= 3)
+			return false;
+
 		if(x == ID3V2_TAG_[id_]) {
 			id_++;
 			i = 0;
@@ -1446,6 +1446,7 @@ FLAC__bool read_metadata_(FLAC__StreamDecoder *decoder)
 		unsigned real_length = length;
 		FLAC__StreamMetadata block;
 
+		memset(&block, 0, sizeof(block));
 		block.is_last = is_last;
 		block.type = (FLAC__MetadataType)type;
 		block.length = length;
@@ -1470,36 +1471,37 @@ FLAC__bool read_metadata_(FLAC__StreamDecoder *decoder)
 				return false; /* read_callback_ sets the state for us */
 		}
 		else {
+			FLAC__bool ok = true;
 			switch(type) {
 				case FLAC__METADATA_TYPE_PADDING:
 					/* skip the padding bytes */
 					if(!FLAC__bitreader_skip_byte_block_aligned_no_crc(decoder->private_->input, real_length))
-						return false; /* read_callback_ sets the state for us */
+						ok = false; /* read_callback_ sets the state for us */
 					break;
 				case FLAC__METADATA_TYPE_APPLICATION:
 					/* remember, we read the ID already */
 					if(real_length > 0) {
 						if(0 == (block.data.application.data = (FLAC__byte*) malloc(real_length))) {
 							decoder->protected_->state = FLAC__STREAM_DECODER_MEMORY_ALLOCATION_ERROR;
-							return false;
+							ok = false;
 						}
-						if(!FLAC__bitreader_read_byte_block_aligned_no_crc(decoder->private_->input, block.data.application.data, real_length))
-							return false; /* read_callback_ sets the state for us */
+						else if(!FLAC__bitreader_read_byte_block_aligned_no_crc(decoder->private_->input, block.data.application.data, real_length))
+							ok = false; /* read_callback_ sets the state for us */
 					}
 					else
 						block.data.application.data = 0;
 					break;
 				case FLAC__METADATA_TYPE_VORBIS_COMMENT:
-					if(!read_metadata_vorbiscomment_(decoder, &block.data.vorbis_comment))
-						return false;
+					if(!read_metadata_vorbiscomment_(decoder, &block.data.vorbis_comment, real_length))
+						ok = false;
 					break;
 				case FLAC__METADATA_TYPE_CUESHEET:
 					if(!read_metadata_cuesheet_(decoder, &block.data.cue_sheet))
-						return false;
+						ok = false;
 					break;
 				case FLAC__METADATA_TYPE_PICTURE:
 					if(!read_metadata_picture_(decoder, &block.data.picture))
-						return false;
+						ok = false;
 					break;
 				case FLAC__METADATA_TYPE_STREAMINFO:
 				case FLAC__METADATA_TYPE_SEEKTABLE:
@@ -1509,16 +1511,16 @@ FLAC__bool read_metadata_(FLAC__StreamDecoder *decoder)
 					if(real_length > 0) {
 						if(0 == (block.data.unknown.data = (FLAC__byte*) malloc(real_length))) {
 							decoder->protected_->state = FLAC__STREAM_DECODER_MEMORY_ALLOCATION_ERROR;
-							return false;
+							ok = false;
 						}
-						if(!FLAC__bitreader_read_byte_block_aligned_no_crc(decoder->private_->input, block.data.unknown.data, real_length))
-							return false; /* read_callback_ sets the state for us */
+						else if(!FLAC__bitreader_read_byte_block_aligned_no_crc(decoder->private_->input, block.data.unknown.data, real_length))
+							ok = false; /* read_callback_ sets the state for us */
 					}
 					else
 						block.data.unknown.data = 0;
 					break;
 			}
-			if(!decoder->private_->is_seeking && decoder->private_->metadata_callback)
+			if(ok && !decoder->private_->is_seeking && decoder->private_->metadata_callback)
 				decoder->private_->metadata_callback(decoder, &block, decoder->private_->client_data);
 
 			/* now we have to free any malloc()ed data in the block */
@@ -1563,6 +1565,9 @@ FLAC__bool read_metadata_(FLAC__StreamDecoder *decoder)
 						free(block.data.unknown.data);
 					break;
 			}
+
+			if(!ok) /* anything that unsets "ok" should also make sure decoder->protected_->state is updated */
+				return false;
 		}
 	}
 
@@ -1689,58 +1694,88 @@ FLAC__bool read_metadata_seektable_(FLAC__StreamDecoder *decoder, FLAC__bool is_
 	return true;
 }
 
-FLAC__bool read_metadata_vorbiscomment_(FLAC__StreamDecoder *decoder, FLAC__StreamMetadata_VorbisComment *obj)
+FLAC__bool read_metadata_vorbiscomment_(FLAC__StreamDecoder *decoder, FLAC__StreamMetadata_VorbisComment *obj, unsigned length)
 {
 	FLAC__uint32 i;
 
 	FLAC__ASSERT(FLAC__bitreader_is_consumed_byte_aligned(decoder->private_->input));
 
 	/* read vendor string */
-	FLAC__ASSERT(FLAC__STREAM_METADATA_VORBIS_COMMENT_ENTRY_LENGTH_LEN == 32);
-	if(!FLAC__bitreader_read_uint32_little_endian(decoder->private_->input, &obj->vendor_string.length))
-		return false; /* read_callback_ sets the state for us */
-	if(obj->vendor_string.length > 0) {
-		if(0 == (obj->vendor_string.entry = (FLAC__byte*) safe_malloc_add_2op_(obj->vendor_string.length, /*+*/1))) {
-			decoder->protected_->state = FLAC__STREAM_DECODER_MEMORY_ALLOCATION_ERROR;
-			return false;
-		}
-		if(!FLAC__bitreader_read_byte_block_aligned_no_crc(decoder->private_->input, obj->vendor_string.entry, obj->vendor_string.length))
+	if (length >= 8) {
+		length -= 8; /* vendor string length + num comments entries alone take 8 bytes */
+		FLAC__ASSERT(FLAC__STREAM_METADATA_VORBIS_COMMENT_ENTRY_LENGTH_LEN == 32);
+		if (!FLAC__bitreader_read_uint32_little_endian(decoder->private_->input, &obj->vendor_string.length))
 			return false; /* read_callback_ sets the state for us */
-		obj->vendor_string.entry[obj->vendor_string.length] = '\0';
-	}
-	else
-		obj->vendor_string.entry = 0;
+		if (obj->vendor_string.length > 0) {
+			if (length < obj->vendor_string.length) {
+				obj->vendor_string.length = 0;
+				obj->vendor_string.entry = 0;
+				goto skip;
+			}
+			else
+				length -= obj->vendor_string.length;
+			if (0 == (obj->vendor_string.entry = (FLAC__byte*) safe_malloc_add_2op_(obj->vendor_string.length, /*+*/1))) {
+				decoder->protected_->state = FLAC__STREAM_DECODER_MEMORY_ALLOCATION_ERROR;
+				return false;
+			}
+			if (!FLAC__bitreader_read_byte_block_aligned_no_crc(decoder->private_->input, obj->vendor_string.entry, obj->vendor_string.length))
+				return false; /* read_callback_ sets the state for us */
+			obj->vendor_string.entry[obj->vendor_string.length] = '\0';
+		}
+		else
+			obj->vendor_string.entry = 0;
 
-	/* read num comments */
-	FLAC__ASSERT(FLAC__STREAM_METADATA_VORBIS_COMMENT_NUM_COMMENTS_LEN == 32);
-	if(!FLAC__bitreader_read_uint32_little_endian(decoder->private_->input, &obj->num_comments))
-		return false; /* read_callback_ sets the state for us */
+		/* read num comments */
+		FLAC__ASSERT(FLAC__STREAM_METADATA_VORBIS_COMMENT_NUM_COMMENTS_LEN == 32);
+		if (!FLAC__bitreader_read_uint32_little_endian(decoder->private_->input, &obj->num_comments))
+			return false; /* read_callback_ sets the state for us */
 
-	/* read comments */
-	if(obj->num_comments > 0) {
-		if(0 == (obj->comments = (FLAC__StreamMetadata_VorbisComment_Entry*) safe_malloc_mul_2op_p(obj->num_comments, /*times*/sizeof(FLAC__StreamMetadata_VorbisComment_Entry)))) {
-			decoder->protected_->state = FLAC__STREAM_DECODER_MEMORY_ALLOCATION_ERROR;
-			return false;
-		}
-		for(i = 0; i < obj->num_comments; i++) {
-			FLAC__ASSERT(FLAC__STREAM_METADATA_VORBIS_COMMENT_ENTRY_LENGTH_LEN == 32);
-			if(!FLAC__bitreader_read_uint32_little_endian(decoder->private_->input, &obj->comments[i].length))
-				return false; /* read_callback_ sets the state for us */
-			if(obj->comments[i].length > 0) {
-				if(0 == (obj->comments[i].entry = (FLAC__byte*) safe_malloc_add_2op_(obj->comments[i].length, /*+*/1))) {
-					decoder->protected_->state = FLAC__STREAM_DECODER_MEMORY_ALLOCATION_ERROR;
-					return false;
+		/* read comments */
+		if (obj->num_comments > 0) {
+			if (0 == (obj->comments = (FLAC__StreamMetadata_VorbisComment_Entry*) safe_malloc_mul_2op_p(obj->num_comments, /*times*/sizeof(FLAC__StreamMetadata_VorbisComment_Entry)))) {
+				decoder->protected_->state = FLAC__STREAM_DECODER_MEMORY_ALLOCATION_ERROR;
+				return false;
+			}
+			for (i = 0; i < obj->num_comments; i++) {
+				FLAC__ASSERT(FLAC__STREAM_METADATA_VORBIS_COMMENT_ENTRY_LENGTH_LEN == 32);
+				if (length < 4) {
+					obj->num_comments = i;
+					goto skip;
 				}
-				if(!FLAC__bitreader_read_byte_block_aligned_no_crc(decoder->private_->input, obj->comments[i].entry, obj->comments[i].length))
+				else
+					length -= 4;
+				if (!FLAC__bitreader_read_uint32_little_endian(decoder->private_->input, &obj->comments[i].length))
 					return false; /* read_callback_ sets the state for us */
-				obj->comments[i].entry[obj->comments[i].length] = '\0';
+				if (obj->comments[i].length > 0) {
+					if (length < obj->comments[i].length) {
+						obj->comments[i].length = 0;
+						obj->comments[i].entry = 0;
+						obj->num_comments = i;
+						goto skip;
+					}
+					else
+						length -= obj->comments[i].length;
+					if (0 == (obj->comments[i].entry = (FLAC__byte*) safe_malloc_add_2op_(obj->comments[i].length, /*+*/1))) {
+						decoder->protected_->state = FLAC__STREAM_DECODER_MEMORY_ALLOCATION_ERROR;
+						return false;
+					}
+					if (!FLAC__bitreader_read_byte_block_aligned_no_crc(decoder->private_->input, obj->comments[i].entry, obj->comments[i].length))
+						return false; /* read_callback_ sets the state for us */
+					obj->comments[i].entry[obj->comments[i].length] = '\0';
+				}
+				else
+					obj->comments[i].entry = 0;
 			}
-			else
-				obj->comments[i].entry = 0;
 		}
+		else
+			obj->comments = 0;
 	}
-	else {
-		obj->comments = 0;
+
+  skip:
+	if (length > 0) {
+		/* This will only happen on files with invalid data in comments */
+		if(!FLAC__bitreader_skip_byte_block_aligned_no_crc(decoder->private_->input, length))
+			return false; /* read_callback_ sets the state for us */
 	}
 
 	return true;
@@ -2655,12 +2690,8 @@ FLAC__bool read_subframe_lpc_(FLAC__StreamDecoder *decoder, unsigned channel, un
 		if( (FLAC__uint64)order * ((((FLAC__uint64)1)<<bps)-1) * ((1<<subframe->qlp_coeff_precision)-1) < (((FLAC__uint64)-1) << 32) )
 		*/
 		if(bps + subframe->qlp_coeff_precision + FLAC__bitmath_ilog2(order) <= 32)
-			if(bps <= 16 && subframe->qlp_coeff_precision <= 16) {
-				if(order <= 8)
-					decoder->private_->local_lpc_restore_signal_16bit_order8(decoder->private_->residual[channel], decoder->private_->frame.header.blocksize-order, subframe->qlp_coeff, order, subframe->quantization_level, decoder->private_->output[channel]+order);
-				else
-					decoder->private_->local_lpc_restore_signal_16bit(decoder->private_->residual[channel], decoder->private_->frame.header.blocksize-order, subframe->qlp_coeff, order, subframe->quantization_level, decoder->private_->output[channel]+order);
-			}
+			if(bps <= 16 && subframe->qlp_coeff_precision <= 16)
+				decoder->private_->local_lpc_restore_signal_16bit(decoder->private_->residual[channel], decoder->private_->frame.header.blocksize-order, subframe->qlp_coeff, order, subframe->quantization_level, decoder->private_->output[channel]+order);
 			else
 				decoder->private_->local_lpc_restore_signal(decoder->private_->residual[channel], decoder->private_->frame.header.blocksize-order, subframe->qlp_coeff, order, subframe->quantization_level, decoder->private_->output[channel]+order);
 		else
@@ -2708,14 +2739,16 @@ FLAC__bool read_residual_partitioned_rice_(FLAC__StreamDecoder *decoder, unsigne
 		if(decoder->private_->frame.header.blocksize < predictor_order) {
 			send_error_to_client_(decoder, FLAC__STREAM_DECODER_ERROR_STATUS_LOST_SYNC);
 			decoder->protected_->state = FLAC__STREAM_DECODER_SEARCH_FOR_FRAME_SYNC;
-			return true;
+			/* We have received a potentially malicious bit stream. All we can do is error out to avoid a heap overflow. */
+			return false;
 		}
 	}
 	else {
 		if(partition_samples < predictor_order) {
 			send_error_to_client_(decoder, FLAC__STREAM_DECODER_ERROR_STATUS_LOST_SYNC);
 			decoder->protected_->state = FLAC__STREAM_DECODER_SEARCH_FOR_FRAME_SYNC;
-			return true;
+			/* We have received a potentially malicious bit stream. All we can do is error out to avoid a heap overflow. */
+			return false;
 		}
 	}
 
@@ -2732,7 +2765,7 @@ FLAC__bool read_residual_partitioned_rice_(FLAC__StreamDecoder *decoder, unsigne
 		if(rice_parameter < pesc) {
 			partitioned_rice_contents->raw_bits[partition] = 0;
 			u = (partition_order == 0 || partition > 0)? partition_samples : partition_samples - predictor_order;
-			if(!decoder->private_->local_bitreader_read_rice_signed_block(decoder->private_->input, residual + sample, u, rice_parameter))
+			if(!FLAC__bitreader_read_rice_signed_block(decoder->private_->input, residual + sample, u, rice_parameter))
 				return false; /* read_callback_ sets the state for us */
 			sample += u;
 		}
@@ -3074,12 +3107,7 @@ FLAC__bool seek_to_absolute_sample_(FLAC__StreamDecoder *decoder, FLAC__uint64 s
 			return false;
 		}
 #ifndef FLAC__INTEGER_ONLY_LIBRARY
-#if defined _MSC_VER || defined __MINGW32__
-		/* with VC++ you have to spoon feed it the casting */
-		pos = (FLAC__int64)lower_bound + (FLAC__int64)((FLAC__double)(FLAC__int64)(target_sample - lower_bound_sample) / (FLAC__double)(FLAC__int64)(upper_bound_sample - lower_bound_sample) * (FLAC__double)(FLAC__int64)(upper_bound - lower_bound)) - approx_bytes_per_frame;
-#else
 		pos = (FLAC__int64)lower_bound + (FLAC__int64)((FLAC__double)(target_sample - lower_bound_sample) / (FLAC__double)(upper_bound_sample - lower_bound_sample) * (FLAC__double)(upper_bound - lower_bound)) - approx_bytes_per_frame;
-#endif
 #else
 		/* a little less accurate: */
 		if(upper_bound - lower_bound < 0xffffffff)
@@ -3203,12 +3231,7 @@ FLAC__bool seek_to_absolute_sample_ogg_(FLAC__StreamDecoder *decoder, FLAC__uint
 			}
 			else {
 #ifndef FLAC__INTEGER_ONLY_LIBRARY
-#if defined _MSC_VER || defined __MINGW32__
-				/* with MSVC you have to spoon feed it the casting */
-				pos = (FLAC__uint64)((FLAC__double)(FLAC__int64)(target_sample - left_sample) / (FLAC__double)(FLAC__int64)(right_sample - left_sample) * (FLAC__double)(FLAC__int64)(right_pos - left_pos));
-#else
 				pos = (FLAC__uint64)((FLAC__double)(target_sample - left_sample) / (FLAC__double)(right_sample - left_sample) * (FLAC__double)(right_pos - left_pos));
-#endif
 #else
 				/* a little less accurate: */
 				if ((target_sample-left_sample <= 0xffffffff) && (right_pos-left_pos <= 0xffffffff))

source/modules/juce_audio_formats/codecs/flac/libFLAC/stream_encoder.c

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2000-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -30,7 +30,7 @@
  * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
-#if HAVE_CONFIG_H
+#ifdef HAVE_CONFIG_H
 #  include <config.h>
 #endif
 
@@ -55,10 +55,10 @@
 #include "include/private/ogg_helper.h"
 #include "include/private/ogg_mapping.h"
 #endif
+#include "include/private/stream_encoder.h"
 #include "include/private/stream_encoder_framing.h"
 #include "include/private/window.h"
 #include "../alloc.h"
-#include "../compat.h"
 
 
 /* Exact Rice codeword length calculation is off by default.  The simple
@@ -103,16 +103,18 @@ static struct CompressionLevels {
 	unsigned min_residual_partition_order;
 	unsigned max_residual_partition_order;
 	unsigned rice_parameter_search_dist;
+	const char *apodization;
 } compression_levels_[] = {
-	{ false, false,  0, 0, false, false, false, 0, 3, 0 },
-	{ true , true ,  0, 0, false, false, false, 0, 3, 0 },
-	{ true , false,  0, 0, false, false, false, 0, 3, 0 },
-	{ false, false,  6, 0, false, false, false, 0, 4, 0 },
-	{ true , true ,  8, 0, false, false, false, 0, 4, 0 },
-	{ true , false,  8, 0, false, false, false, 0, 5, 0 },
-	{ true , false,  8, 0, false, false, false, 0, 6, 0 },
-	{ true , false,  8, 0, false, false, true , 0, 6, 0 },
-	{ true , false, 12, 0, false, false, true , 0, 6, 0 }
+	{ false, false,  0, 0, false, false, false, 0, 3, 0, "tukey(5e-1)" },
+	{ true , true ,  0, 0, false, false, false, 0, 3, 0, "tukey(5e-1)" },
+	{ true , false,  0, 0, false, false, false, 0, 3, 0, "tukey(5e-1)" },
+	{ false, false,  6, 0, false, false, false, 0, 4, 0, "tukey(5e-1)" },
+	{ true , true ,  8, 0, false, false, false, 0, 4, 0, "tukey(5e-1)" },
+	{ true , false,  8, 0, false, false, false, 0, 5, 0, "tukey(5e-1)" },
+	{ true , false,  8, 0, false, false, false, 0, 6, 0, "tukey(5e-1);partial_tukey(2)" },
+	{ true , false, 12, 0, false, false, false, 0, 6, 0, "tukey(5e-1);partial_tukey(2)" },
+	{ true , false, 12, 0, false, false, false, 0, 6, 0, "tukey(5e-1);partial_tukey(2);punchout_tukey(3)" }
+	/* here we use locale-independent 5e-1 instead of 0.5 or 0,5 */
 };
 
 
@@ -342,10 +344,13 @@ typedef struct FLAC__StreamEncoderPrivate {
 	unsigned current_frame_number;
 	FLAC__MD5Context md5context;
 	FLAC__CPUInfo cpuinfo;
+	void (*local_precompute_partition_info_sums)(const FLAC__int32 residual[], FLAC__uint64 abs_residual_partition_sums[], unsigned residual_samples, unsigned predictor_order, unsigned min_partition_order, unsigned max_partition_order, unsigned bps);
 #ifndef FLAC__INTEGER_ONLY_LIBRARY
 	unsigned (*local_fixed_compute_best_predictor)(const FLAC__int32 data[], unsigned data_len, FLAC__float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1]);
+	unsigned (*local_fixed_compute_best_predictor_wide)(const FLAC__int32 data[], unsigned data_len, FLAC__float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1]);
 #else
 	unsigned (*local_fixed_compute_best_predictor)(const FLAC__int32 data[], unsigned data_len, FLAC__fixedpoint residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1]);
+	unsigned (*local_fixed_compute_best_predictor_wide)(const FLAC__int32 data[], unsigned data_len, FLAC__fixedpoint residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1]);
 #endif
 #ifndef FLAC__INTEGER_ONLY_LIBRARY
 	void (*local_lpc_compute_autocorrelation)(const FLAC__real data[], unsigned data_len, unsigned lag, FLAC__real autoc[]);
@@ -873,7 +878,9 @@ static FLAC__StreamEncoderInitStatus init_stream_internal_(
 #ifndef FLAC__INTEGER_ONLY_LIBRARY
 	encoder->private_->local_lpc_compute_autocorrelation = FLAC__lpc_compute_autocorrelation;
 #endif
+	encoder->private_->local_precompute_partition_info_sums = precompute_partition_info_sums_;
 	encoder->private_->local_fixed_compute_best_predictor = FLAC__fixed_compute_best_predictor;
+	encoder->private_->local_fixed_compute_best_predictor_wide = FLAC__fixed_compute_best_predictor_wide;
 #ifndef FLAC__INTEGER_ONLY_LIBRARY
 	encoder->private_->local_lpc_compute_residual_from_qlp_coefficients = FLAC__lpc_compute_residual_from_qlp_coefficients;
 	encoder->private_->local_lpc_compute_residual_from_qlp_coefficients_64bit = FLAC__lpc_compute_residual_from_qlp_coefficients_wide;
@@ -886,21 +893,23 @@ static FLAC__StreamEncoderInitStatus init_stream_internal_(
 #  ifdef FLAC__CPU_IA32
 		FLAC__ASSERT(encoder->private_->cpuinfo.type == FLAC__CPUINFO_TYPE_IA32);
 #   ifdef FLAC__HAS_NASM
-		if(encoder->private_->cpuinfo.data.ia32.sse) {
+		if(encoder->private_->cpuinfo.ia32.sse) {
 			if(encoder->protected_->max_lpc_order < 4)
 				encoder->private_->local_lpc_compute_autocorrelation = FLAC__lpc_compute_autocorrelation_asm_ia32_sse_lag_4;
 			else if(encoder->protected_->max_lpc_order < 8)
 				encoder->private_->local_lpc_compute_autocorrelation = FLAC__lpc_compute_autocorrelation_asm_ia32_sse_lag_8;
 			else if(encoder->protected_->max_lpc_order < 12)
 				encoder->private_->local_lpc_compute_autocorrelation = FLAC__lpc_compute_autocorrelation_asm_ia32_sse_lag_12;
+			else if(encoder->protected_->max_lpc_order < 16)
+				encoder->private_->local_lpc_compute_autocorrelation = FLAC__lpc_compute_autocorrelation_asm_ia32_sse_lag_16;
 			else
 				encoder->private_->local_lpc_compute_autocorrelation = FLAC__lpc_compute_autocorrelation_asm_ia32;
 		}
-		else if(encoder->private_->cpuinfo.data.ia32._3dnow)
-			encoder->private_->local_lpc_compute_autocorrelation = FLAC__lpc_compute_autocorrelation_asm_ia32_3dnow;
 		else
 			encoder->private_->local_lpc_compute_autocorrelation = FLAC__lpc_compute_autocorrelation_asm_ia32;
-		if(encoder->private_->cpuinfo.data.ia32.mmx) {
+
+		encoder->private_->local_lpc_compute_residual_from_qlp_coefficients_64bit = FLAC__lpc_compute_residual_from_qlp_coefficients_wide_asm_ia32; /* OPT_IA32: was really necessary for GCC < 4.9 */
+		if(encoder->private_->cpuinfo.ia32.mmx) {
 			encoder->private_->local_lpc_compute_residual_from_qlp_coefficients = FLAC__lpc_compute_residual_from_qlp_coefficients_asm_ia32;
 			encoder->private_->local_lpc_compute_residual_from_qlp_coefficients_16bit = FLAC__lpc_compute_residual_from_qlp_coefficients_asm_ia32_mmx;
 		}
@@ -908,16 +917,137 @@ static FLAC__StreamEncoderInitStatus init_stream_internal_(
 			encoder->private_->local_lpc_compute_residual_from_qlp_coefficients = FLAC__lpc_compute_residual_from_qlp_coefficients_asm_ia32;
 			encoder->private_->local_lpc_compute_residual_from_qlp_coefficients_16bit = FLAC__lpc_compute_residual_from_qlp_coefficients_asm_ia32;
 		}
-		if(encoder->private_->cpuinfo.data.ia32.mmx && encoder->private_->cpuinfo.data.ia32.cmov)
+
+		if(encoder->private_->cpuinfo.ia32.mmx && encoder->private_->cpuinfo.ia32.cmov)
 			encoder->private_->local_fixed_compute_best_predictor = FLAC__fixed_compute_best_predictor_asm_ia32_mmx_cmov;
 #   endif /* FLAC__HAS_NASM */
-#  endif /* FLAC__CPU_IA32 */
+#   ifdef FLAC__HAS_X86INTRIN
+#    if defined FLAC__SSE_SUPPORTED
+		if(encoder->private_->cpuinfo.ia32.sse) {
+			if(encoder->protected_->max_lpc_order < 4)
+				encoder->private_->local_lpc_compute_autocorrelation = FLAC__lpc_compute_autocorrelation_intrin_sse_lag_4;
+			else if(encoder->protected_->max_lpc_order < 8)
+				encoder->private_->local_lpc_compute_autocorrelation = FLAC__lpc_compute_autocorrelation_intrin_sse_lag_8;
+			else if(encoder->protected_->max_lpc_order < 12)
+				encoder->private_->local_lpc_compute_autocorrelation = FLAC__lpc_compute_autocorrelation_intrin_sse_lag_12;
+			else if(encoder->protected_->max_lpc_order < 16)
+				encoder->private_->local_lpc_compute_autocorrelation = FLAC__lpc_compute_autocorrelation_intrin_sse_lag_16;
+			else
+				encoder->private_->local_lpc_compute_autocorrelation = FLAC__lpc_compute_autocorrelation;
+		}
+#    endif
+
+#    ifdef FLAC__SSE2_SUPPORTED
+		if(encoder->private_->cpuinfo.ia32.sse2) {
+			encoder->private_->local_lpc_compute_residual_from_qlp_coefficients       = FLAC__lpc_compute_residual_from_qlp_coefficients_intrin_sse2;
+			encoder->private_->local_lpc_compute_residual_from_qlp_coefficients_16bit = FLAC__lpc_compute_residual_from_qlp_coefficients_16_intrin_sse2;
+		}
+#    endif
+#    ifdef FLAC__SSE4_1_SUPPORTED
+		if(encoder->private_->cpuinfo.ia32.sse41) {
+			encoder->private_->local_lpc_compute_residual_from_qlp_coefficients       = FLAC__lpc_compute_residual_from_qlp_coefficients_intrin_sse41;
+			encoder->private_->local_lpc_compute_residual_from_qlp_coefficients_64bit = FLAC__lpc_compute_residual_from_qlp_coefficients_wide_intrin_sse41;
+		}
+#    endif
+#    ifdef FLAC__AVX2_SUPPORTED
+		if(encoder->private_->cpuinfo.ia32.avx2) {
+			encoder->private_->local_lpc_compute_residual_from_qlp_coefficients_16bit = FLAC__lpc_compute_residual_from_qlp_coefficients_16_intrin_avx2;
+			encoder->private_->local_lpc_compute_residual_from_qlp_coefficients       = FLAC__lpc_compute_residual_from_qlp_coefficients_intrin_avx2;
+			encoder->private_->local_lpc_compute_residual_from_qlp_coefficients_64bit = FLAC__lpc_compute_residual_from_qlp_coefficients_wide_intrin_avx2;
+		}
+#    endif
+
+#    ifdef FLAC__SSE2_SUPPORTED
+		if (encoder->private_->cpuinfo.ia32.sse2) {
+			encoder->private_->local_fixed_compute_best_predictor      = FLAC__fixed_compute_best_predictor_intrin_sse2;
+			encoder->private_->local_fixed_compute_best_predictor_wide = FLAC__fixed_compute_best_predictor_wide_intrin_sse2;
+		}
+#    endif
+#    ifdef FLAC__SSSE3_SUPPORTED
+		if (encoder->private_->cpuinfo.ia32.ssse3) {
+			encoder->private_->local_fixed_compute_best_predictor      = FLAC__fixed_compute_best_predictor_intrin_ssse3;
+			encoder->private_->local_fixed_compute_best_predictor_wide = FLAC__fixed_compute_best_predictor_wide_intrin_ssse3;
+		}
+#    endif
+#   endif /* FLAC__HAS_X86INTRIN */
+#  elif defined FLAC__CPU_X86_64
+		FLAC__ASSERT(encoder->private_->cpuinfo.type == FLAC__CPUINFO_TYPE_X86_64);
+#   ifdef FLAC__HAS_X86INTRIN
+#    ifdef FLAC__SSE_SUPPORTED
+		if(encoder->protected_->max_lpc_order < 4)
+			encoder->private_->local_lpc_compute_autocorrelation = FLAC__lpc_compute_autocorrelation_intrin_sse_lag_4;
+		else if(encoder->protected_->max_lpc_order < 8)
+			encoder->private_->local_lpc_compute_autocorrelation = FLAC__lpc_compute_autocorrelation_intrin_sse_lag_8;
+		else if(encoder->protected_->max_lpc_order < 12)
+			encoder->private_->local_lpc_compute_autocorrelation = FLAC__lpc_compute_autocorrelation_intrin_sse_lag_12;
+		else if(encoder->protected_->max_lpc_order < 16)
+			encoder->private_->local_lpc_compute_autocorrelation = FLAC__lpc_compute_autocorrelation_intrin_sse_lag_16;
+#    endif
+
+#    ifdef FLAC__SSE2_SUPPORTED
+		encoder->private_->local_lpc_compute_residual_from_qlp_coefficients_16bit = FLAC__lpc_compute_residual_from_qlp_coefficients_16_intrin_sse2;
+#    endif
+#    ifdef FLAC__SSE4_1_SUPPORTED
+		if(encoder->private_->cpuinfo.x86.sse41) {
+			encoder->private_->local_lpc_compute_residual_from_qlp_coefficients = FLAC__lpc_compute_residual_from_qlp_coefficients_intrin_sse41;
+		}
+#    endif
+#    ifdef FLAC__AVX2_SUPPORTED
+		if(encoder->private_->cpuinfo.x86.avx2) {
+			encoder->private_->local_lpc_compute_residual_from_qlp_coefficients_16bit = FLAC__lpc_compute_residual_from_qlp_coefficients_16_intrin_avx2;
+			encoder->private_->local_lpc_compute_residual_from_qlp_coefficients       = FLAC__lpc_compute_residual_from_qlp_coefficients_intrin_avx2;
+			encoder->private_->local_lpc_compute_residual_from_qlp_coefficients_64bit = FLAC__lpc_compute_residual_from_qlp_coefficients_wide_intrin_avx2;
+		}
+#    endif
+
+#    ifdef FLAC__SSE2_SUPPORTED
+		encoder->private_->local_fixed_compute_best_predictor      = FLAC__fixed_compute_best_predictor_intrin_sse2;
+		encoder->private_->local_fixed_compute_best_predictor_wide = FLAC__fixed_compute_best_predictor_wide_intrin_sse2;
+#    endif
+#    ifdef FLAC__SSSE3_SUPPORTED
+		if (encoder->private_->cpuinfo.x86.ssse3) {
+			encoder->private_->local_fixed_compute_best_predictor      = FLAC__fixed_compute_best_predictor_intrin_ssse3;
+			encoder->private_->local_fixed_compute_best_predictor_wide = FLAC__fixed_compute_best_predictor_wide_intrin_ssse3;
+		}
+#    endif
+#   endif /* FLAC__HAS_X86INTRIN */
+#  endif /* FLAC__CPU_... */
 	}
 # endif /* !FLAC__NO_ASM */
 #endif /* !FLAC__INTEGER_ONLY_LIBRARY */
+#if !defined FLAC__NO_ASM && defined FLAC__HAS_X86INTRIN
+	if(encoder->private_->cpuinfo.use_asm) {
+# if defined FLAC__CPU_IA32
+#  ifdef FLAC__SSE2_SUPPORTED
+		if(encoder->private_->cpuinfo.ia32.sse2)
+			encoder->private_->local_precompute_partition_info_sums = FLAC__precompute_partition_info_sums_intrin_sse2;
+#  endif
+#  ifdef FLAC__SSSE3_SUPPORTED
+		if(encoder->private_->cpuinfo.ia32.ssse3)
+			encoder->private_->local_precompute_partition_info_sums = FLAC__precompute_partition_info_sums_intrin_ssse3;
+#  endif
+#  ifdef FLAC__AVX2_SUPPORTED
+		if(encoder->private_->cpuinfo.ia32.avx2)
+			encoder->private_->local_precompute_partition_info_sums = FLAC__precompute_partition_info_sums_intrin_avx2;
+#  endif
+# elif defined FLAC__CPU_X86_64
+#  ifdef FLAC__SSE2_SUPPORTED
+		encoder->private_->local_precompute_partition_info_sums = FLAC__precompute_partition_info_sums_intrin_sse2;
+#  endif
+#  ifdef FLAC__SSSE3_SUPPORTED
+		if(encoder->private_->cpuinfo.x86.ssse3)
+			encoder->private_->local_precompute_partition_info_sums = FLAC__precompute_partition_info_sums_intrin_ssse3;
+#  endif
+#  ifdef FLAC__AVX2_SUPPORTED
+		if(encoder->private_->cpuinfo.x86.avx2)
+			encoder->private_->local_precompute_partition_info_sums = FLAC__precompute_partition_info_sums_intrin_avx2;
+#  endif
+# endif /* FLAC__CPU_... */
+	}
+#endif /* !FLAC__NO_ASM && FLAC__HAS_X86INTRIN */
 	/* finally override based on wide-ness if necessary */
 	if(encoder->private_->use_wide_by_block) {
-		encoder->private_->local_fixed_compute_best_predictor = FLAC__fixed_compute_best_predictor_wide;
+		encoder->private_->local_fixed_compute_best_predictor = encoder->private_->local_fixed_compute_best_predictor_wide;
 	}
 
 	/* set state to OK; from here on, errors are fatal and we'll override the state then */
@@ -1152,7 +1282,7 @@ static FLAC__StreamEncoderInitStatus init_FILE_internal_(
 	FLAC__StreamEncoder *encoder,
 	FILE *file,
 	FLAC__StreamEncoderProgressCallback progress_callback,
-	void * /*client_data*/,
+	void *client_data,
 	FLAC__bool is_ogg
 )
 {
@@ -1178,6 +1308,13 @@ static FLAC__StreamEncoderInitStatus init_FILE_internal_(
 	if(file == stdout)
 		file = get_binary_stdout_(); /* just to be safe */
 
+#ifdef _WIN32
+	/*
+	 * Windows can suffer quite badly from disk fragmentation. This can be
+	 * reduced significantly by setting the output buffer size to be 10MB.
+	 */
+	setvbuf(file, NULL, _IOFBF, 10*1024*1024);
+#endif
 	encoder->private_->file = file;
 
 	encoder->private_->progress_callback = progress_callback;
@@ -1447,11 +1584,10 @@ FLAC_API FLAC__bool FLAC__stream_encoder_set_compression_level(FLAC__StreamEncod
 	ok &= FLAC__stream_encoder_set_do_mid_side_stereo          (encoder, compression_levels_[value].do_mid_side_stereo);
 	ok &= FLAC__stream_encoder_set_loose_mid_side_stereo       (encoder, compression_levels_[value].loose_mid_side_stereo);
 #ifndef FLAC__INTEGER_ONLY_LIBRARY
-#if 0
-	/* was: */
+#if 1
 	ok &= FLAC__stream_encoder_set_apodization                 (encoder, compression_levels_[value].apodization);
-	/* but it's too hard to specify the string in a locale-specific way */
 #else
+	/* equivalent to -A tukey(0.5) */
 	encoder->protected_->num_apodizations = 1;
 	encoder->protected_->apodizations[0].type = FLAC__APODIZATION_TUKEY;
 	encoder->protected_->apodizations[0].parameters.tukey.p = 0.5;
@@ -1555,6 +1691,48 @@ FLAC_API FLAC__bool FLAC__stream_encoder_set_apodization(FLAC__StreamEncoder *en
 				encoder->protected_->apodizations[encoder->protected_->num_apodizations++].type = FLAC__APODIZATION_TUKEY;
 			}
 		}
+		else if(n>15   && 0 == strncmp("partial_tukey("       , specification, 14)) {
+			FLAC__int32 tukey_parts = (FLAC__int32)strtod(specification+14, 0);
+			const char *si_1 = strchr(specification, '/');
+			FLAC__real overlap = si_1?flac_min((FLAC__real)strtod(si_1+1, 0),0.99f):0.1f;
+			FLAC__real overlap_units = 1.0f/(1.0f - overlap) - 1.0f;
+			const char *si_2 = strchr((si_1?(si_1+1):specification), '/');
+			FLAC__real tukey_p = si_2?(FLAC__real)strtod(si_2+1, 0):0.2f;
+
+			if (tukey_parts <= 1) {
+				encoder->protected_->apodizations[encoder->protected_->num_apodizations].parameters.tukey.p = tukey_p;
+				encoder->protected_->apodizations[encoder->protected_->num_apodizations++].type = FLAC__APODIZATION_TUKEY;
+			}else if (encoder->protected_->num_apodizations + tukey_parts < 32){
+				FLAC__int32 m;
+				for(m = 0; m < tukey_parts; m++){
+					encoder->protected_->apodizations[encoder->protected_->num_apodizations].parameters.multiple_tukey.p = tukey_p;
+					encoder->protected_->apodizations[encoder->protected_->num_apodizations].parameters.multiple_tukey.start = m/(tukey_parts+overlap_units);
+					encoder->protected_->apodizations[encoder->protected_->num_apodizations].parameters.multiple_tukey.end = (m+1+overlap_units)/(tukey_parts+overlap_units);
+					encoder->protected_->apodizations[encoder->protected_->num_apodizations++].type = FLAC__APODIZATION_PARTIAL_TUKEY;
+				}
+			}
+		}
+		else if(n>16   && 0 == strncmp("punchout_tukey("       , specification, 15)) {
+			FLAC__int32 tukey_parts = (FLAC__int32)strtod(specification+15, 0);
+			const char *si_1 = strchr(specification, '/');
+			FLAC__real overlap = si_1?flac_min((FLAC__real)strtod(si_1+1, 0),0.99f):0.2f;
+			FLAC__real overlap_units = 1.0f/(1.0f - overlap) - 1.0f;
+			const char *si_2 = strchr((si_1?(si_1+1):specification), '/');
+			FLAC__real tukey_p = si_2?(FLAC__real)strtod(si_2+1, 0):0.2f;
+
+			if (tukey_parts <= 1) {
+				encoder->protected_->apodizations[encoder->protected_->num_apodizations].parameters.tukey.p = tukey_p;
+				encoder->protected_->apodizations[encoder->protected_->num_apodizations++].type = FLAC__APODIZATION_TUKEY;
+			}else if (encoder->protected_->num_apodizations + tukey_parts < 32){
+				FLAC__int32 m;
+				for(m = 0; m < tukey_parts; m++){
+					encoder->protected_->apodizations[encoder->protected_->num_apodizations].parameters.multiple_tukey.p = tukey_p;
+					encoder->protected_->apodizations[encoder->protected_->num_apodizations].parameters.multiple_tukey.start = m/(tukey_parts+overlap_units);
+					encoder->protected_->apodizations[encoder->protected_->num_apodizations].parameters.multiple_tukey.end = (m+1+overlap_units)/(tukey_parts+overlap_units);
+					encoder->protected_->apodizations[encoder->protected_->num_apodizations++].type = FLAC__APODIZATION_PUNCHOUT_TUKEY;
+				}
+			}
+		}
 		else if(n==5  && 0 == strncmp("welch"        , specification, n))
 			encoder->protected_->apodizations[encoder->protected_->num_apodizations++].type = FLAC__APODIZATION_WELCH;
 		if (encoder->protected_->num_apodizations == 32)
@@ -2236,8 +2414,8 @@ FLAC__bool resize_buffers_(FLAC__StreamEncoder *encoder, unsigned new_blocksize)
 
 	ok = true;
 
-	/* WATCHOUT: FLAC__lpc_compute_residual_from_qlp_coefficients_asm_ia32_mmx()
-	 * requires that the input arrays (in our case the integer signals)
+	/* WATCHOUT: FLAC__lpc_compute_residual_from_qlp_coefficients_asm_ia32_mmx() and ..._intrin_sse2()
+	 * require that the input arrays (in our case the integer signals)
 	 * have a buffer of up to 3 zeroes in front (at negative indices) for
 	 * alignment purposes; we use 4 in front to keep the data well-aligned.
 	 */
@@ -2334,6 +2512,12 @@ FLAC__bool resize_buffers_(FLAC__StreamEncoder *encoder, unsigned new_blocksize)
 				case FLAC__APODIZATION_TUKEY:
 					FLAC__window_tukey(encoder->private_->window[i], new_blocksize, encoder->protected_->apodizations[i].parameters.tukey.p);
 					break;
+				case FLAC__APODIZATION_PARTIAL_TUKEY:
+					FLAC__window_partial_tukey(encoder->private_->window[i], new_blocksize, encoder->protected_->apodizations[i].parameters.multiple_tukey.p, encoder->protected_->apodizations[i].parameters.multiple_tukey.start, encoder->protected_->apodizations[i].parameters.multiple_tukey.end);
+					break;
+				case FLAC__APODIZATION_PUNCHOUT_TUKEY:
+					FLAC__window_punchout_tukey(encoder->private_->window[i], new_blocksize, encoder->protected_->apodizations[i].parameters.multiple_tukey.p, encoder->protected_->apodizations[i].parameters.multiple_tukey.start, encoder->protected_->apodizations[i].parameters.multiple_tukey.end);
+					break;
 				case FLAC__APODIZATION_WELCH:
 					FLAC__window_welch(encoder->private_->window[i], new_blocksize);
 					break;
@@ -3165,7 +3349,7 @@ FLAC__bool process_subframe_(
 #endif
 #ifndef FLAC__INTEGER_ONLY_LIBRARY
 	FLAC__double lpc_residual_bits_per_sample;
-	FLAC__real autoc[FLAC__MAX_LPC_ORDER+1]; /* WATCHOUT: the size is important even though encoder->protected_->max_lpc_order might be less; some asm routines need all the space */
+	FLAC__real autoc[FLAC__MAX_LPC_ORDER+1]; /* WATCHOUT: the size is important even though encoder->protected_->max_lpc_order might be less; some asm and x86 intrinsic routines need all the space */
 	FLAC__double lpc_error[FLAC__MAX_LPC_ORDER];
 	unsigned min_lpc_order, max_lpc_order, lpc_order;
 	unsigned min_qlp_coeff_precision, max_qlp_coeff_precision, qlp_coeff_precision;
@@ -3318,9 +3502,9 @@ FLAC__bool process_subframe_(
 								}
 								if(encoder->protected_->do_qlp_coeff_prec_search) {
 									min_qlp_coeff_precision = FLAC__MIN_QLP_COEFF_PRECISION;
-									/* try to ensure a 32-bit datapath throughout for 16bps(+1bps for side channel) or less */
-									if(subframe_bps <= 17) {
-										max_qlp_coeff_precision = flac_min(32 - subframe_bps - lpc_order, FLAC__MAX_QLP_COEFF_PRECISION);
+									/* try to keep qlp coeff precision such that only 32-bit math is required for decode of <=16bps streams */
+									if(subframe_bps <= 16) {
+										max_qlp_coeff_precision = flac_min(32 - subframe_bps - FLAC__bitmath_ilog2(lpc_order), FLAC__MAX_QLP_COEFF_PRECISION);
 										max_qlp_coeff_precision = flac_max(max_qlp_coeff_precision, min_qlp_coeff_precision);
 									}
 									else
@@ -3556,7 +3740,7 @@ unsigned evaluate_lpc_subframe_(
 	FLAC__EntropyCodingMethod_PartitionedRiceContents *partitioned_rice_contents
 )
 {
-	FLAC__int32 qlp_coeff[FLAC__MAX_LPC_ORDER];
+	FLAC__int32 qlp_coeff[FLAC__MAX_LPC_ORDER]; /* WATCHOUT: the size is important; some x86 intrinsic routines need more than lpc order elements */
 	unsigned i, residual_bits, estimate;
 	int quantization, ret;
 	const unsigned residual_samples = blocksize - order;
@@ -3671,7 +3855,7 @@ unsigned find_best_partition_order_(
 	max_partition_order = FLAC__format_get_max_rice_partition_order_from_blocksize_limited_max_and_predictor_order(max_partition_order, blocksize, predictor_order);
 	min_partition_order = flac_min(min_partition_order, max_partition_order);
 
-	precompute_partition_info_sums_(residual, abs_residual_partition_sums, residual_samples, predictor_order, min_partition_order, max_partition_order, bps);
+	private_->local_precompute_partition_info_sums(residual, abs_residual_partition_sums, residual_samples, predictor_order, min_partition_order, max_partition_order, bps);
 
 	if(do_escape_coding)
 		precompute_partition_info_escapes_(residual, raw_bits_per_partition, residual_samples, predictor_order, min_partition_order, max_partition_order);
@@ -3743,17 +3927,6 @@ unsigned find_best_partition_order_(
 	return best_residual_bits;
 }
 
-#if defined(FLAC__CPU_IA32) && !defined FLAC__NO_ASM && defined FLAC__HAS_NASM
-extern void precompute_partition_info_sums_32bit_asm_ia32_(
-	const FLAC__int32 residual[],
-	FLAC__uint64 abs_residual_partition_sums[],
-	unsigned blocksize,
-	unsigned predictor_order,
-	unsigned min_partition_order,
-	unsigned max_partition_order
-);
-#endif
-
 void precompute_partition_info_sums_(
 	const FLAC__int32 residual[],
 	FLAC__uint64 abs_residual_partition_sums[],
@@ -3769,21 +3942,12 @@ void precompute_partition_info_sums_(
 
 	FLAC__ASSERT(default_partition_samples > predictor_order);
 
-#if defined(FLAC__CPU_IA32) && !defined FLAC__NO_ASM && defined FLAC__HAS_NASM
-	/* slightly pessimistic but still catches all common cases */
-	/* WATCHOUT: "+ bps" is an assumption that the average residual magnitude will not be more than "bps" bits */
-	if(FLAC__bitmath_ilog2(default_partition_samples) + bps < 32) {
-		precompute_partition_info_sums_32bit_asm_ia32_(residual, abs_residual_partition_sums, residual_samples + predictor_order, predictor_order, min_partition_order, max_partition_order);
-		return;
-	}
-#endif
-
 	/* first do max_partition_order */
 	{
 		unsigned partition, residual_sample, end = (unsigned)(-(int)predictor_order);
-		/* slightly pessimistic but still catches all common cases */
-		/* WATCHOUT: "+ bps" is an assumption that the average residual magnitude will not be more than "bps" bits */
-		if(FLAC__bitmath_ilog2(default_partition_samples) + bps < 32) {
+		/* WATCHOUT: "+ bps + FLAC__MAX_EXTRA_RESIDUAL_BPS" is the maximum
+		 * assumed size of the average residual magnitude */
+		if(FLAC__bitmath_ilog2(default_partition_samples) + bps + FLAC__MAX_EXTRA_RESIDUAL_BPS < 32) {
 			FLAC__uint32 abs_residual_partition_sum;
 
 			for(partition = residual_sample = 0; partition < partitions; partition++) {
@@ -4027,8 +4191,35 @@ FLAC__bool set_partitioned_rice_(
 			 * in the partition, so the actual mean is
 			 * mean/partition_samples
 			 */
+#if 0 /* old simple code */
 			for(rice_parameter = 0, k = partition_samples; k < mean; rice_parameter++, k <<= 1)
 				;
+#else
+#if defined FLAC__CPU_X86_64 /* and other 64-bit arch, too */
+			if(mean <= 0x80000000/512) { /* 512: more or less optimal for both 16- and 24-bit input */
+#else
+			if(mean <= 0x80000000/8) { /* 32-bit arch: use 32-bit math if possible */
+#endif
+				FLAC__uint32 k2, mean2 = (FLAC__uint32) mean;
+				rice_parameter = 0; k2 = partition_samples;
+				while(k2*8 < mean2) { /* requires: mean <= (2^31)/8 */
+					rice_parameter += 4; k2 <<= 4; /* tuned for 16-bit input */
+				}
+				while(k2 < mean2) { /* requires: mean <= 2^31 */
+					rice_parameter++; k2 <<= 1;
+				}
+			}
+			else {
+				rice_parameter = 0; k = partition_samples;
+				if(mean <= FLAC__U64L(0x8000000000000000)/128) /* usually mean is _much_ smaller than this value */
+					while(k*128 < mean) { /* requires: mean <= (2^63)/128 */
+						rice_parameter += 8; k <<= 8; /* tuned for 24-bit input */
+					}
+				while(k < mean) { /* requires: mean <= 2^63 */
+					rice_parameter++; k <<= 1;
+				}
+			}
+#endif
 			if(rice_parameter >= rice_parameter_limit) {
 #ifdef DEBUG_VERBOSE
 				fprintf(stderr, "clipping rice_parameter (%u -> %u) @6\n", rice_parameter, rice_parameter_limit - 1);

source/modules/juce_audio_formats/codecs/flac/libFLAC/stream_encoder_framing.c

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2000-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -30,7 +30,7 @@
  * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
-#if HAVE_CONFIG_H
+#ifdef HAVE_CONFIG_H
 #  include <config.h>
 #endif
 

source/modules/juce_audio_formats/codecs/flac/libFLAC/window_flac.c

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2006-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -30,7 +30,7 @@
  * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
-#if HAVE_CONFIG_H
+#ifdef HAVE_CONFIG_H
 #  include <config.h>
 #endif
 
@@ -41,11 +41,6 @@
 
 #ifndef FLAC__INTEGER_ONLY_LIBRARY
 
-#ifndef M_PI
-/* math.h in VC++ doesn't seem to have this (how Microsoft is that?) */
-#define M_PI 3.14159265358979323846
-#endif
-
 
 void FLAC__window_bartlett(FLAC__real *window, const FLAC__int32 L)
 {
@@ -62,7 +57,7 @@ void FLAC__window_bartlett(FLAC__real *window, const FLAC__int32 L)
 		for (n = 0; n <= L/2-1; n++)
 			window[n] = 2.0f * n / (float)N;
 		for (; n <= N; n++)
-			window[n] = 2.0f - 2.0f * (N-n) / (float)N;
+			window[n] = 2.0f - 2.0f * n / (float)N;
 	}
 }
 
@@ -72,7 +67,7 @@ void FLAC__window_bartlett_hann(FLAC__real *window, const FLAC__int32 L)
 	FLAC__int32 n;
 
 	for (n = 0; n < L; n++)
-		window[n] = (FLAC__real)(0.62f - 0.48f * fabs((float)n/(float)N+0.5f) + 0.38f * cos(2.0f * M_PI * ((float)n/(float)N+0.5f)));
+		window[n] = (FLAC__real)(0.62f - 0.48f * fabs((float)n/(float)N-0.5f) - 0.38f * cos(2.0f * M_PI * ((float)n/(float)N)));
 }
 
 void FLAC__window_blackman(FLAC__real *window, const FLAC__int32 L)
@@ -177,16 +172,16 @@ void FLAC__window_triangle(FLAC__real *window, const FLAC__int32 L)
 	FLAC__int32 n;
 
 	if (L & 1) {
-		for (n = 1; n <= L+1/2; n++)
+		for (n = 1; n <= (L+1)/2; n++)
 			window[n-1] = 2.0f * n / ((float)L + 1.0f);
 		for (; n <= L; n++)
-			window[n-1] = - (float)(2 * (L - n + 1)) / ((float)L + 1.0f);
+			window[n-1] = (float)(2 * (L - n + 1)) / ((float)L + 1.0f);
 	}
 	else {
 		for (n = 1; n <= L/2; n++)
-			window[n-1] = 2.0f * n / (float)L;
+			window[n-1] = 2.0f * n / ((float)L + 1.0f);
 		for (; n <= L; n++)
-			window[n-1] = ((float)(2 * (L - n)) + 1.0f) / (float)L;
+			window[n-1] = (float)(2 * (L - n + 1)) / ((float)L + 1.0f);
 	}
 }
 
@@ -211,6 +206,66 @@ void FLAC__window_tukey(FLAC__real *window, const FLAC__int32 L, const FLAC__rea
 	}
 }
 
+void FLAC__window_partial_tukey(FLAC__real *window, const FLAC__int32 L, const FLAC__real p, const FLAC__real start, const FLAC__real end)
+{
+	const FLAC__int32 start_n = (FLAC__int32)(start * L);
+	const FLAC__int32 end_n = (FLAC__int32)(end * L);
+	const FLAC__int32 N = end_n - start_n;
+	FLAC__int32 Np, n, i;
+
+	if (p <= 0.0f)
+		FLAC__window_partial_tukey(window, L, 0.05f, start, end);
+	else if (p >= 1.0f)
+		FLAC__window_partial_tukey(window, L, 0.95f, start, end);
+	else {
+
+		Np = (FLAC__int32)(p / 2.0f * N);
+
+		for (n = 0; n < start_n && n < L; n++)
+			window[n] = 0.0f;
+		for (i = 1; n < (start_n+Np) && n < L; n++, i++)
+			window[n] = (FLAC__real)(0.5f - 0.5f * cos(M_PI * i / Np));
+		for (; n < (end_n-Np) && n < L; n++)
+			window[n] = 1.0f;
+		for (i = Np; n < end_n && n < L; n++, i--)
+			window[n] = (FLAC__real)(0.5f - 0.5f * cos(M_PI * i / Np));
+		for (; n < L; n++)
+			window[n] = 0.0f;
+	}
+}
+
+void FLAC__window_punchout_tukey(FLAC__real *window, const FLAC__int32 L, const FLAC__real p, const FLAC__real start, const FLAC__real end)
+{
+	const FLAC__int32 start_n = (FLAC__int32)(start * L);
+	const FLAC__int32 end_n = (FLAC__int32)(end * L);
+	FLAC__int32 Ns, Ne, n, i;
+
+	if (p <= 0.0f)
+		FLAC__window_punchout_tukey(window, L, 0.05f, start, end);
+	else if (p >= 1.0f)
+		FLAC__window_punchout_tukey(window, L, 0.95f, start, end);
+	else {
+
+		Ns = (FLAC__int32)(p / 2.0f * start_n);
+		Ne = (FLAC__int32)(p / 2.0f * (L - end_n));
+
+		for (n = 0, i = 1; n < Ns && n < L; n++, i++)
+			window[n] = (FLAC__real)(0.5f - 0.5f * cos(M_PI * i / Ns));
+		for (; n < start_n-Ns && n < L; n++)
+			window[n] = 1.0f;
+		for (i = Ns; n < start_n && n < L; n++, i--)
+			window[n] = (FLAC__real)(0.5f - 0.5f * cos(M_PI * i / Ns));
+		for (; n < end_n && n < L; n++)
+			window[n] = 0.0f;
+		for (i = 1; n < end_n+Ne && n < L; n++, i++)
+			window[n] = (FLAC__real)(0.5f - 0.5f * cos(M_PI * i / Ne));
+		for (; n < L - (Ne) && n < L; n++)
+			window[n] = 1.0f;
+		for (i = Ne; n < L; n++, i--)
+			window[n] = (FLAC__real)(0.5f - 0.5f * cos(M_PI * i / Ne));
+	}
+}
+
 void FLAC__window_welch(FLAC__real *window, const FLAC__int32 L)
 {
 	const FLAC__int32 N = L - 1;

source/modules/juce_audio_formats/codecs/flac/metadata.h

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2001-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -500,7 +500,7 @@ FLAC_API FLAC__MetadataType FLAC__metadata_simple_iterator_get_block_type(const
  * \retval unsigned
  *    The length of the metadata block at the current iterator position.
  *    The is same length as that in the
- *    <a href="http://flac.sourceforge.net/format.html#metadata_block_header">metadata block header</a>,
+ *    <a href="http://xiph.org/flac/format.html#metadata_block_header">metadata block header</a>,
  *    i.e. the length of the metadata body that follows the header.
  */
 FLAC_API unsigned FLAC__metadata_simple_iterator_get_block_length(const FLAC__Metadata_SimpleIterator *iterator);

source/modules/juce_audio_formats/codecs/flac/ordinals.h

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2000-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

source/modules/juce_audio_formats/codecs/flac/stream_decoder.h

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2000-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

source/modules/juce_audio_formats/codecs/flac/stream_encoder.h

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2000-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -830,28 +830,28 @@ FLAC_API FLAC__bool FLAC__stream_encoder_set_sample_rate(FLAC__StreamEncoder *en
  * The actual values set for each level are:
  * <table>
  * <tr>
- *  <td><b>level</b><td>
- *  <td>do mid-side stereo<td>
- *  <td>loose mid-side stereo<td>
- *  <td>apodization<td>
- *  <td>max lpc order<td>
- *  <td>qlp coeff precision<td>
- *  <td>qlp coeff prec search<td>
- *  <td>escape coding<td>
- *  <td>exhaustive model search<td>
- *  <td>min residual partition order<td>
- *  <td>max residual partition order<td>
- *  <td>rice parameter search dist<td>
+ *  <td><b>level</b></td>
+ *  <td>do mid-side stereo</td>
+ *  <td>loose mid-side stereo</td>
+ *  <td>apodization</td>
+ *  <td>max lpc order</td>
+ *  <td>qlp coeff precision</td>
+ *  <td>qlp coeff prec search</td>
+ *  <td>escape coding</td>
+ *  <td>exhaustive model search</td>
+ *  <td>min residual partition order</td>
+ *  <td>max residual partition order</td>
+ *  <td>rice parameter search dist</td>
  * </tr>
- * <tr>  <td><b>0</b><td>  <td>false<td>  <td>false<td>  <td>tukey(0.5)<td>  <td>0<td>   <td>0<td>  <td>false<td>  <td>false<td>  <td>false<td>  <td>0<td>  <td>3<td>  <td>0<td>  </tr>
- * <tr>  <td><b>1</b><td>  <td>true<td>   <td>true<td>   <td>tukey(0.5)<td>  <td>0<td>   <td>0<td>  <td>false<td>  <td>false<td>  <td>false<td>  <td>0<td>  <td>3<td>  <td>0<td>  </tr>
- * <tr>  <td><b>2</b><td>  <td>true<td>   <td>false<td>  <td>tukey(0.5)<td>  <td>0<td>   <td>0<td>  <td>false<td>  <td>false<td>  <td>false<td>  <td>0<td>  <td>3<td>  <td>0<td>  </tr>
- * <tr>  <td><b>3</b><td>  <td>false<td>  <td>false<td>  <td>tukey(0.5)<td>  <td>6<td>   <td>0<td>  <td>false<td>  <td>false<td>  <td>false<td>  <td>0<td>  <td>4<td>  <td>0<td>  </tr>
- * <tr>  <td><b>4</b><td>  <td>true<td>   <td>true<td>   <td>tukey(0.5)<td>  <td>8<td>   <td>0<td>  <td>false<td>  <td>false<td>  <td>false<td>  <td>0<td>  <td>4<td>  <td>0<td>  </tr>
- * <tr>  <td><b>5</b><td>  <td>true<td>   <td>false<td>  <td>tukey(0.5)<td>  <td>8<td>   <td>0<td>  <td>false<td>  <td>false<td>  <td>false<td>  <td>0<td>  <td>5<td>  <td>0<td>  </tr>
- * <tr>  <td><b>6</b><td>  <td>true<td>   <td>false<td>  <td>tukey(0.5)<td>  <td>8<td>   <td>0<td>  <td>false<td>  <td>false<td>  <td>false<td>  <td>0<td>  <td>6<td>  <td>0<td>  </tr>
- * <tr>  <td><b>7</b><td>  <td>true<td>   <td>false<td>  <td>tukey(0.5)<td>  <td>8<td>   <td>0<td>  <td>false<td>  <td>false<td>  <td>true<td>   <td>0<td>  <td>6<td>  <td>0<td>  </tr>
- * <tr>  <td><b>8</b><td>  <td>true<td>   <td>false<td>  <td>tukey(0.5)<td>  <td>12<td>  <td>0<td>  <td>false<td>  <td>false<td>  <td>true<td>   <td>0<td>  <td>6<td>  <td>0<td>  </tr>
+ * <tr>  <td><b>0</b></td> <td>false</td> <td>false</td> <td>tukey(0.5)<td>                                     <td>0</td>  <td>0</td> <td>false</td> <td>false</td> <td>false</td> <td>0</td> <td>3</td> <td>0</td> </tr>
+ * <tr>  <td><b>1</b></td> <td>true</td>  <td>true</td>  <td>tukey(0.5)<td>                                     <td>0</td>  <td>0</td> <td>false</td> <td>false</td> <td>false</td> <td>0</td> <td>3</td> <td>0</td> </tr>
+ * <tr>  <td><b>2</b></td> <td>true</td>  <td>false</td> <td>tukey(0.5)<td>                                     <td>0</td>  <td>0</td> <td>false</td> <td>false</td> <td>false</td> <td>0</td> <td>3</td> <td>0</td> </tr>
+ * <tr>  <td><b>3</b></td> <td>false</td> <td>false</td> <td>tukey(0.5)<td>                                     <td>6</td>  <td>0</td> <td>false</td> <td>false</td> <td>false</td> <td>0</td> <td>4</td> <td>0</td> </tr>
+ * <tr>  <td><b>4</b></td> <td>true</td>  <td>true</td>  <td>tukey(0.5)<td>                                     <td>8</td>  <td>0</td> <td>false</td> <td>false</td> <td>false</td> <td>0</td> <td>4</td> <td>0</td> </tr>
+ * <tr>  <td><b>5</b></td> <td>true</td>  <td>false</td> <td>tukey(0.5)<td>                                     <td>8</td>  <td>0</td> <td>false</td> <td>false</td> <td>false</td> <td>0</td> <td>5</td> <td>0</td> </tr>
+ * <tr>  <td><b>6</b></td> <td>true</td>  <td>false</td> <td>tukey(0.5);partial_tukey(2)<td>                    <td>8</td>  <td>0</td> <td>false</td> <td>false</td> <td>false</td> <td>0</td> <td>6</td> <td>0</td> </tr>
+ * <tr>  <td><b>7</b></td> <td>true</td>  <td>false</td> <td>tukey(0.5);partial_tukey(2)<td>                    <td>12</td> <td>0</td> <td>false</td> <td>false</td> <td>false</td> <td>0</td> <td>6</td> <td>0</td> </tr>
+ * <tr>  <td><b>8</b></td> <td>true</td>  <td>false</td> <td>tukey(0.5);partial_tukey(2);punchout_tukey(3)</td> <td>12</td> <td>0</td> <td>false</td> <td>false</td> <td>false</td> <td>0</td> <td>6</td> <td>0</td> </tr>
  * </table>
  *
  * \default \c 5
@@ -920,7 +920,8 @@ FLAC_API FLAC__bool FLAC__stream_encoder_set_loose_mid_side_stereo(FLAC__StreamE
  * The available functions are \c bartlett, \c bartlett_hann,
  * \c blackman, \c blackman_harris_4term_92db, \c connes, \c flattop,
  * \c gauss(STDDEV), \c hamming, \c hann, \c kaiser_bessel, \c nuttall,
- * \c rectangle, \c triangle, \c tukey(P), \c welch.
+ * \c rectangle, \c triangle, \c tukey(P), \c partial_tukey(n[/ov[/P]]),
+ * \c punchout_tukey(n[/ov[/P]]), \c welch.
  *
  * For \c gauss(STDDEV), STDDEV specifies the standard deviation
  * (0<STDDEV<=0.5).
@@ -929,6 +930,24 @@ FLAC_API FLAC__bool FLAC__stream_encoder_set_loose_mid_side_stereo(FLAC__StreamE
  * tapered (0<=P<=1).  P=0 corresponds to \c rectangle and P=1
  * corresponds to \c hann.
  *
+ * Specifying \c partial_tukey or \c punchout_tukey works a little
+ * different. These do not specify a single apodization function, but
+ * a series of them with some overlap. partial_tukey specifies a series
+ * of small windows (all treated separately) while punchout_tukey
+ * specifies a series of windows that have a hole in them. In this way,
+ * the predictor is constructed with only a part of the block, which
+ * helps in case a block consists of dissimilar parts.
+ *
+ * The three parameters that can be specified for the functions are
+ * n, ov and P. n is the number of functions to add, ov is the overlap
+ * of the windows in case of partial_tukey and the overlap in the gaps
+ * in case of punchout_tukey. P is the fraction of the window that is
+ * tapered, like with a regular tukey window. The function can be
+ * specified with only a number, a number and an overlap, or a number
+ * an overlap and a P, for example, partial_tukey(3), partial_tukey(3/0.3)
+ * and partial_tukey(3/0.3/0.5) are all valid. ov should be smaller than 1
+ * and can be negative.
+ *
  * Example specifications are \c "blackman" or
  * \c "hann;triangle;tukey(0.5);tukey(0.25);tukey(0.125)"
  *
@@ -941,7 +960,9 @@ FLAC_API FLAC__bool FLAC__stream_encoder_set_loose_mid_side_stereo(FLAC__StreamE
  * results in the smallest compressed subframe.
  *
  * Note that each function specified causes the encoder to occupy a
- * floating point array in which to store the window.
+ * floating point array in which to store the window. Also note that the
+ * values of P, STDDEV and ov are locale-specific, so if the comma
+ * separator specified by the locale is a comma, a comma should be used.
  *
  * \default \c "tukey(0.5)"
  * \param  encoder        An encoder instance to set.

source/modules/juce_audio_formats/codecs/flac/win_utf8_io.h

@@ -0,0 +1,69 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2013-2014  Xiph.Org Foundation
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifdef _WIN32
+
+#ifndef flac__win_utf8_io_h
+#define flac__win_utf8_io_h
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#include <stdio.h>
+#include <sys/stat.h>
+#include <stdarg.h>
+#include <windows.h>
+
+int get_utf8_argv(int *argc, char ***argv);
+
+int printf_utf8(const char *format, ...);
+int fprintf_utf8(FILE *stream, const char *format, ...);
+int vfprintf_utf8(FILE *stream, const char *format, va_list argptr);
+
+FILE *fopen_utf8(const char *filename, const char *mode);
+int stat_utf8(const char *path, struct stat *buffer);
+int _stat64_utf8(const char *path, struct __stat64 *buffer);
+int chmod_utf8(const char *filename, int pmode);
+int utime_utf8(const char *filename, struct utimbuf *times);
+int unlink_utf8(const char *filename);
+int rename_utf8(const char *oldname, const char *newname);
+size_t strlen_utf8(const char *str);
+int win_get_console_width(void);
+int print_console(FILE *stream, const wchar_t *text, size_t len);
+HANDLE WINAPI CreateFile_utf8(const char *lpFileName, DWORD dwDesiredAccess, DWORD dwShareMode, LPSECURITY_ATTRIBUTES lpSecurityAttributes, DWORD dwCreationDisposition, DWORD dwFlagsAndAttributes, HANDLE hTemplateFile);
+
+#ifdef __cplusplus
+} /* extern "C" */
+#endif
+
+#endif
+#endif

source/modules/juce_audio_formats/codecs/juce_AiffAudioFormat.cpp

@@ -193,47 +193,72 @@ namespace AiffFileHelpers
    #endif
 
     //==============================================================================
-    static String readCATEChunk (InputStream& input, const uint32 length)
+    namespace CATEChunk
     {
-        MemoryBlock mb;
-        input.skipNextBytes (4);
-        input.readIntoMemoryBlock (mb, (ssize_t) length - 4);
-
-        static const char* appleGenres[] =
-        {
-            "Rock/Blues",
-            "Electronic/Dance",
-            "Jazz",
-            "Urban",
-            "World/Ethnic",
-            "Cinematic/New Age",
-            "Orchestral",
-            "Country/Folk",
-            "Experimental",
-            "Other Genre",
-            nullptr
-        };
+        static bool isValidTag (const char* d) noexcept
+        {
+            return CharacterFunctions::isLetterOrDigit (d[0]) && CharacterFunctions::isUpperCase (d[0])
+                && CharacterFunctions::isLetterOrDigit (d[1]) && CharacterFunctions::isLowerCase (d[1])
+                && CharacterFunctions::isLetterOrDigit (d[2]) && CharacterFunctions::isLowerCase (d[2]);
+        }
 
-        const StringArray genres (appleGenres);
-        StringArray tagsArray;
+        static bool isAppleGenre (const String& tag) noexcept
+        {
+            static const char* appleGenres[] =
+            {
+                "Rock/Blues",
+                "Electronic/Dance",
+                "Jazz",
+                "Urban",
+                "World/Ethnic",
+                "Cinematic/New Age",
+                "Orchestral",
+                "Country/Folk",
+                "Experimental",
+                "Other Genre"
+            };
+
+            for (int i = 0; i < numElementsInArray (appleGenres); ++i)
+                if (tag == appleGenres[i])
+                    return true;
 
-        int bytesLeft = (int) mb.getSize();
-        const char* data = static_cast<const char*> (mb.getData());
+            return false;
+        }
 
-        while (bytesLeft > 0)
+        static String read (InputStream& input, const uint32 length)
         {
-            const String tag (CharPointer_UTF8 (data),
-                              CharPointer_UTF8 (data + bytesLeft));
+            MemoryBlock mb;
+            input.skipNextBytes (4);
+            input.readIntoMemoryBlock (mb, (ssize_t) length - 4);
 
-            if (tag.isNotEmpty())
-                tagsArray.add (data);
+            StringArray tagsArray;
 
-            const int numBytesInTag = genres.contains (tag) ? 118 : 50;
-            data += numBytesInTag;
-            bytesLeft -= numBytesInTag;
-        }
+            const char* data = static_cast<const char*> (mb.getData());
+            const char* dataEnd = data + mb.getSize();
+
+            while (data < dataEnd)
+            {
+                bool isGenre = false;
 
-        return tagsArray.joinIntoString (";");
+                if (isValidTag (data))
+                {
+                    const String tag = String (CharPointer_UTF8 (data), CharPointer_UTF8 (dataEnd));
+                    isGenre = isAppleGenre (tag);
+                    tagsArray.add (tag);
+                }
+
+                data += isGenre ? 118 : 50;
+
+                if (data[0] == 0)
+                {
+                    if      (data + 52  < dataEnd && isValidTag (data + 50))   data += 50;
+                    else if (data + 120 < dataEnd && isValidTag (data + 118))  data += 118;
+                    else if (data + 170 < dataEnd && isValidTag (data + 168))  data += 168;
+                }
+            }
+
+            return tagsArray.joinIntoString (";");
+        }
     }
 
     //==============================================================================
@@ -518,7 +543,7 @@ public:
                     else if (type == chunkName ("cate"))
                     {
                         metadataValues.set (AiffAudioFormat::appleTag,
-                                            AiffFileHelpers::readCATEChunk (*input, length));;
+                                            AiffFileHelpers::CATEChunk::read (*input, length));
                     }
                     else if ((hasGotVer && hasGotData && hasGotType)
                               || chunkEnd < input->getPosition()
@@ -821,12 +846,54 @@ public:
         return true;
     }
 
-    void readMaxLevels (int64 startSampleInFile, int64 numSamples,
-                        float& min0, float& max0, float& min1, float& max1)
+    void getSample (int64 sample, float* result) const noexcept override
+    {
+        const int num = (int) numChannels;
+
+        if (map == nullptr || ! mappedSection.contains (sample))
+        {
+            jassertfalse; // you must make sure that the window contains all the samples you're going to attempt to read.
+
+            zeromem (result, sizeof (float) * (size_t) num);
+            return;
+        }
+
+        float** dest = &result;
+        const void* source = sampleToPointer (sample);
+
+        if (littleEndian)
+        {
+            switch (bitsPerSample)
+            {
+                case 8:     ReadHelper<AudioData::Float32, AudioData::UInt8, AudioData::LittleEndian>::read (dest, 0, 1, source, 1, num); break;
+                case 16:    ReadHelper<AudioData::Float32, AudioData::Int16, AudioData::LittleEndian>::read (dest, 0, 1, source, 1, num); break;
+                case 24:    ReadHelper<AudioData::Float32, AudioData::Int24, AudioData::LittleEndian>::read (dest, 0, 1, source, 1, num); break;
+                case 32:    if (usesFloatingPointData) ReadHelper<AudioData::Float32, AudioData::Float32, AudioData::LittleEndian>::read (dest, 0, 1, source, 1, num);
+                            else                       ReadHelper<AudioData::Float32, AudioData::Int32,   AudioData::LittleEndian>::read (dest, 0, 1, source, 1, num); break;
+                default:    jassertfalse; break;
+            }
+        }
+        else
+        {
+            switch (bitsPerSample)
+            {
+                case 8:     ReadHelper<AudioData::Float32, AudioData::UInt8, AudioData::BigEndian>::read (dest, 0, 1, source, 1, num); break;
+                case 16:    ReadHelper<AudioData::Float32, AudioData::Int16, AudioData::BigEndian>::read (dest, 0, 1, source, 1, num); break;
+                case 24:    ReadHelper<AudioData::Float32, AudioData::Int24, AudioData::BigEndian>::read (dest, 0, 1, source, 1, num); break;
+                case 32:    if (usesFloatingPointData) ReadHelper<AudioData::Float32, AudioData::Float32, AudioData::BigEndian>::read (dest, 0, 1, source, 1, num);
+                            else                       ReadHelper<AudioData::Float32, AudioData::Int32,   AudioData::BigEndian>::read (dest, 0, 1, source, 1, num); break;
+                default:    jassertfalse; break;
+            }
+        }
+    }
+
+    void readMaxLevels (int64 startSampleInFile, int64 numSamples, Range<float>* results, int numChannelsToRead) override
     {
         if (numSamples <= 0)
         {
-            min0 = max0 = min1 = max1 = 0;
+            for (int i = 0; i < numChannelsToRead; ++i)
+                results[i] = Range<float>();
+
             return;
         }
 
@@ -834,17 +901,19 @@ public:
         {
             jassertfalse; // you must make sure that the window contains all the samples you're going to attempt to read.
 
-            min0 = max0 = min1 = max1 = 0;
+            for (int i = 0; i < numChannelsToRead; ++i)
+                results[i] = Range<float>();
+
             return;
         }
 
         switch (bitsPerSample)
         {
-            case 8:     scanMinAndMax<AudioData::UInt8> (startSampleInFile, numSamples, min0, max0, min1, max1); break;
-            case 16:    scanMinAndMax<AudioData::Int16> (startSampleInFile, numSamples, min0, max0, min1, max1); break;
-            case 24:    scanMinAndMax<AudioData::Int24> (startSampleInFile, numSamples, min0, max0, min1, max1); break;
-            case 32:    if (usesFloatingPointData) scanMinAndMax<AudioData::Float32> (startSampleInFile, numSamples, min0, max0, min1, max1);
-                        else                       scanMinAndMax<AudioData::Int32>   (startSampleInFile, numSamples, min0, max0, min1, max1); break;
+            case 8:     scanMinAndMax<AudioData::UInt8> (startSampleInFile, numSamples, results, numChannelsToRead); break;
+            case 16:    scanMinAndMax<AudioData::Int16> (startSampleInFile, numSamples, results, numChannelsToRead); break;
+            case 24:    scanMinAndMax<AudioData::Int24> (startSampleInFile, numSamples, results, numChannelsToRead); break;
+            case 32:    if (usesFloatingPointData) scanMinAndMax<AudioData::Float32> (startSampleInFile, numSamples, results, numChannelsToRead);
+                        else                       scanMinAndMax<AudioData::Int32>   (startSampleInFile, numSamples, results, numChannelsToRead); break;
             default:    jassertfalse; break;
         }
     }
@@ -853,24 +922,17 @@ private:
     const bool littleEndian;
 
     template <typename SampleType>
-    void scanMinAndMax (int64 startSampleInFile, int64 numSamples,
-                        float& min0, float& max0, float& min1, float& max1) const noexcept
+    void scanMinAndMax (int64 startSampleInFile, int64 numSamples, Range<float>* results, int numChannelsToRead) const noexcept
     {
-        scanMinAndMax2<SampleType> (0, startSampleInFile, numSamples, min0, max0);
-
-        if (numChannels > 1)
-            scanMinAndMax2<SampleType> (1, startSampleInFile, numSamples, min1, max1);
-        else
-            min1 = max1 = 0;
+        for (int i = 0; i < numChannelsToRead; ++i)
+            results[i] = scanMinAndMaxForChannel<SampleType> (i, startSampleInFile, numSamples);
     }
 
     template <typename SampleType>
-    void scanMinAndMax2 (int channel, int64 startSampleInFile, int64 numSamples, float& mn, float& mx) const noexcept
+    Range<float> scanMinAndMaxForChannel (int channel, int64 startSampleInFile, int64 numSamples) const noexcept
     {
-        if (littleEndian)
-            scanMinAndMaxInterleaved<SampleType, AudioData::LittleEndian> (channel, startSampleInFile, numSamples, mn, mx);
-        else
-            scanMinAndMaxInterleaved<SampleType, AudioData::BigEndian>    (channel, startSampleInFile, numSamples, mn, mx);
+        return littleEndian ? scanMinAndMaxInterleaved<SampleType, AudioData::LittleEndian> (channel, startSampleInFile, numSamples)
+                            : scanMinAndMaxInterleaved<SampleType, AudioData::BigEndian>    (channel, startSampleInFile, numSamples);
     }
 
     JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (MemoryMappedAiffReader)

source/modules/juce_audio_formats/codecs/juce_MP3AudioFormat.cpp

@@ -607,7 +607,7 @@ private:
             float* costab = cosTables[i];
 
             for (int k = 0; k < kr; ++k)
-                costab[k] = (float) (1.0 / (2.0 * cos (double_Pi * (k * 2 + 1) / divv)));
+                costab[k] = (float) (1.0 / (2.0 * std::cos (double_Pi * (k * 2 + 1) / divv)));
         }
 
         for (i = 0, j = 0; i < 256; ++i, ++j, table += 32)
@@ -692,23 +692,23 @@ private:
 
         for (i = 0; i < 18; ++i)
         {
-            win[0][i] = win[1][i] = (float) (0.5 * sin (double_Pi / 72.0 * (2 * i + 1)) / cos (double_Pi * (2 * i + 19) / 72.0));
-            win[0][i + 18] = win[3][i + 18] = (float) (0.5 * sin (double_Pi / 72.0 * (2 * (i + 18) + 1)) / cos (double_Pi * (2 * (i + 18) + 19) / 72.0));
+            win[0][i]      = win[1][i]      = (float) (0.5 * std::sin (double_Pi / 72.0 * (2 * i + 1))        / std::cos (double_Pi * (2 * i + 19)        / 72.0));
+            win[0][i + 18] = win[3][i + 18] = (float) (0.5 * std::sin (double_Pi / 72.0 * (2 * (i + 18) + 1)) / std::cos (double_Pi * (2 * (i + 18) + 19) / 72.0));
         }
 
         const double piOver72 = double_Pi;
 
         for (i = 0; i < 6; ++i)
         {
-            win[1][i + 18] = (float) (0.5 / cos (piOver72 * (2 * (i + 18) + 19)));
-            win[3][i + 12] = (float) (0.5 / cos (piOver72 * (2 * (i + 12) + 19)));
-            win[1][i + 24] = (float) (0.5 * sin (double_Pi / 24.0 * (2 * i + 13)) / cos (piOver72 * (2 * (i + 24) + 19)));
+            win[1][i + 18] = (float) (0.5 / std::cos (piOver72 * (2 * (i + 18) + 19)));
+            win[3][i + 12] = (float) (0.5 / std::cos (piOver72 * (2 * (i + 12) + 19)));
+            win[1][i + 24] = (float) (0.5 * std::sin (double_Pi / 24.0 * (2 * i + 13)) / std::cos (piOver72 * (2 * (i + 24) + 19)));
             win[1][i + 30] = win[3][i] = 0;
-            win[3][i + 6]  = (float) (0.5 * sin (double_Pi / 24.0 * (2 * i + 1)) / cos (piOver72 * (2 * (i + 6) + 19)));
+            win[3][i + 6]  = (float) (0.5 * std::sin (double_Pi / 24.0 * (2 * i + 1)) / std::cos (piOver72 * (2 * (i + 6) + 19)));
         }
 
         for (i = 0; i < 12; ++i)
-            win[2][i] = (float) (0.5 * sin (double_Pi / 24.0 * (2 * i + 1)) / cos (double_Pi * (2 * i + 7) / 24.0));
+            win[2][i] = (float) (0.5 * std::sin (double_Pi / 24.0 * (2 * i + 1)) / std::cos (double_Pi * (2 * i + 7) / 24.0));
 
         for (j = 0; j < 4; ++j)
         {
@@ -721,7 +721,7 @@ private:
 
         for (i = 0; i < 16; ++i)
         {
-            const double t = tan (i * double_Pi / 12.0);
+            const double t = std::tan (i * double_Pi / 12.0);
             tan1_1[i] = (float) (t / (1.0 + t));
             tan2_1[i] = (float) (1.0 / (1.0 + t));
             tan1_2[i] = (float) (sqrt2 * t / (1.0 + t));

source/modules/juce_audio_formats/codecs/juce_WavAudioFormat.cpp

@@ -1289,12 +1289,39 @@ public:
         return true;
     }
 
-    void readMaxLevels (int64 startSampleInFile, int64 numSamples,
-                        float& min0, float& max0, float& min1, float& max1) override
+    void getSample (int64 sample, float* result) const noexcept override
+    {
+        const int num = (int) numChannels;
+
+        if (map == nullptr || ! mappedSection.contains (sample))
+        {
+            jassertfalse; // you must make sure that the window contains all the samples you're going to attempt to read.
+
+            zeromem (result, sizeof (float) * (size_t) num);
+            return;
+        }
+
+        float** dest = &result;
+        const void* source = sampleToPointer (sample);
+
+        switch (bitsPerSample)
+        {
+            case 8:     ReadHelper<AudioData::Float32, AudioData::UInt8, AudioData::LittleEndian>::read (dest, 0, 1, source, 1, num); break;
+            case 16:    ReadHelper<AudioData::Float32, AudioData::Int16, AudioData::LittleEndian>::read (dest, 0, 1, source, 1, num); break;
+            case 24:    ReadHelper<AudioData::Float32, AudioData::Int24, AudioData::LittleEndian>::read (dest, 0, 1, source, 1, num); break;
+            case 32:    if (usesFloatingPointData) ReadHelper<AudioData::Float32, AudioData::Float32, AudioData::LittleEndian>::read (dest, 0, 1, source, 1, num);
+                        else                       ReadHelper<AudioData::Float32, AudioData::Int32,   AudioData::LittleEndian>::read (dest, 0, 1, source, 1, num); break;
+            default:    jassertfalse; break;
+        }
+    }
+
+    void readMaxLevels (int64 startSampleInFile, int64 numSamples, Range<float>* results, int numChannelsToRead) override
     {
         if (numSamples <= 0)
         {
-            min0 = max0 = min1 = max1 = 0;
+            for (int i = 0; i < numChannelsToRead; ++i)
+                results[i] = Range<float>();
+
             return;
         }
 
@@ -1302,32 +1329,29 @@ public:
         {
             jassertfalse; // you must make sure that the window contains all the samples you're going to attempt to read.
 
-            min0 = max0 = min1 = max1 = 0;
+            for (int i = 0; i < numChannelsToRead; ++i)
+                results[i] = Range<float>();
+
             return;
         }
 
         switch (bitsPerSample)
         {
-            case 8:     scanMinAndMax<AudioData::UInt8> (startSampleInFile, numSamples, min0, max0, min1, max1); break;
-            case 16:    scanMinAndMax<AudioData::Int16> (startSampleInFile, numSamples, min0, max0, min1, max1); break;
-            case 24:    scanMinAndMax<AudioData::Int24> (startSampleInFile, numSamples, min0, max0, min1, max1); break;
-            case 32:    if (usesFloatingPointData) scanMinAndMax<AudioData::Float32> (startSampleInFile, numSamples, min0, max0, min1, max1);
-                        else                       scanMinAndMax<AudioData::Int32>   (startSampleInFile, numSamples, min0, max0, min1, max1); break;
+            case 8:     scanMinAndMax<AudioData::UInt8> (startSampleInFile, numSamples, results, numChannelsToRead); break;
+            case 16:    scanMinAndMax<AudioData::Int16> (startSampleInFile, numSamples, results, numChannelsToRead); break;
+            case 24:    scanMinAndMax<AudioData::Int24> (startSampleInFile, numSamples, results, numChannelsToRead); break;
+            case 32:    if (usesFloatingPointData) scanMinAndMax<AudioData::Float32> (startSampleInFile, numSamples, results, numChannelsToRead);
+                        else                       scanMinAndMax<AudioData::Int32>   (startSampleInFile, numSamples, results, numChannelsToRead); break;
             default:    jassertfalse; break;
         }
     }
 
 private:
     template <typename SampleType>
-    void scanMinAndMax (int64 startSampleInFile, int64 numSamples,
-                        float& min0, float& max0, float& min1, float& max1) const noexcept
+    void scanMinAndMax (int64 startSampleInFile, int64 numSamples, Range<float>* results, int numChannelsToRead) const noexcept
     {
-        scanMinAndMaxInterleaved<SampleType, AudioData::LittleEndian> (0, startSampleInFile, numSamples, min0, max0);
-
-        if (numChannels > 1)
-            scanMinAndMaxInterleaved<SampleType, AudioData::LittleEndian> (1, startSampleInFile, numSamples, min1, max1);
-        else
-            min1 = max1 = 0;
+        for (int i = 0; i < numChannelsToRead; ++i)
+            results[i] = scanMinAndMaxInterleaved<SampleType, AudioData::LittleEndian> (i, startSampleInFile, numSamples);
     }
 
     JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (MemoryMappedWavReader)

source/modules/juce_audio_formats/format/juce_AudioFormatReader.cpp

@@ -46,6 +46,7 @@ bool AudioFormatReader::read (int* const* destSamples,
 {
     jassert (numDestChannels > 0); // you have to actually give this some channels to work with!
 
+    const size_t originalNumSamplesToRead = (size_t) numSamplesToRead;
     int startOffsetInDestBuffer = 0;
 
     if (startSampleInSource < 0)
@@ -64,7 +65,7 @@ bool AudioFormatReader::read (int* const* destSamples,
     if (numSamplesToRead <= 0)
         return true;
 
-    if (! readSamples (const_cast <int**> (destSamples),
+    if (! readSamples (const_cast<int**> (destSamples),
                        jmin ((int) numChannels, numDestChannels), startOffsetInDestBuffer,
                        startSampleInSource, numSamplesToRead))
         return false;
@@ -87,13 +88,13 @@ bool AudioFormatReader::read (int* const* destSamples,
             if (lastFullChannel != nullptr)
                 for (int i = (int) numChannels; i < numDestChannels; ++i)
                     if (destSamples[i] != nullptr)
-                        memcpy (destSamples[i], lastFullChannel, sizeof (int) * (size_t) numSamplesToRead);
+                        memcpy (destSamples[i], lastFullChannel, sizeof (int) * originalNumSamplesToRead);
         }
         else
         {
             for (int i = (int) numChannels; i < numDestChannels; ++i)
                 if (destSamples[i] != nullptr)
-                    zeromem (destSamples[i], sizeof (int) * (size_t) numSamplesToRead);
+                    zeromem (destSamples[i], sizeof (int) * originalNumSamplesToRead);
         }
     }
 
@@ -229,20 +230,21 @@ void AudioFormatReader::readMaxLevels (int64 startSampleInFile, int64 numSamples
                                        float& lowestRight, float& highestRight)
 {
     Range<float> levels[2];
-    readMaxLevels (startSampleInFile, numSamples, levels, jmin (2, (int) numChannels));
-    lowestLeft  = levels[0].getStart();
-    highestLeft = levels[0].getEnd();
 
-    if (numChannels > 1)
+    if (numChannels < 2)
     {
-        lowestRight  = levels[1].getStart();
-        highestRight = levels[1].getEnd();
+        readMaxLevels (startSampleInFile, numSamples, levels, (int) numChannels);
+        levels[1] = levels[0];
     }
     else
     {
-        lowestRight  = lowestLeft;
-        highestRight = highestLeft;
+        readMaxLevels (startSampleInFile, numSamples, levels, 2);
     }
+
+    lowestLeft   = levels[0].getStart();
+    highestLeft  = levels[0].getEnd();
+    lowestRight  = levels[1].getStart();
+    highestRight = levels[1].getEnd();
 }
 
 int64 AudioFormatReader::searchForLevel (int64 startSample,

source/modules/juce_audio_formats/format/juce_AudioFormatReader.h

@@ -44,7 +44,7 @@ protected:
         @param sourceStream     the stream to read from - this will be deleted
                                 by this object when it is no longer needed. (Some
                                 specialised readers might not use this parameter and
-                                can leave it as 0).
+                                can leave it as nullptr).
         @param formatName       the description that will be returned by the getFormatName()
                                 method
     */
@@ -247,8 +247,8 @@ protected:
     template <class DestSampleType, class SourceSampleType, class SourceEndianness>
     struct ReadHelper
     {
-        typedef AudioData::Pointer <DestSampleType, AudioData::NativeEndian, AudioData::NonInterleaved, AudioData::NonConst>    DestType;
-        typedef AudioData::Pointer <SourceSampleType, SourceEndianness, AudioData::Interleaved, AudioData::Const>               SourceType;
+        typedef AudioData::Pointer<DestSampleType, AudioData::NativeEndian, AudioData::NonInterleaved, AudioData::NonConst>    DestType;
+        typedef AudioData::Pointer<SourceSampleType, SourceEndianness, AudioData::Interleaved, AudioData::Const>               SourceType;
 
         template <typename TargetType>
         static void read (TargetType* const* destData, int destOffset, int numDestChannels,

source/modules/juce_audio_formats/format/juce_AudioSubsectionReader.cpp

@@ -57,17 +57,10 @@ bool AudioSubsectionReader::readSamples (int** destSamples, int numDestChannels,
                                 startSampleInFile + startSample, numSamples);
 }
 
-void AudioSubsectionReader::readMaxLevels (int64 startSampleInFile,
-                                           int64 numSamples,
-                                           float& lowestLeft,
-                                           float& highestLeft,
-                                           float& lowestRight,
-                                           float& highestRight)
+void AudioSubsectionReader::readMaxLevels (int64 startSampleInFile, int64 numSamples, Range<float>* results, int numChannelsToRead)
 {
     startSampleInFile = jmax ((int64) 0, startSampleInFile);
     numSamples = jmax ((int64) 0, jmin (numSamples, length - startSampleInFile));
 
-    source->readMaxLevels (startSampleInFile + startSample, numSamples,
-                           lowestLeft, highestLeft,
-                           lowestRight, highestRight);
+    source->readMaxLevels (startSampleInFile + startSample, numSamples, results, numChannelsToRead);
 }

source/modules/juce_audio_formats/format/juce_AudioSubsectionReader.h

@@ -68,8 +68,7 @@ public:
                       int64 startSampleInFile, int numSamples) override;
 
     void readMaxLevels (int64 startSample, int64 numSamples,
-                        float& lowestLeft,  float& highestLeft,
-                        float& lowestRight, float& highestRight) override;
+                        Range<float>* results, int numChannelsToRead) override;
 
 
 private:

source/modules/juce_audio_formats/format/juce_BufferingAudioFormatReader.cpp

@@ -28,7 +28,8 @@ BufferingAudioReader::BufferingAudioReader (AudioFormatReader* sourceReader,
     : AudioFormatReader (nullptr, sourceReader->getFormatName()),
       source (sourceReader), thread (timeSliceThread),
       nextReadPosition (0),
-      numBlocks (1 + (samplesToBuffer / samplesPerBlock))
+      numBlocks (1 + (samplesToBuffer / samplesPerBlock)),
+      timeoutMs (0)
 {
     sampleRate            = source->sampleRate;
     lengthInSamples       = source->lengthInSamples;

source/modules/juce_audio_formats/format/juce_MemoryMappedAudioFormatReader.h

@@ -70,6 +70,12 @@ public:
     /** Touches the memory for the given sample, to force it to be loaded into active memory. */
     void touchSample (int64 sample) const noexcept;
 
+    /** Returns the samples for all channels at a given sample position.
+        The result array must be large enough to hold a value for each channel
+        that this reader contains.
+    */
+    virtual void getSample (int64 sampleIndex, float* result) const noexcept = 0;
+
     /** Returns the number of bytes currently being mapped */
     size_t getNumBytesUsed() const                          { return map != nullptr ? map->getSize() : 0; }
 
@@ -91,12 +97,12 @@ protected:
 
     /** Used by AudioFormatReader subclasses to scan for min/max ranges in interleaved data. */
     template <typename SampleType, typename Endianness>
-    void scanMinAndMaxInterleaved (int channel, int64 startSampleInFile, int64 numSamples, float& mn, float& mx) const noexcept
+    Range<float> scanMinAndMaxInterleaved (int channel, int64 startSampleInFile, int64 numSamples) const noexcept
     {
         typedef AudioData::Pointer <SampleType, Endianness, AudioData::Interleaved, AudioData::Const> SourceType;
 
-        SourceType (addBytesToPointer (sampleToPointer (startSampleInFile), ((int) bitsPerSample / 8) * channel), (int) numChannels)
-           .findMinAndMax ((size_t) numSamples, mn, mx);
+        return SourceType (addBytesToPointer (sampleToPointer (startSampleInFile), ((int) bitsPerSample / 8) * channel), (int) numChannels)
+                .findMinAndMax ((size_t) numSamples);
     }
 
     JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (MemoryMappedAudioFormatReader)

source/modules/juce_audio_formats/sampler/juce_Sampler.cpp

@@ -119,7 +119,7 @@ void SamplerVoice::startNote (const int midiNoteNumber,
         if (sound->releaseSamples > 0)
             releaseDelta = (float) (-pitchRatio / sound->releaseSamples);
         else
-            releaseDelta = 0.0f;
+            releaseDelta = -1.0f;
     }
     else
     {

source/modules/juce_audio_processors/format_types/juce_VST3PluginFormat.cpp

@@ -364,7 +364,6 @@ class VST3HostContext  : public Vst::IComponentHandler,  // From VST V3.0.0
                          public Vst::IComponentHandler3, // From VST V3.5.0 (also very well named!)
                          public Vst::IContextMenuTarget,
                          public Vst::IHostApplication,
-                         public Vst::IParamValueQueue,
                          public Vst::IUnitHandler
 {
 public:
@@ -684,31 +683,6 @@ public:
         return kNotImplemented;
     }
 
-    //==============================================================================
-    Vst::ParamID PLUGIN_API getParameterId() override
-    {
-        jassertfalse;
-        return 0;
-    }
-
-    Steinberg::int32 PLUGIN_API getPointCount() override
-    {
-        jassertfalse;
-        return 0;
-    }
-
-    tresult PLUGIN_API getPoint (Steinberg::int32, Steinberg::int32&, Vst::ParamValue&) override
-    {
-        jassertfalse;
-        return kResultFalse;
-    }
-
-    tresult PLUGIN_API addPoint (Steinberg::int32, Vst::ParamValue, Steinberg::int32&) override
-    {
-        jassertfalse;
-        return kResultFalse;
-    }
-
     //==============================================================================
     tresult PLUGIN_API notifyUnitSelection (Vst::UnitID) override
     {
@@ -736,7 +710,6 @@ public:
         TEST_FOR_AND_RETURN_IF_VALID (iid, Vst::IComponentHandler3)
         TEST_FOR_AND_RETURN_IF_VALID (iid, Vst::IContextMenuTarget)
         TEST_FOR_AND_RETURN_IF_VALID (iid, Vst::IHostApplication)
-        TEST_FOR_AND_RETURN_IF_VALID (iid, Vst::IParamValueQueue)
         TEST_FOR_AND_RETURN_IF_VALID (iid, Vst::IUnitHandler)
         TEST_FOR_COMMON_BASE_AND_RETURN_IF_VALID (iid, FUnknown, Vst::IComponentHandler)
 
@@ -837,47 +810,26 @@ private:
         //==============================================================================
         tresult PLUGIN_API setInt (AttrID id, Steinberg::int64 value) override
         {
-            jassert (id != nullptr);
-
-            if (! setValueForId (id, value))
-                owner->messageQueue.add (ComSmartPtr<Message> (new Message (*owner, this, id, value)));
-
+            addMessageToQueue (id, value);
             return kResultTrue;
         }
 
         tresult PLUGIN_API setFloat (AttrID id, double value) override
         {
-            jassert (id != nullptr);
-
-            if (! setValueForId (id, value))
-                owner->messageQueue.add (ComSmartPtr<Message> (new Message (*owner, this, id, value)));
-
+            addMessageToQueue (id, value);
             return kResultTrue;
         }
 
         tresult PLUGIN_API setString (AttrID id, const Vst::TChar* string) override
         {
-            jassert (id != nullptr);
-            jassert (string != nullptr);
-
-            const String text (toString (string));
-
-            if (! setValueForId (id, text))
-                owner->messageQueue.add (ComSmartPtr<Message> (new Message (*owner, this, id, text)));
-
+            addMessageToQueue (id, toString (string));
             return kResultTrue;
         }
 
         tresult PLUGIN_API setBinary (AttrID id, const void* data, Steinberg::uint32 size) override
         {
-            jassert (id != nullptr);
-            jassert (data != nullptr && size > 0);
-
-            MemoryBlock block (data, (size_t) size);
-
-            if (! setValueForId (id, block))
-                owner->messageQueue.add (ComSmartPtr<Message> (new Message (*owner, this, id, block)));
-
+            jassert (size >= 0 && (data != nullptr || size == 0));
+            addMessageToQueue (id, MemoryBlock (data, (size_t) size));
             return kResultTrue;
         }
 
@@ -886,7 +838,7 @@ private:
         {
             jassert (id != nullptr);
 
-            if (fetchValueForId (id, result))
+            if (findMessageOnQueueWithID (id, result))
                 return kResultTrue;
 
             jassertfalse;
@@ -897,7 +849,7 @@ private:
         {
             jassert (id != nullptr);
 
-            if (fetchValueForId (id, result))
+            if (findMessageOnQueueWithID (id, result))
                 return kResultTrue;
 
             jassertfalse;
@@ -909,7 +861,7 @@ private:
             jassert (id != nullptr);
 
             String stringToFetch;
-            if (fetchValueForId (id, stringToFetch))
+            if (findMessageOnQueueWithID (id, stringToFetch))
             {
                 Steinberg::String str (stringToFetch.toRawUTF8());
                 str.copyTo (result, 0, (Steinberg::int32) jmin (length, (Steinberg::uint32) std::numeric_limits<Steinberg::int32>::max()));
@@ -949,7 +901,7 @@ private:
 
         //==============================================================================
         template<typename Type>
-        bool setValueForId (AttrID id, const Type& value)
+        void addMessageToQueue (AttrID id, const Type& value)
         {
             jassert (id != nullptr);
 
@@ -960,15 +912,15 @@ private:
                 if (std::strcmp (message->getMessageID(), id) == 0)
                 {
                     message->value = value;
-                    return true;
+                    return;
                 }
             }
 
-            return false; // No message found with that Id
+            owner->messageQueue.add (ComSmartPtr<Message> (new Message (*owner, this, id, value)));
         }
 
         template<typename Type>
-        bool fetchValueForId (AttrID id, Type& value)
+        bool findMessageOnQueueWithID (AttrID id, Type& value)
         {
             jassert (id != nullptr);
 
@@ -1632,8 +1584,8 @@ public:
       : module (handle),
         numInputAudioBusses (0),
         numOutputAudioBusses (0),
-        inputParameterChanges (new ParameterChangeList()),
-        outputParameterChanges (new ParameterChangeList()),
+        inputParameterChanges (new ParamValueQueueList()),
+        outputParameterChanges (new ParamValueQueueList()),
         midiInputs (new MidiEventList()),
         midiOutputs (new MidiEventList()),
         isComponentInitialised (false),
@@ -1847,6 +1799,8 @@ public:
             processor->process (data);
 
             MidiEventList::toMidiBuffer (midiMessages, *midiOutputs);
+
+            inputParameterChanges->clearAllQueues();
         }
     }
 
@@ -1977,8 +1931,11 @@ public:
     {
         if (editController != nullptr)
         {
-            const uint32 id = getParameterInfoForIndex (parameterIndex).id;
-            editController->setParamNormalized (id, (double) newValue);
+            const uint32 paramID = getParameterInfoForIndex (parameterIndex).id;
+            editController->setParamNormalized (paramID, (double) newValue);
+
+            Steinberg::int32 index;
+            inputParameterChanges->addParameterData (paramID, index)->addPoint (0, newValue, index);
         }
     }
 
@@ -2053,6 +2010,115 @@ public:
         (void) sizeInBytes;
     }
 
+    //==============================================================================
+    // NB: this class and its subclasses must be public to avoid problems in
+    // DLL builds under MSVC.
+    class ParamValueQueueList  : public Vst::IParameterChanges
+    {
+    public:
+        ParamValueQueueList() {}
+        virtual ~ParamValueQueueList() {}
+
+        JUCE_DECLARE_VST3_COM_REF_METHODS
+        JUCE_DECLARE_VST3_COM_QUERY_METHODS
+
+        Steinberg::int32 PLUGIN_API getParameterCount() override                                { return (Steinberg::int32) queues.size(); }
+        Vst::IParamValueQueue* PLUGIN_API getParameterData (Steinberg::int32 index) override    { return queues[(int) index]; }
+
+        Vst::IParamValueQueue* PLUGIN_API addParameterData (const Vst::ParamID& id, Steinberg::int32& index) override
+        {
+            for (int i = queues.size(); --i >= 0;)
+            {
+                if (queues.getUnchecked (i)->getParameterId() == id)
+                {
+                    index = (Steinberg::int32) i;
+                    return queues.getUnchecked (i);
+                }
+            }
+
+            index = getParameterCount();
+            return queues.add (new ParamValueQueue (id));
+        }
+
+        void clearAllQueues() noexcept
+        {
+            for (int i = queues.size(); --i >= 0;)
+                queues.getUnchecked (i)->clear();
+        }
+
+        struct ParamValueQueue  : public Vst::IParamValueQueue
+        {
+            ParamValueQueue (Vst::ParamID parameterID) : paramID (parameterID)
+            {
+                points.ensureStorageAllocated (1024);
+            }
+
+            virtual ~ParamValueQueue() {}
+
+            JUCE_DECLARE_VST3_COM_REF_METHODS
+            JUCE_DECLARE_VST3_COM_QUERY_METHODS
+
+            Steinberg::Vst::ParamID PLUGIN_API getParameterId() override    { return paramID; }
+            Steinberg::int32 PLUGIN_API getPointCount() override            { return (Steinberg::int32) points.size(); }
+
+            Steinberg::tresult PLUGIN_API getPoint (Steinberg::int32 index,
+                                                    Steinberg::int32& sampleOffset,
+                                                    Steinberg::Vst::ParamValue& value) override
+            {
+                const ScopedLock sl (pointLock);
+
+                if (isPositiveAndBelow ((int) index, points.size()))
+                {
+                    ParamPoint e (points.getUnchecked ((int) index));
+                    sampleOffset = e.sampleOffset;
+                    value = e.value;
+                    return kResultTrue;
+                }
+
+                sampleOffset = -1;
+                value = 0.0;
+                return kResultFalse;
+            }
+
+            Steinberg::tresult PLUGIN_API addPoint (Steinberg::int32 sampleOffset,
+                                                    Steinberg::Vst::ParamValue value,
+                                                    Steinberg::int32& index) override
+            {
+                ParamPoint p = { sampleOffset, value };
+
+                const ScopedLock sl (pointLock);
+                index = (Steinberg::int32) points.size();
+                points.add (p);
+                return kResultTrue;
+            }
+
+            void clear() noexcept
+            {
+                const ScopedLock sl (pointLock);
+                points.clearQuick();
+            }
+
+        private:
+            struct ParamPoint
+            {
+                Steinberg::int32 sampleOffset;
+                Steinberg::Vst::ParamValue value;
+            };
+
+            Atomic<int> refCount;
+            const Vst::ParamID paramID;
+            Array<ParamPoint> points;
+            CriticalSection pointLock;
+
+            JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (ParamValueQueue)
+        };
+
+        Atomic<int> refCount;
+        OwnedArray<ParamValueQueue> queues;
+
+        JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (ParamValueQueueList)
+    };
+
 private:
     //==============================================================================
     VST3ModuleHandle::Ptr module;
@@ -2098,7 +2164,7 @@ private:
 
             if (object->getState (&stream) == kResultTrue)
             {
-                MemoryBlock info (stream.getData(), (std::size_t) stream.getSize());
+                MemoryBlock info (stream.getData(), (size_t) stream.getSize());
                 head.createNewChildElement (identifier)->addTextElement (info.toBase64Encoding());
             }
         }
@@ -2123,36 +2189,7 @@ private:
         return stream;
     }
 
-    //==============================================================================
-    class ParameterChangeList  : public Vst::IParameterChanges
-    {
-    public:
-        ParameterChangeList() {}
-        virtual ~ParameterChangeList() {}
-
-        JUCE_DECLARE_VST3_COM_REF_METHODS
-        JUCE_DECLARE_VST3_COM_QUERY_METHODS
-
-        Steinberg::int32 PLUGIN_API getParameterCount() override   { return 0; }
-
-        Vst::IParamValueQueue* PLUGIN_API getParameterData (Steinberg::int32) override
-        {
-            return nullptr;
-        }
-
-        Vst::IParamValueQueue* PLUGIN_API addParameterData (const Vst::ParamID&, Steinberg::int32& index) override
-        {
-            index = 0;
-            return nullptr;
-        }
-
-    private:
-        Atomic<int> refCount;
-
-        JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (ParameterChangeList)
-    };
-
-    ComSmartPtr<ParameterChangeList> inputParameterChanges, outputParameterChanges;
+    ComSmartPtr<ParamValueQueueList> inputParameterChanges, outputParameterChanges;
     ComSmartPtr<MidiEventList> midiInputs, midiOutputs;
     Vst::ProcessContext timingInfo; //< Only use this in processBlock()!
     bool isComponentInitialised, isControllerInitialised, isActive;

source/modules/juce_audio_processors/format_types/juce_VSTPluginFormat.cpp

@@ -135,7 +135,14 @@ namespace
         char chunk[8];          // variable
     };
 
-    static VstInt32 fxbName (const char* name) noexcept   { return (VstInt32) ByteOrder::bigEndianInt (name); }
+    // Compares a magic value in either endianness.
+    static bool compareMagic (VstInt32 magic, const char* name) noexcept
+    {
+        return magic == (VstInt32) ByteOrder::littleEndianInt (name)
+            || magic == (VstInt32) ByteOrder::bigEndianInt (name);
+    }
+
+    static VstInt32 fxbName (const char* name) noexcept   { return (VstInt32) ByteOrder::littleEndianInt (name); }
     static VstInt32 fxbSwap (const VstInt32 x) noexcept   { return (VstInt32) ByteOrder::swapIfLittleEndian ((uint32) x); }
 
     static float fxbSwapFloat (const float x) noexcept
@@ -1013,12 +1020,12 @@ public:
                 if (position.isLooping)
                 {
                     vstHostTime.cycleStartPos = position.ppqLoopStart;
-                    vstHostTime.cycleEndPos = position.ppqLoopEnd;
-                    vstHostTime.flags |= kVstCyclePosValid;
+                    vstHostTime.cycleEndPos   = position.ppqLoopEnd;
+                    vstHostTime.flags |= (kVstCyclePosValid | kVstTransportCycleActive);
                 }
                 else
                 {
-                    vstHostTime.flags &= ~kVstCyclePosValid;
+                    vstHostTime.flags &= ~(kVstCyclePosValid | kVstTransportCycleActive);
                 }
             }
 
@@ -1432,11 +1439,10 @@ public:
 
         const fxSet* const set = (const fxSet*) data;
 
-        if ((set->chunkMagic != fxbName ("CcnK") && set->chunkMagic != fxbName ("KncC"))
-             || fxbSwap (set->version) > fxbVersionNum)
+        if ((! compareMagic (set->chunkMagic, "CcnK")) || fxbSwap (set->version) > fxbVersionNum)
             return false;
 
-        if (set->fxMagic == fxbName ("FxBk"))
+        if (compareMagic (set->fxMagic, "FxBk"))
         {
             // bank of programs
             if (fxbSwap (set->numPrograms) >= 0)
@@ -1472,12 +1478,12 @@ public:
                     return false;
             }
         }
-        else if (set->fxMagic == fxbName ("FxCk"))
+        else if (compareMagic (set->fxMagic, "FxCk"))
         {
             // single program
             const fxProgram* const prog = (const fxProgram*) data;
 
-            if (prog->chunkMagic != fxbName ("CcnK"))
+            if (! compareMagic (prog->chunkMagic, "CcnK"))
                 return false;
 
             changeProgramName (getCurrentProgram(), prog->prgName);
@@ -1485,7 +1491,7 @@ public:
             for (int i = 0; i < fxbSwap (prog->numParams); ++i)
                 setParameter (i, fxbSwapFloat (prog->params[i]));
         }
-        else if (set->fxMagic == fxbName ("FBCh") || set->fxMagic == fxbName ("hCBF"))
+        else if (compareMagic (set->fxMagic, "FBCh"))
         {
             // non-preset chunk
             const fxChunkSet* const cset = (const fxChunkSet*) data;
@@ -1495,7 +1501,7 @@ public:
 
             setChunkData (cset->chunk, fxbSwap (cset->chunkSize), false);
         }
-        else if (set->fxMagic == fxbName ("FPCh") || set->fxMagic == fxbName ("hCPF"))
+        else if (compareMagic (set->fxMagic, "FPCh"))
         {
             // preset chunk
             const fxProgramSet* const cset = (const fxProgramSet*) data;
@@ -1671,7 +1677,8 @@ private:
 
     bool restoreProgramSettings (const fxProgram* const prog)
     {
-        if (prog->chunkMagic == fxbName ("CcnK") && prog->fxMagic == fxbName ("FxCk"))
+        if (compareMagic (prog->chunkMagic, "CcnK")
+             && compareMagic (prog->fxMagic, "FxCk"))
         {
             changeProgramName (getCurrentProgram(), prog->prgName);
 

source/modules/juce_audio_processors/juce_audio_processors.cpp

@@ -79,71 +79,61 @@ static inline bool arrayContainsPlugin (const OwnedArray<PluginDescription>& lis
 
 #if JUCE_MAC
 //==============================================================================
-struct AutoResizingNSViewComponent : public NSViewComponent,
-                                     private AsyncUpdater {
-    AutoResizingNSViewComponent();
-    void childBoundsChanged(Component*) override;
-    void handleAsyncUpdate() override;
-    bool recursive;
-};
-
-struct AutoResizingNSViewComponentWithParent : public AutoResizingNSViewComponent,
-                                               private Timer {
-    AutoResizingNSViewComponentWithParent();
-    NSView* getChildView() const;
-    void timerCallback() override;
-};
-
-//==============================================================================
-AutoResizingNSViewComponent::AutoResizingNSViewComponent()
-    : recursive (false) {}
-
-void AutoResizingNSViewComponent::childBoundsChanged(Component*)
+struct AutoResizingNSViewComponent  : public NSViewComponent,
+                                      private AsyncUpdater
 {
-    if (recursive)
-    {
-        triggerAsyncUpdate();
-    }
-    else
+    AutoResizingNSViewComponent() : recursive (false) {}
+
+    void childBoundsChanged (Component*) override
     {
-        recursive = true;
-        resizeToFitView();
-        recursive = true;
+        if (recursive)
+        {
+            triggerAsyncUpdate();
+        }
+        else
+        {
+            recursive = true;
+            resizeToFitView();
+            recursive = true;
+        }
     }
-}
 
-void AutoResizingNSViewComponent::handleAsyncUpdate()
-{
-    resizeToFitView();
-}
+    void handleAsyncUpdate() override               { resizeToFitView(); }
+
+    bool recursive;
+};
 
 //==============================================================================
-AutoResizingNSViewComponentWithParent::AutoResizingNSViewComponentWithParent()
+struct AutoResizingNSViewComponentWithParent  : public AutoResizingNSViewComponent,
+                                                private Timer
 {
-    NSView* v = [[NSView alloc] init];
-    setView (v);
-    [v release];
+    AutoResizingNSViewComponentWithParent()
+    {
+        NSView* v = [[NSView alloc] init];
+        setView (v);
+        [v release];
 
-    startTimer(500);
-}
+        startTimer (30);
+    }
 
-NSView* AutoResizingNSViewComponentWithParent::getChildView() const
-{
-    if (NSView* parent = (NSView*)getView())
-        if ([[parent subviews] count] > 0)
-            return [[parent subviews] objectAtIndex: 0];
+    NSView* getChildView() const
+    {
+        if (NSView* parent = (NSView*) getView())
+            if ([[parent subviews] count] > 0)
+                return [[parent subviews] objectAtIndex: 0];
 
-    return nil;
-}
+        return nil;
+    }
 
-void AutoResizingNSViewComponentWithParent::timerCallback()
-{
-    if (NSView* child = getChildView())
+    void timerCallback() override
     {
-        stopTimer();
-        setView(child);
+        if (NSView* child = getChildView())
+        {
+            stopTimer();
+            setView (child);
+        }
     }
-}
+};
 #endif
 
 #if JUCE_CLANG

source/modules/juce_audio_processors/processors/juce_AudioPlayHead.h

@@ -130,8 +130,10 @@ public:
     /** Fills-in the given structure with details about the transport's
         position at the start of the current processing block.
 
-        This method must ONLY be called from within your AudioProcessor::processBlock()
-        method. Calling it at any other time will probably cause a nasty crash.
+        You can ONLY call this from your processBlock() method! Calling it at other
+        times will produce undefined behaviour, as the host may not have any context
+        in which a time would make sense, and some hosts will almost certainly have
+        multithreading issues if it's not called on the audio thread.
     */
     virtual bool getCurrentPosition (CurrentPositionInfo& result) = 0;
 };

source/modules/juce_audio_processors/processors/juce_AudioProcessor.h

@@ -140,11 +140,17 @@ public:
     /** Returns the current AudioPlayHead object that should be used to find
         out the state and position of the playhead.
 
-        You can call this from your processBlock() method, and use the AudioPlayHead
-        object to get the details about the time of the start of the block currently
-        being processed.
+        You can ONLY call this from your processBlock() method! Calling it at other
+        times will produce undefined behaviour, as the host may not have any context
+        in which a time would make sense, and some hosts will almost certainly have
+        multithreading issues if it's not called on the audio thread.
 
-        If the host hasn't supplied a playhead object, this will return nullptr.
+        The AudioPlayHead object that is returned can be used to get the details about
+        the time of the start of the block currently being processed. But do not
+        store this pointer or use it outside of the current audio callback, because
+        the host may delete or re-use it.
+
+        If the host can't or won't provide any time info, this will return nullptr.
     */
     AudioPlayHead* getPlayHead() const noexcept                 { return playHead; }
 

source/modules/juce_audio_processors/processors/juce_AudioProcessorGraph.cpp

@@ -29,151 +29,129 @@ namespace GraphRenderingOps
 {
 
 //==============================================================================
-class AudioGraphRenderingOp
+struct AudioGraphRenderingOp
 {
-public:
-    AudioGraphRenderingOp() {}
-    virtual ~AudioGraphRenderingOp()  {}
+    AudioGraphRenderingOp() noexcept {}
+    virtual ~AudioGraphRenderingOp() {}
 
     virtual void perform (AudioSampleBuffer& sharedBufferChans,
-                          const OwnedArray <MidiBuffer>& sharedMidiBuffers,
+                          const OwnedArray<MidiBuffer>& sharedMidiBuffers,
                           const int numSamples) = 0;
 
     JUCE_LEAK_DETECTOR (AudioGraphRenderingOp)
 };
 
 //==============================================================================
-class ClearChannelOp : public AudioGraphRenderingOp
+struct ClearChannelOp  : public AudioGraphRenderingOp
 {
-public:
-    ClearChannelOp (const int channelNum_)
-        : channelNum (channelNum_)
-    {}
+    ClearChannelOp (const int channel) noexcept  : channelNum (channel)  {}
 
-    void perform (AudioSampleBuffer& sharedBufferChans, const OwnedArray <MidiBuffer>&, const int numSamples)
+    void perform (AudioSampleBuffer& sharedBufferChans, const OwnedArray<MidiBuffer>&, const int numSamples)
     {
         sharedBufferChans.clear (channelNum, 0, numSamples);
     }
 
-private:
     const int channelNum;
 
     JUCE_DECLARE_NON_COPYABLE (ClearChannelOp)
 };
 
 //==============================================================================
-class CopyChannelOp : public AudioGraphRenderingOp
+struct CopyChannelOp  : public AudioGraphRenderingOp
 {
-public:
-    CopyChannelOp (const int srcChannelNum_, const int dstChannelNum_)
-        : srcChannelNum (srcChannelNum_),
-          dstChannelNum (dstChannelNum_)
+    CopyChannelOp (const int srcChan, const int dstChan) noexcept
+        : srcChannelNum (srcChan), dstChannelNum (dstChan)
     {}
 
-    void perform (AudioSampleBuffer& sharedBufferChans, const OwnedArray <MidiBuffer>&, const int numSamples)
+    void perform (AudioSampleBuffer& sharedBufferChans, const OwnedArray<MidiBuffer>&, const int numSamples)
     {
         sharedBufferChans.copyFrom (dstChannelNum, 0, sharedBufferChans, srcChannelNum, 0, numSamples);
     }
 
-private:
     const int srcChannelNum, dstChannelNum;
 
     JUCE_DECLARE_NON_COPYABLE (CopyChannelOp)
 };
 
 //==============================================================================
-class AddChannelOp : public AudioGraphRenderingOp
+struct AddChannelOp  : public AudioGraphRenderingOp
 {
-public:
-    AddChannelOp (const int srcChannelNum_, const int dstChannelNum_)
-        : srcChannelNum (srcChannelNum_),
-          dstChannelNum (dstChannelNum_)
+    AddChannelOp (const int srcChan, const int dstChan) noexcept
+        : srcChannelNum (srcChan), dstChannelNum (dstChan)
     {}
 
-    void perform (AudioSampleBuffer& sharedBufferChans, const OwnedArray <MidiBuffer>&, const int numSamples)
+    void perform (AudioSampleBuffer& sharedBufferChans, const OwnedArray<MidiBuffer>&, const int numSamples)
     {
         sharedBufferChans.addFrom (dstChannelNum, 0, sharedBufferChans, srcChannelNum, 0, numSamples);
     }
 
-private:
     const int srcChannelNum, dstChannelNum;
 
     JUCE_DECLARE_NON_COPYABLE (AddChannelOp)
 };
 
 //==============================================================================
-class ClearMidiBufferOp : public AudioGraphRenderingOp
+struct ClearMidiBufferOp  : public AudioGraphRenderingOp
 {
-public:
-    ClearMidiBufferOp (const int bufferNum_)
-        : bufferNum (bufferNum_)
-    {}
+    ClearMidiBufferOp (const int buffer) noexcept  : bufferNum (buffer)  {}
 
-    void perform (AudioSampleBuffer&, const OwnedArray <MidiBuffer>& sharedMidiBuffers, const int)
+    void perform (AudioSampleBuffer&, const OwnedArray<MidiBuffer>& sharedMidiBuffers, const int)
     {
         sharedMidiBuffers.getUnchecked (bufferNum)->clear();
     }
 
-private:
     const int bufferNum;
 
     JUCE_DECLARE_NON_COPYABLE (ClearMidiBufferOp)
 };
 
 //==============================================================================
-class CopyMidiBufferOp : public AudioGraphRenderingOp
+struct CopyMidiBufferOp  : public AudioGraphRenderingOp
 {
-public:
-    CopyMidiBufferOp (const int srcBufferNum_, const int dstBufferNum_)
-        : srcBufferNum (srcBufferNum_),
-          dstBufferNum (dstBufferNum_)
+    CopyMidiBufferOp (const int srcBuffer, const int dstBuffer) noexcept
+        : srcBufferNum (srcBuffer), dstBufferNum (dstBuffer)
     {}
 
-    void perform (AudioSampleBuffer&, const OwnedArray <MidiBuffer>& sharedMidiBuffers, const int)
+    void perform (AudioSampleBuffer&, const OwnedArray<MidiBuffer>& sharedMidiBuffers, const int)
     {
         *sharedMidiBuffers.getUnchecked (dstBufferNum) = *sharedMidiBuffers.getUnchecked (srcBufferNum);
     }
 
-private:
     const int srcBufferNum, dstBufferNum;
 
     JUCE_DECLARE_NON_COPYABLE (CopyMidiBufferOp)
 };
 
 //==============================================================================
-class AddMidiBufferOp : public AudioGraphRenderingOp
+struct AddMidiBufferOp  : public AudioGraphRenderingOp
 {
-public:
-    AddMidiBufferOp (const int srcBufferNum_, const int dstBufferNum_)
-        : srcBufferNum (srcBufferNum_),
-          dstBufferNum (dstBufferNum_)
+    AddMidiBufferOp (const int srcBuffer, const int dstBuffer)
+        : srcBufferNum (srcBuffer), dstBufferNum (dstBuffer)
     {}
 
-    void perform (AudioSampleBuffer&, const OwnedArray <MidiBuffer>& sharedMidiBuffers, const int numSamples)
+    void perform (AudioSampleBuffer&, const OwnedArray<MidiBuffer>& sharedMidiBuffers, const int numSamples)
     {
         sharedMidiBuffers.getUnchecked (dstBufferNum)
             ->addEvents (*sharedMidiBuffers.getUnchecked (srcBufferNum), 0, numSamples, 0);
     }
 
-private:
     const int srcBufferNum, dstBufferNum;
 
     JUCE_DECLARE_NON_COPYABLE (AddMidiBufferOp)
 };
 
 //==============================================================================
-class DelayChannelOp : public AudioGraphRenderingOp
+struct DelayChannelOp  : public AudioGraphRenderingOp
 {
-public:
-    DelayChannelOp (const int channel_, const int numSamplesDelay_)
-        : channel (channel_),
-          bufferSize (numSamplesDelay_ + 1),
-          readIndex (0), writeIndex (numSamplesDelay_)
+    DelayChannelOp (const int chan, const int delaySize)
+        : channel (chan),
+          bufferSize (delaySize + 1),
+          readIndex (0), writeIndex (delaySize)
     {
         buffer.calloc ((size_t) bufferSize);
     }
 
-    void perform (AudioSampleBuffer& sharedBufferChans, const OwnedArray <MidiBuffer>&, const int numSamples)
+    void perform (AudioSampleBuffer& sharedBufferChans, const OwnedArray<MidiBuffer>&, const int numSamples)
     {
         float* data = sharedBufferChans.getWritePointer (channel, 0);
 
@@ -197,18 +175,17 @@ private:
 
 
 //==============================================================================
-class ProcessBufferOp : public AudioGraphRenderingOp
+struct ProcessBufferOp   : public AudioGraphRenderingOp
 {
-public:
-    ProcessBufferOp (const AudioProcessorGraph::Node::Ptr& node_,
-                     const Array <int>& audioChannelsToUse_,
-                     const int totalChans_,
-                     const int midiBufferToUse_)
-        : node (node_),
-          processor (node_->getProcessor()),
-          audioChannelsToUse (audioChannelsToUse_),
-          totalChans (jmax (1, totalChans_)),
-          midiBufferToUse (midiBufferToUse_)
+    ProcessBufferOp (const AudioProcessorGraph::Node::Ptr& n,
+                     const Array<int>& audioChannels,
+                     const int totalNumChans,
+                     const int midiBuffer)
+        : node (n),
+          processor (n->getProcessor()),
+          audioChannelsToUse (audioChannels),
+          totalChans (jmax (1, totalNumChans)),
+          midiBufferToUse (midiBuffer)
     {
         channels.calloc ((size_t) totalChans);
 
@@ -216,7 +193,7 @@ public:
             audioChannelsToUse.add (0);
     }
 
-    void perform (AudioSampleBuffer& sharedBufferChans, const OwnedArray <MidiBuffer>& sharedMidiBuffers, const int numSamples)
+    void perform (AudioSampleBuffer& sharedBufferChans, const OwnedArray<MidiBuffer>& sharedMidiBuffers, const int numSamples)
     {
         for (int i = totalChans; --i >= 0;)
             channels[i] = sharedBufferChans.getWritePointer (audioChannelsToUse.getUnchecked (i), 0);
@@ -232,8 +209,8 @@ public:
 private:
     Array<int> audioChannelsToUse;
     HeapBlock<float*> channels;
-    int totalChans;
-    int midiBufferToUse;
+    const int totalChans;
+    const int midiBufferToUse;
 
     JUCE_DECLARE_NON_COPYABLE (ProcessBufferOp)
 };
@@ -242,15 +219,13 @@ private:
 /** Used to calculate the correct sequence of rendering ops needed, based on
     the best re-use of shared buffers at each stage.
 */
-class RenderingOpSequenceCalculator
+struct RenderingOpSequenceCalculator
 {
-public:
-    //==============================================================================
-    RenderingOpSequenceCalculator (AudioProcessorGraph& graph_,
-                                   const Array<void*>& orderedNodes_,
+    RenderingOpSequenceCalculator (AudioProcessorGraph& g,
+                                   const Array<AudioProcessorGraph::Node*>& nodes,
                                    Array<void*>& renderingOps)
-        : graph (graph_),
-          orderedNodes (orderedNodes_),
+        : graph (g),
+          orderedNodes (nodes),
           totalLatency (0)
     {
         nodeIds.add ((uint32) zeroNodeID); // first buffer is read-only zeros
@@ -260,34 +235,32 @@ public:
 
         for (int i = 0; i < orderedNodes.size(); ++i)
         {
-            createRenderingOpsForNode ((AudioProcessorGraph::Node*) orderedNodes.getUnchecked(i),
-                                       renderingOps, i);
-
+            createRenderingOpsForNode (*orderedNodes.getUnchecked(i), renderingOps, i);
             markAnyUnusedBuffersAsFree (i);
         }
 
         graph.setLatencySamples (totalLatency);
     }
 
-    int getNumBuffersNeeded() const         { return nodeIds.size(); }
-    int getNumMidiBuffersNeeded() const     { return midiNodeIds.size(); }
+    int getNumBuffersNeeded() const noexcept         { return nodeIds.size(); }
+    int getNumMidiBuffersNeeded() const noexcept     { return midiNodeIds.size(); }
 
 private:
     //==============================================================================
     AudioProcessorGraph& graph;
-    const Array<void*>& orderedNodes;
-    Array <int> channels;
-    Array <uint32> nodeIds, midiNodeIds;
+    const Array<AudioProcessorGraph::Node*>& orderedNodes;
+    Array<int> channels;
+    Array<uint32> nodeIds, midiNodeIds;
 
     enum { freeNodeID = 0xffffffff, zeroNodeID = 0xfffffffe };
 
-    static bool isNodeBusy (uint32 nodeID) noexcept { return nodeID != freeNodeID && nodeID != zeroNodeID; }
+    static bool isNodeBusy (uint32 nodeID) noexcept     { return nodeID != freeNodeID && nodeID != zeroNodeID; }
 
-    Array <uint32> nodeDelayIDs;
-    Array <int> nodeDelays;
+    Array<uint32> nodeDelayIDs;
+    Array<int> nodeDelays;
     int totalLatency;
 
-    int getNodeDelay (const uint32 nodeID) const          { return nodeDelays [nodeDelayIDs.indexOf (nodeID)]; }
+    int getNodeDelay (const uint32 nodeID) const        { return nodeDelays [nodeDelayIDs.indexOf (nodeID)]; }
 
     void setNodeDelay (const uint32 nodeID, const int latency)
     {
@@ -320,30 +293,31 @@ private:
     }
 
     //==============================================================================
-    void createRenderingOpsForNode (AudioProcessorGraph::Node* const node,
+    void createRenderingOpsForNode (AudioProcessorGraph::Node& node,
                                     Array<void*>& renderingOps,
                                     const int ourRenderingIndex)
     {
-        const int numIns = node->getProcessor()->getNumInputChannels();
-        const int numOuts = node->getProcessor()->getNumOutputChannels();
+        AudioProcessor& processor = *node.getProcessor();
+        const int numIns  = processor.getNumInputChannels();
+        const int numOuts = processor.getNumOutputChannels();
         const int totalChans = jmax (numIns, numOuts);
 
-        Array <int> audioChannelsToUse;
+        Array<int> audioChannelsToUse;
         int midiBufferToUse = -1;
 
-        int maxLatency = getInputLatencyForNode (node->nodeId);
+        int maxLatency = getInputLatencyForNode (node.nodeId);
 
         for (int inputChan = 0; inputChan < numIns; ++inputChan)
         {
             // get a list of all the inputs to this node
-            Array <uint32> sourceNodes;
+            Array<uint32> sourceNodes;
             Array<int> sourceOutputChans;
 
             for (int i = graph.getNumConnections(); --i >= 0;)
             {
                 const AudioProcessorGraph::Connection* const c = graph.getConnection (i);
 
-                if (c->destNodeId == node->nodeId && c->destChannelIndex == inputChan)
+                if (c->destNodeId == node.nodeId && c->destChannelIndex == inputChan)
                 {
                     sourceNodes.add (c->sourceNodeId);
                     sourceOutputChans.add (c->sourceChannelIndex);
@@ -493,7 +467,7 @@ private:
             audioChannelsToUse.add (bufIndex);
 
             if (inputChan < numOuts)
-                markBufferAsContaining (bufIndex, node->nodeId, inputChan);
+                markBufferAsContaining (bufIndex, node.nodeId, inputChan);
         }
 
         for (int outputChan = numIns; outputChan < numOuts; ++outputChan)
@@ -502,17 +476,17 @@ private:
             jassert (bufIndex != 0);
             audioChannelsToUse.add (bufIndex);
 
-            markBufferAsContaining (bufIndex, node->nodeId, outputChan);
+            markBufferAsContaining (bufIndex, node.nodeId, outputChan);
         }
 
         // Now the same thing for midi..
-        Array <uint32> midiSourceNodes;
+        Array<uint32> midiSourceNodes;
 
         for (int i = graph.getNumConnections(); --i >= 0;)
         {
             const AudioProcessorGraph::Connection* const c = graph.getConnection (i);
 
-            if (c->destNodeId == node->nodeId && c->destChannelIndex == AudioProcessorGraph::midiChannelIndex)
+            if (c->destNodeId == node.nodeId && c->destChannelIndex == AudioProcessorGraph::midiChannelIndex)
                 midiSourceNodes.add (c->sourceNodeId);
         }
 
@@ -521,7 +495,7 @@ private:
             // No midi inputs..
             midiBufferToUse = getFreeBuffer (true); // need to pick a buffer even if the processor doesn't use midi
 
-            if (node->getProcessor()->acceptsMidi() || node->getProcessor()->producesMidi())
+            if (processor.acceptsMidi() || processor.producesMidi())
                 renderingOps.add (new ClearMidiBufferOp (midiBufferToUse));
         }
         else if (midiSourceNodes.size() == 1)
@@ -601,16 +575,16 @@ private:
             }
         }
 
-        if (node->getProcessor()->producesMidi())
-            markBufferAsContaining (midiBufferToUse, node->nodeId,
+        if (processor.producesMidi())
+            markBufferAsContaining (midiBufferToUse, node.nodeId,
                                     AudioProcessorGraph::midiChannelIndex);
 
-        setNodeDelay (node->nodeId, maxLatency + node->getProcessor()->getLatencySamples());
+        setNodeDelay (node.nodeId, maxLatency + processor.getLatencySamples());
 
         if (numOuts == 0)
             totalLatency = maxLatency;
 
-        renderingOps.add (new ProcessBufferOp (node, audioChannelsToUse,
+        renderingOps.add (new ProcessBufferOp (&node, audioChannelsToUse,
                                                totalChans, midiBufferToUse));
     }
 
@@ -872,18 +846,16 @@ struct ConnectionSorter
 }
 
 //==============================================================================
-AudioProcessorGraph::Connection::Connection (const uint32 sourceNodeId_, const int sourceChannelIndex_,
-                                             const uint32 destNodeId_, const int destChannelIndex_) noexcept
-    : sourceNodeId (sourceNodeId_), sourceChannelIndex (sourceChannelIndex_),
-      destNodeId (destNodeId_), destChannelIndex (destChannelIndex_)
+AudioProcessorGraph::Connection::Connection (const uint32 sourceID, const int sourceChannel,
+                                             const uint32 destID, const int destChannel) noexcept
+    : sourceNodeId (sourceID), sourceChannelIndex (sourceChannel),
+      destNodeId (destID), destChannelIndex (destChannel)
 {
 }
 
 //==============================================================================
-AudioProcessorGraph::Node::Node (const uint32 nodeId_, AudioProcessor* const processor_) noexcept
-    : nodeId (nodeId_),
-      processor (processor_),
-      isPrepared (false)
+AudioProcessorGraph::Node::Node (const uint32 nodeID, AudioProcessor* const p) noexcept
+    : nodeId (nodeID), processor (p), isPrepared (false)
 {
     jassert (processor != nullptr);
 }
@@ -916,7 +888,7 @@ void AudioProcessorGraph::Node::unprepare()
 void AudioProcessorGraph::Node::setParentGraph (AudioProcessorGraph* const graph) const
 {
     if (AudioProcessorGraph::AudioGraphIOProcessor* const ioProc
-            = dynamic_cast <AudioProcessorGraph::AudioGraphIOProcessor*> (processor.get()))
+            = dynamic_cast<AudioProcessorGraph::AudioGraphIOProcessor*> (processor.get()))
         ioProc->setParentGraph (graph);
 }
 
@@ -1213,7 +1185,7 @@ void AudioProcessorGraph::buildRenderingSequence()
     {
         MessageManagerLock mml;
 
-        Array<void*> orderedNodes;
+        Array<Node*> orderedNodes;
 
         {
             const GraphRenderingOps::ConnectionLookupTable table (connections);
@@ -1298,6 +1270,22 @@ void AudioProcessorGraph::reset()
         nodes.getUnchecked(i)->getProcessor()->reset();
 }
 
+void AudioProcessorGraph::setNonRealtime (bool isNonRealtime) noexcept
+{
+    const ScopedLock sl (getCallbackLock());
+
+    for (int i = 0; i < nodes.size(); ++i)
+        nodes.getUnchecked(i)->getProcessor()->setNonRealtime (isNonRealtime);
+}
+
+void AudioProcessorGraph::setPlayHead (AudioPlayHead* audioPlayHead)
+{
+    const ScopedLock sl (getCallbackLock());
+
+    for (int i = 0; i < nodes.size(); ++i)
+        nodes.getUnchecked(i)->getProcessor()->setPlayHead (audioPlayHead);
+}
+
 void AudioProcessorGraph::processBlock (AudioSampleBuffer& buffer, MidiBuffer& midiMessages)
 {
     const int numSamples = buffer.getNumSamples();
@@ -1333,20 +1321,19 @@ const String AudioProcessorGraph::getOutputChannelName (int channelIndex) const
     return "Output " + String (channelIndex + 1);
 }
 
-bool AudioProcessorGraph::isInputChannelStereoPair (int /*index*/) const    { return true; }
-bool AudioProcessorGraph::isOutputChannelStereoPair (int /*index*/) const   { return true; }
-bool AudioProcessorGraph::silenceInProducesSilenceOut() const               { return false; }
-double AudioProcessorGraph::getTailLengthSeconds() const                    { return 0; }
-bool AudioProcessorGraph::acceptsMidi() const   { return true; }
-bool AudioProcessorGraph::producesMidi() const  { return true; }
-void AudioProcessorGraph::getStateInformation (juce::MemoryBlock& /*destData*/)   {}
-void AudioProcessorGraph::setStateInformation (const void* /*data*/, int /*sizeInBytes*/)   {}
+bool AudioProcessorGraph::isInputChannelStereoPair (int) const      { return true; }
+bool AudioProcessorGraph::isOutputChannelStereoPair (int) const     { return true; }
+bool AudioProcessorGraph::silenceInProducesSilenceOut() const       { return false; }
+double AudioProcessorGraph::getTailLengthSeconds() const            { return 0; }
+bool AudioProcessorGraph::acceptsMidi() const                       { return true; }
+bool AudioProcessorGraph::producesMidi() const                      { return true; }
+void AudioProcessorGraph::getStateInformation (juce::MemoryBlock&)  {}
+void AudioProcessorGraph::setStateInformation (const void*, int)    {}
 
 
 //==============================================================================
-AudioProcessorGraph::AudioGraphIOProcessor::AudioGraphIOProcessor (const IODeviceType type_)
-    : type (type_),
-      graph (nullptr)
+AudioProcessorGraph::AudioGraphIOProcessor::AudioGraphIOProcessor (const IODeviceType deviceType)
+    : type (deviceType), graph (nullptr)
 {
 }
 
@@ -1492,19 +1479,12 @@ bool AudioProcessorGraph::AudioGraphIOProcessor::isOutputChannelStereoPair (int
     return isInputChannelStereoPair (index);
 }
 
-bool AudioProcessorGraph::AudioGraphIOProcessor::isInput() const   { return type == audioInputNode  || type == midiInputNode; }
-bool AudioProcessorGraph::AudioGraphIOProcessor::isOutput() const  { return type == audioOutputNode || type == midiOutputNode; }
+bool AudioProcessorGraph::AudioGraphIOProcessor::isInput() const noexcept           { return type == audioInputNode  || type == midiInputNode; }
+bool AudioProcessorGraph::AudioGraphIOProcessor::isOutput() const noexcept          { return type == audioOutputNode || type == midiOutputNode; }
 
 bool AudioProcessorGraph::AudioGraphIOProcessor::hasEditor() const                  { return false; }
 AudioProcessorEditor* AudioProcessorGraph::AudioGraphIOProcessor::createEditor()    { return nullptr; }
 
-int AudioProcessorGraph::AudioGraphIOProcessor::getNumParameters()                  { return 0; }
-const String AudioProcessorGraph::AudioGraphIOProcessor::getParameterName (int)     { return String(); }
-
-float AudioProcessorGraph::AudioGraphIOProcessor::getParameter (int)                { return 0.0f; }
-const String AudioProcessorGraph::AudioGraphIOProcessor::getParameterText (int)     { return String(); }
-void AudioProcessorGraph::AudioGraphIOProcessor::setParameter (int, float)          { }
-
 int AudioProcessorGraph::AudioGraphIOProcessor::getNumPrograms()                    { return 0; }
 int AudioProcessorGraph::AudioGraphIOProcessor::getCurrentProgram()                 { return 0; }
 void AudioProcessorGraph::AudioGraphIOProcessor::setCurrentProgram (int)            { }
@@ -1522,7 +1502,7 @@ void AudioProcessorGraph::AudioGraphIOProcessor::setParentGraph (AudioProcessorG
     if (graph != nullptr)
     {
         setPlayConfigDetails (type == audioOutputNode ? graph->getNumOutputChannels() : 0,
-                              type == audioInputNode ? graph->getNumInputChannels() : 0,
+                              type == audioInputNode  ? graph->getNumInputChannels()  : 0,
                               getSampleRate(),
                               getBlockSize());
 

source/modules/juce_audio_processors/processors/juce_AudioProcessorGraph.h

@@ -80,7 +80,7 @@ public:
 
         //==============================================================================
         /** A convenient typedef for referring to a pointer to a node object. */
-        typedef ReferenceCountedObjectPtr <Node> Ptr;
+        typedef ReferenceCountedObjectPtr<Node> Ptr;
 
     private:
         //==============================================================================
@@ -151,13 +151,13 @@ public:
     void clear();
 
     /** Returns the number of nodes in the graph. */
-    int getNumNodes() const                                         { return nodes.size(); }
+    int getNumNodes() const noexcept                                { return nodes.size(); }
 
     /** Returns a pointer to one of the nodes in the graph.
         This will return nullptr if the index is out of range.
         @see getNodeForId
     */
-    Node* getNode (const int index) const                           { return nodes [index]; }
+    Node* getNode (const int index) const noexcept                  { return nodes [index]; }
 
     /** Searches the graph for a node with the given ID number and returns it.
         If no such node was found, this returns nullptr.
@@ -289,54 +289,47 @@ public:
 
         //==============================================================================
         /** Returns the mode of this processor. */
-        IODeviceType getType() const                                { return type; }
+        IODeviceType getType() const noexcept                       { return type; }
 
         /** Returns the parent graph to which this processor belongs, or nullptr if it
             hasn't yet been added to one. */
-        AudioProcessorGraph* getParentGraph() const                 { return graph; }
+        AudioProcessorGraph* getParentGraph() const noexcept        { return graph; }
 
         /** True if this is an audio or midi input. */
-        bool isInput() const;
+        bool isInput() const noexcept;
         /** True if this is an audio or midi output. */
-        bool isOutput() const;
+        bool isOutput() const noexcept;
 
         //==============================================================================
         AudioGraphIOProcessor (const IODeviceType type);
         ~AudioGraphIOProcessor();
 
-        const String getName() const;
-        void fillInPluginDescription (PluginDescription&) const;
-
-        void prepareToPlay (double sampleRate, int estimatedSamplesPerBlock);
-        void releaseResources();
+        const String getName() const override;
+        void fillInPluginDescription (PluginDescription&) const override;
+        void prepareToPlay (double sampleRate, int estimatedSamplesPerBlock) override;
+        void releaseResources() override;
         void processBlock (AudioSampleBuffer&, MidiBuffer&);
 
-        const String getInputChannelName (int channelIndex) const;
-        const String getOutputChannelName (int channelIndex) const;
-        bool isInputChannelStereoPair (int index) const;
-        bool isOutputChannelStereoPair (int index) const;
-        bool silenceInProducesSilenceOut() const;
-        double getTailLengthSeconds() const;
-        bool acceptsMidi() const;
-        bool producesMidi() const;
-
-        bool hasEditor() const;
-        AudioProcessorEditor* createEditor();
-
-        int getNumParameters();
-        const String getParameterName (int);
-        float getParameter (int);
-        const String getParameterText (int);
-        void setParameter (int, float);
-
-        int getNumPrograms();
-        int getCurrentProgram();
-        void setCurrentProgram (int);
-        const String getProgramName (int);
-        void changeProgramName (int, const String&);
-
-        void getStateInformation (juce::MemoryBlock& destData);
-        void setStateInformation (const void* data, int sizeInBytes);
+        const String getInputChannelName (int channelIndex) const override;
+        const String getOutputChannelName (int channelIndex) const override;
+        bool isInputChannelStereoPair (int index) const override;
+        bool isOutputChannelStereoPair (int index) const override;
+        bool silenceInProducesSilenceOut() const override;
+        double getTailLengthSeconds() const override;
+        bool acceptsMidi() const override;
+        bool producesMidi() const override;
+
+        bool hasEditor() const override;
+        AudioProcessorEditor* createEditor() override;
+
+        int getNumPrograms() override;
+        int getCurrentProgram() override;
+        void setCurrentProgram (int) override;
+        const String getProgramName (int) override;
+        void changeProgramName (int, const String&) override;
+
+        void getStateInformation (juce::MemoryBlock& destData) override;
+        void setStateInformation (const void* data, int sizeInBytes) override;
 
         /** @internal */
         void setParentGraph (AudioProcessorGraph*);
@@ -349,42 +342,34 @@ public:
     };
 
     //==============================================================================
-    // AudioProcessor methods:
-
-    const String getName() const;
-
-    void prepareToPlay (double sampleRate, int estimatedSamplesPerBlock);
-    void releaseResources();
-    void processBlock (AudioSampleBuffer&, MidiBuffer&);
-    void reset();
-
-    const String getInputChannelName (int channelIndex) const;
-    const String getOutputChannelName (int channelIndex) const;
-    bool isInputChannelStereoPair (int index) const;
-    bool isOutputChannelStereoPair (int index) const;
-    bool silenceInProducesSilenceOut() const;
-    double getTailLengthSeconds() const;
-
-    bool acceptsMidi() const;
-    bool producesMidi() const;
-
-    bool hasEditor() const                          { return false; }
-    AudioProcessorEditor* createEditor()            { return nullptr; }
-
-    int getNumParameters()                          { return 0; }
-    const String getParameterName (int)             { return String(); }
-    float getParameter (int)                        { return 0; }
-    const String getParameterText (int)             { return String(); }
-    void setParameter (int, float)                  { }
-
-    int getNumPrograms()                            { return 0; }
-    int getCurrentProgram()                         { return 0; }
-    void setCurrentProgram (int)                    { }
-    const String getProgramName (int)               { return String(); }
-    void changeProgramName (int, const String&)     { }
-
-    void getStateInformation (juce::MemoryBlock&);
-    void setStateInformation (const void* data, int sizeInBytes);
+    const String getName() const override;
+    void prepareToPlay (double, int) override;
+    void releaseResources() override;
+    void processBlock (AudioSampleBuffer&, MidiBuffer&) override;
+
+    void reset() override;
+    void setNonRealtime (bool) noexcept override;
+    void setPlayHead (AudioPlayHead*) override;
+
+    const String getInputChannelName (int) const override;
+    const String getOutputChannelName (int) const override;
+    bool isInputChannelStereoPair (int) const override;
+    bool isOutputChannelStereoPair (int) const override;
+
+    bool silenceInProducesSilenceOut() const override;
+    double getTailLengthSeconds() const override;
+    bool acceptsMidi() const override;
+    bool producesMidi() const override;
+
+    bool hasEditor() const override                         { return false; }
+    AudioProcessorEditor* createEditor() override           { return nullptr; }
+    int getNumPrograms() override                           { return 0; }
+    int getCurrentProgram() override                        { return 0; }
+    void setCurrentProgram (int) override                   { }
+    const String getProgramName (int) override              { return String(); }
+    void changeProgramName (int, const String&) override    { }
+    void getStateInformation (juce::MemoryBlock&) override;
+    void setStateInformation (const void* data, int sizeInBytes) override;
 
 private:
     //==============================================================================

source/modules/juce_audio_processors/processors/juce_GenericAudioProcessorEditor.cpp

@@ -67,7 +67,7 @@ public:
         if (paramHasChanged)
         {
             refresh();
-            startTimer (1000 / 50);
+            startTimerHz (50);
         }
         else
         {

source/modules/juce_core/containers/juce_Array.h

@@ -96,8 +96,7 @@ public:
         @param values   the array to copy from
     */
     template <typename TypeToCreateFrom>
-    explicit Array (const TypeToCreateFrom* values)
-       : numUsed (0)
+    explicit Array (const TypeToCreateFrom* values)  : numUsed (0)
     {
         while (*values != TypeToCreateFrom())
             add (*values++);
@@ -109,8 +108,7 @@ public:
         @param numValues    the number of values in the array
     */
     template <typename TypeToCreateFrom>
-    Array (const TypeToCreateFrom* values, int numValues)
-       : numUsed (numValues)
+    Array (const TypeToCreateFrom* values, int numValues)  : numUsed (numValues)
     {
         data.setAllocatedSize (numValues);
 
@@ -118,6 +116,14 @@ public:
             new (data.elements + i) ElementType (values[i]);
     }
 
+   #if JUCE_COMPILER_SUPPORTS_INITIALIZER_LISTS
+    template <typename TypeToCreateFrom>
+    Array (const std::initializer_list<TypeToCreateFrom>& items)  : numUsed (0)
+    {
+        addArray (items);
+    }
+   #endif
+
     /** Destructor. */
     ~Array()
     {
@@ -474,7 +480,11 @@ public:
             numUsed += numberOfTimesToInsertIt;
 
             while (--numberOfTimesToInsertIt >= 0)
-                new (insertPos++) ElementType (newElement);
+            {
+                new (insertPos) ElementType (newElement);
+                ++insertPos; // NB: this increment is done separately from the
+                             // new statement to avoid a compiler bug in VS2014
+            }
         }
     }
 
@@ -600,6 +610,21 @@ public:
         }
     }
 
+   #if JUCE_COMPILER_SUPPORTS_INITIALIZER_LISTS
+    template <typename TypeToCreateFrom>
+    void addArray (const std::initializer_list<TypeToCreateFrom>& items)
+    {
+        const ScopedLockType lock (getLock());
+        data.ensureAllocatedSize (numUsed + (int) items.size());
+
+        for (auto& item : items)
+        {
+            new (data.elements + numUsed) ElementType (item);
+            ++numUsed;
+        }
+    }
+   #endif
+
     /** Adds elements from a null-terminated array of pointers to the end of this array.
 
         @param elementsToAdd    an array of pointers to some kind of object from which elements

source/modules/juce_core/containers/juce_DynamicObject.cpp

@@ -45,7 +45,7 @@ bool DynamicObject::hasProperty (const Identifier& propertyName) const
     return v != nullptr && ! v->isMethod();
 }
 
-var DynamicObject::getProperty (const Identifier& propertyName) const
+const var& DynamicObject::getProperty (const Identifier& propertyName) const
 {
     return properties [propertyName];
 }

source/modules/juce_core/containers/juce_DynamicObject.h

@@ -60,7 +60,7 @@ public:
     /** Returns a named property.
         This returns var::null if no such property exists.
     */
-    virtual var getProperty (const Identifier& propertyName) const;
+    virtual const var& getProperty (const Identifier& propertyName) const;
 
     /** Sets a named property. */
     virtual void setProperty (const Identifier& propertyName, const var& newValue);

source/modules/juce_core/containers/juce_ReferenceCountedArray.h

@@ -38,7 +38,7 @@
     The template parameter specifies the class of the object you want to point to - the easiest
     way to make a class reference-countable is to simply make it inherit from ReferenceCountedObject
     or SingleThreadedReferenceCountedObject, but if you need to, you can roll your own reference-countable
-    class by implementing a set of mathods called incReferenceCount(), decReferenceCount(), and
+    class by implementing a set of methods called incReferenceCount(), decReferenceCount(), and
     decReferenceCountWithoutDeleting(). See ReferenceCountedObject for examples of how these methods
     should behave.
 

source/modules/juce_core/containers/juce_Variant.cpp

@@ -174,7 +174,7 @@ public:
     int toInt (const ValueUnion& data) const noexcept override       { return (int) data.doubleValue; };
     int64 toInt64 (const ValueUnion& data) const noexcept override   { return (int64) data.doubleValue; };
     double toDouble (const ValueUnion& data) const noexcept override { return data.doubleValue; }
-    String toString (const ValueUnion& data) const override          { return String (data.doubleValue); }
+    String toString (const ValueUnion& data) const override          { return String (data.doubleValue, 20); }
     bool toBool (const ValueUnion& data) const noexcept override     { return data.doubleValue != 0; }
     bool isDouble() const noexcept override                          { return true; }
 
@@ -253,8 +253,8 @@ public:
     }
 
 private:
-    static inline const String* getString (const ValueUnion& data) noexcept { return reinterpret_cast <const String*> (data.stringValue); }
-    static inline String* getString (ValueUnion& data) noexcept             { return reinterpret_cast <String*> (data.stringValue); }
+    static inline const String* getString (const ValueUnion& data) noexcept { return reinterpret_cast<const String*> (data.stringValue); }
+    static inline String* getString (ValueUnion& data) noexcept             { return reinterpret_cast<String*> (data.stringValue); }
 };
 
 //==============================================================================
@@ -487,7 +487,7 @@ var::operator String() const                            { return type->toString
 ReferenceCountedObject* var::getObject() const noexcept { return type->toObject (value); }
 Array<var>* var::getArray() const noexcept              { return type->toArray (value); }
 MemoryBlock* var::getBinaryData() const noexcept        { return type->toBinary (value); }
-DynamicObject* var::getDynamicObject() const noexcept   { return dynamic_cast <DynamicObject*> (getObject()); }
+DynamicObject* var::getDynamicObject() const noexcept   { return dynamic_cast<DynamicObject*> (getObject()); }
 
 //==============================================================================
 void var::swapWith (var& other) noexcept
@@ -576,15 +576,15 @@ var var::clone() const noexcept
 }
 
 //==============================================================================
-var var::operator[] (const Identifier propertyName) const
+const var& var::operator[] (Identifier propertyName) const
 {
     if (DynamicObject* const o = getDynamicObject())
         return o->getProperty (propertyName);
 
-    return var();
+    return var::null;
 }
 
-var var::operator[] (const char* const propertyName) const
+const var& var::operator[] (const char* const propertyName) const
 {
     return operator[] (Identifier (propertyName));
 }