A few small bug fixes related to bad pointer arithmetic

src/contrib/m4adecoder/libfaad/common.h

@@ -313,17 +313,17 @@ char *strchr(), *strrchr();
 
 
   #if defined(_WIN32) && !defined(__MINGW32__)
-    #define HAS_LRINTF
-    static INLINE int lrintf(float f)
-    {
-        int i;
-        __asm
-        {
-            fld   f
-            fistp i
-        }
-        return i;
-    }
+    //#define HAS_LRINTF
+    //static INLINE int lrintf(float f)
+    //{
+    //    int i;
+    //    __asm
+    //    {
+    //        fld   f
+    //        fistp i
+    //    }
+    //    return i;
+    //}
   #elif (defined(__i386__) && defined(__GNUC__) && \
 	!defined(__CYGWIN__) && !defined(__MINGW32__))
     #ifndef HAVE_LRINTF

src/core/audio/Buffer.cpp

@@ -57,7 +57,6 @@ Buffer::Buffer(void)
 , internalBufferSize(0)
 , sampleRate(44100)
 , channels(2) {
-
 }
 
 Buffer::~Buffer() {
@@ -105,11 +104,7 @@ void Buffer::CopyFormat(BufferPtr fromBuffer) {
 
 void Buffer::ResizeBuffer() {
     if (this->sampleSize > this->internalBufferSize) {
-        if (this->buffer) {
-            delete[] this->buffer;
-            this->buffer = nullptr;
-        }
-
+        delete[] this->buffer;
         this->buffer = new float[this->sampleSize];
         this->internalBufferSize = this->sampleSize;
     }
@@ -145,42 +140,31 @@ void Buffer::Copy(float* buffer, long samples) {
 
 void Buffer::Append(float* src, long samples) {
     /* number of floats (not bytes) in buffer */
-    long newBufferSize = (this->Samples() + samples);
+    long newBufferSize = this->sampleSize + samples;
 
-    if (newBufferSize > this->internalBufferSize) { /* resize, then copy, if too small */
+    if (newBufferSize > this->internalBufferSize) {
+        /* resize, then copy, if too small */
         float *newBuffer = new float[newBufferSize];
 
         CopyFloat(newBuffer, this->buffer, this->sampleSize);
-        float *dst = &newBuffer[this->sampleSize];
-        CopyFloat(dst, src, samples);
+        CopyFloat(newBuffer + this->sampleSize, src, samples);
 
         delete[] this->buffer;
 
         this->buffer = newBuffer;
         this->internalBufferSize = newBufferSize;
     }
-    else { /* append, no resize required */
-        float *dst = &this->buffer[this->sampleSize];
-        CopyFloat(dst, src, samples);
+    else {
+        /* append, no resize required */
+        CopyFloat(this->buffer + this->sampleSize, src, samples);
         this->internalBufferSize += samples;
     }
 
     this->sampleSize = newBufferSize;
 }
 
-bool Buffer::Append(BufferPtr appendBuffer) {
-    if (this->SampleRate() == appendBuffer->SampleRate() &&
-        this->Channels() == appendBuffer->Channels())
-    {
-        this->Append(appendBuffer->BufferPointer(), appendBuffer->Samples());
-        return true;
-    }
-
-    return false;
-}
-
 bool Buffer::Fft(float* buffer, int size) {
-    if (this->sampleSize < FFT_BUFFER_SIZE || size != FFT_BUFFER_SIZE) {
+    if (this->sampleSize / this->channels < FFT_BUFFER_SIZE || size != FFT_BUFFER_SIZE) {
         return false;
     }
 
@@ -188,8 +172,9 @@ bool Buffer::Fft(float* buffer, int size) {
 
     /* de-interleave the audio first */
     float* deinterleaved = new float[FFT_BUFFER_SIZE * count];
+    int to;
     for (int i = 0; i < count * FFT_BUFFER_SIZE; i++) {
-        const int to = ((i % this->channels) * FFT_BUFFER_SIZE) + (i / count);
+        to = ((i % this->channels) * FFT_BUFFER_SIZE) + (i / count);
         deinterleaved[to] = this->buffer[i];
     }
     
@@ -207,7 +192,7 @@ bool Buffer::Fft(float* buffer, int size) {
     fft_perform(this->buffer, buffer, state);
     
     for (int i = 1; i < count; i++) {
-        fft_perform(&deinterleaved[i * FFT_BUFFER_SIZE], scratch, state);
+        fft_perform(deinterleaved + (i * FFT_BUFFER_SIZE), scratch, state);
 
         /* average with the previous read */
         for (int j = 0; j < FFT_BUFFER_SIZE; j++) {

src/core/audio/Buffer.h

@@ -67,7 +67,6 @@ namespace musik { namespace core { namespace audio {
             void SetPosition(double position);
             void Copy(float* buffer, long samples);
             void Append(float* buffer, long samples);
-            bool Append(BufferPtr appendBuffer);
             void CopyFormat(BufferPtr fromBuffer);
 
         private:

src/core/audio/DynamicStream.cpp

@@ -46,12 +46,11 @@ using musik::core::PluginFactory;
 static std::string TAG = "DynamicStream";
 
 DynamicStream::DynamicStream(unsigned int options)
-    : preferedBufferSampleSize(4096)
-    , options(options)
-    , decoderSampleRate(0)
-    , decoderChannels(0)
-    , decoderSamplePosition(0)
-{
+: preferedBufferSampleSize(4096)
+, options(options)
+, decoderSampleRate(0)
+, decoderChannels(0)
+, decoderSamplePosition(0) {
     if ((this->options & NoDSP) == 0) {
         typedef PluginFactory::DestroyDeleter<IDSP> Deleter;
         this->dsps = PluginFactory::Instance().QueryInterface<IDSP, Deleter>("GetDSP");
@@ -164,13 +163,18 @@ BufferPtr DynamicStream::GetNextProcessedOutputBuffer() {
     BufferPtr currentBuffer = this->GetNextBufferFromDecoder();
 
     if (currentBuffer) {
+        void * f = &free;
+
         /* try to fill the buffer to its optimal size; if the decoder didn't return
         a full buffer, ask it for some more data. */
         bool moreBuffers = true;
         while (currentBuffer->Samples() < this->preferedBufferSampleSize && moreBuffers) {
             BufferPtr bufferToAppend = this->GetNextBufferFromDecoder();
             if (bufferToAppend) {
-                currentBuffer->Append(bufferToAppend);
+                currentBuffer->Append(
+                    bufferToAppend->BufferPointer(),
+                    bufferToAppend->Samples());
+
                 this->RecycleBuffer(bufferToAppend);
             }
             else {
@@ -184,7 +188,7 @@ BufferPtr DynamicStream::GetNextProcessedOutputBuffer() {
 
             for (Dsps::iterator dsp = this->dsps.begin(); dsp != this->dsps.end(); ++dsp) {
                 oldBuffer->CopyFormat(currentBuffer);
-                currentBuffer->SetPosition(oldBuffer->Position());
+                oldBuffer->SetPosition(currentBuffer->Position());
 
                 if ((*dsp)->Process(currentBuffer.get(), oldBuffer.get())) {
                     currentBuffer.swap(oldBuffer);

src/core/audio/FixedSizeStream.cpp

@@ -52,7 +52,7 @@ static std::string TAG = "FixedSizeStream";
         ((double) this->decoderSampleRate)); 
 
 #define COPY_BUFFER(target, current, count, offset) \
-    target->Copy(&current->BufferPointer()[offset], count); \
+    target->Copy(current->BufferPointer() + offset, count); \
     SET_OFFSET(target, offset) \
 
 FixedSizeStream::FixedSizeStream(int samplesPerChannel, int bufferCount, unsigned int options)

src/core/audio/Player.cpp

@@ -138,8 +138,12 @@ int Player::State() {
 }
 
 void Player::ThreadLoop() {
-    /* create and open the stream */
-    this->stream = FixedSizeStream::Create();
+    /* fixed size streams are better for visualizations because they provide
+    consistent buffer sizes. dynamic streams are more efficient otherwise 
+    because we don't need to worry about manually chunking data, we can just
+    send audio data to the output straight from the decoder. */
+    this->stream = vis::SelectedVisualizer()
+        ? FixedSizeStream::Create() : DynamicStream::Create();
 
     BufferPtr buffer;
 

src/core/library/Indexer.cpp

@@ -84,6 +84,7 @@ Indexer::Indexer(const std::string& libraryPath, const std::string& dbFilename)
 : thread(nullptr)
 , restart(false)
 , filesSaved(0)
+, exit(false)
 , prefs(Preferences::ForComponent(prefs::components::Settings))
 , readSemaphore(prefs->GetInt(prefs::keys::MaxTagReadThreads, MAX_THREADS)) {
     this->dbFilename = dbFilename;