Changed type of bufferSizes to std::size_t and removed the i prefix o…

…f the name at many related parts of the code

.codespellignorelines

@@ -4,24 +4,24 @@
                  pInOut, pInOut,
         CSAMPLE* pInOut,
     CSAMPLE* pInOut,
-void EnginePregain::process(CSAMPLE* pInOut, const int iBufferSize) {
-void EngineDelay::process(CSAMPLE* pInOut, const int iBufferSize) {
+void EnginePregain::process(CSAMPLE* pInOut, const size_t bufferSize) {
+void EngineDelay::process(CSAMPLE* pInOut, const size_t bufferSize) {
         pInOut[i] = (CSAMPLE) processSample(fbuf1, (double) pInOut[i]);
         pInOut[i + 1] = (CSAMPLE) processSample(fbuf2, (double) pInOut[i + 1]);
             m_pDelayBuffer[m_iDelayPos] = pInOut[i];
             pInOut[i] = m_pDelayBuffer[iDelaySourcePos];
-        SampleUtil::applyRampingGain(&pInOut[0], m_fPrevGain, 0, iBufferSize / 2);
-        SampleUtil::applyRampingGain(&pInOut[iBufferSize / 2], 0, totalGain, iBufferSize / 2);
-        SampleUtil::applyRampingGain(pInOut, m_fPrevGain, totalGain, iBufferSize);
-        SampleUtil::applyGain(pInOut, totalGain, iBufferSize);
-    void process(CSAMPLE* pInOut, const int iBufferSize) override;
-    void process(CSAMPLE* pInOut, const int iBufferSize);
-    virtual void process(CSAMPLE* pInOut, const int iBufferSize);
+        SampleUtil::applyRampingGain(&pInOut[0], m_fPrevGain, 0, bufferSize / 2);
+        SampleUtil::applyRampingGain(&pInOut[bufferSize / 2], 0, totalGain, bufferSize / 2);
+        SampleUtil::applyRampingGain(pInOut, m_fPrevGain, totalGain, bufferSize);
+        SampleUtil::applyGain(pInOut, totalGain, bufferSize);
+    void process(CSAMPLE* pInOut, const size_t bufferSize) override;
+    void process(CSAMPLE* pInOut, const size_t bufferSize);
+    virtual void process(CSAMPLE* pInOut, const size_t bufferSize);
                 "CSAMPLE_GAIN gain%(i)din, CSAMPLE_GAIN gain%(i)dout"
                 "gain%(i)dout == CSAMPLE_GAIN_ZERO) {"
                     "pSrc%(i)d, gain%(i)din, gain%(i)dout" % {"i": j}
                 "(gain%(i)dout - gain%(i)din) / (iNumSamples / 2);"
-    MOCK_METHOD(void, process, (CSAMPLE * pInOut, const int iBufferSize), (override));
+    MOCK_METHOD(void, process, (CSAMPLE * pInOut, const size_t bufferSize), (override));
     const QString kName3 = QStringLiteral("I'm blue, da ba dee");
     float m_k2vg; // IIF factor
     float m_k2vgNew; // IIF factor
@@ -71,6 +71,6 @@ void EngineEffectsDelay::process(CSAMPLE* pInOut,
     // Source: FIPS 180-4 Secure Hash Standard (SHS)
         // ALAC/CAF has been added in version 1.0.26
         QStringLiteral("caf"),
-void EngineFilter::process(CSAMPLE* pInOut, const int iBufferSize)
+void EngineFilter::process(CSAMPLE* pInOut, const size_t bufferSize)
     // Note(RRyan/Max Linke):
     // https://github.com/codders/libshout/blob/a17fb84671d3732317b0353d7281cc47e2df6cf6/src/timing/timing.c

src/encoder/encoder.h

@@ -29,7 +29,7 @@ class Encoder {
 
     virtual int initEncoder(mixxx::audio::SampleRate sampleRate, QString* pUserErrorMessage) = 0;
     // encodes the provided buffer of audio.
-    virtual void encodeBuffer(const CSAMPLE *samples, const int size) = 0;
+    virtual void encodeBuffer(const CSAMPLE* samples, const std::size_t bufferSize) = 0;
     // Adds metadata to the encoded audio, i.e., the ID3 tag. Currently only used
     // by EngineRecord, ShoutConnection does something different.
     virtual void updateMetaData(const QString& artist, const QString& title, const QString& album) = 0;

src/encoder/encoderfdkaac.cpp

@@ -351,11 +351,11 @@ int EncoderFdkAac::initEncoder(mixxx::audio::SampleRate sampleRate, QString* pUs
     return 0;
 }
 
-void EncoderFdkAac::encodeBuffer(const CSAMPLE* samples, const int sampleCount) {
+void EncoderFdkAac::encodeBuffer(const CSAMPLE* samples, const std::size_t bufferSize) {
     if (!m_pInputFifo) {
         return;
     }
-    int writeCount = sampleCount;
+    int writeCount = static_cast<int>(bufferSize);
     int writeAvailable = m_pInputFifo->writeAvailable();
     if (writeCount > writeAvailable) {
         kLogger.warning() << "FIFO buffer too small, losing samples!"

src/encoder/encoderfdkaac.h

@@ -15,7 +15,7 @@ class EncoderFdkAac : public Encoder {
     virtual ~EncoderFdkAac();
 
     int initEncoder(mixxx::audio::SampleRate sampleRate, QString* pUserErrorMessage) override;
-    void encodeBuffer(const CSAMPLE* samples, const int sampleCount) override;
+    void encodeBuffer(const CSAMPLE* samples, const std::size_t bufferSize) override;
     void updateMetaData(const QString& artist, const QString& title, const QString& album) override;
     void flush() override;
     void setEncoderSettings(const EncoderSettings& settings) override;

src/encoder/encoderffmpegcore.cpp

@@ -99,7 +99,7 @@ int EncoderFfmpegCore::getSerial() {
     return l_iSerial;
 }
 
-void EncoderFfmpegCore::encodeBuffer(const CSAMPLE *samples, const int size) {
+void EncoderFfmpegCore::encodeBuffer(const CSAMPLE* samples, const std::size_t bufferSize) {
     unsigned char *l_strBuffer = NULL;
     int l_iBufferLen = 0;
     //int l_iAudioCpyLen = m_iAudioInputFrameSize *

src/encoder/encoderffmpegcore.h

@@ -43,7 +43,7 @@ class EncoderFfmpegCore : public Encoder {
 #endif
     ~EncoderFfmpegCore();
     int initEncoder(mixxx::audio::SampleRate sampleRate, QString* pUserErrorMessage) override;
-    void encodeBuffer(const CSAMPLE *samples, const int size) override;
+    void encodeBuffer(const CSAMPLE* samples, const std::size_t bufferSize) override;
     void updateMetaData(const QString& artist, const QString& title, const QString& album) override;
     void flush() override;
     void setEncoderSettings(const EncoderSettings& settings) override;

src/encoder/encodermp3.cpp

@@ -81,32 +81,32 @@ void EncoderMp3::setEncoderSettings(const EncoderSettings& settings) {
     }
 }
 
-int EncoderMp3::bufferOutGrow(int size) {
-    if (m_bufferOutSize >= size) {
+int EncoderMp3::bufferOutGrow(std::size_t bufferSsize) {
+    if (m_bufferOutSize >= bufferSsize) {
         return 0;
     }
 
-    m_bufferOut = (unsigned char *)realloc(m_bufferOut, size);
+    m_bufferOut = (unsigned char*)realloc(m_bufferOut, bufferSsize);
     if (m_bufferOut == nullptr) {
         return -1;
     }
 
-    m_bufferOutSize = size;
+    m_bufferOutSize = bufferSsize;
     return 0;
 }
 
-int EncoderMp3::bufferInGrow(int size) {
-    if (m_bufferInSize >= size) {
+int EncoderMp3::bufferInGrow(std::size_t bufferSsize) {
+    if (m_bufferInSize >= bufferSsize) {
         return 0;
     }
 
-    m_bufferIn[0] = (float *)realloc(m_bufferIn[0], size * sizeof(float));
-    m_bufferIn[1] = (float *)realloc(m_bufferIn[1], size * sizeof(float));
+    m_bufferIn[0] = (float*)realloc(m_bufferIn[0], bufferSsize * sizeof(float));
+    m_bufferIn[1] = (float*)realloc(m_bufferIn[1], bufferSsize * sizeof(float));
     if ((m_bufferIn[0] == nullptr) || (m_bufferIn[1] == nullptr)) {
         return -1;
     }
 
-    m_bufferInSize = size;
+    m_bufferInSize = bufferSsize;
     return 0;
 }
 
@@ -117,7 +117,7 @@ void EncoderMp3::flush() {
         return;
     }
     // Flush also writes ID3 tags.
-    int rc = lame_encode_flush(m_lameFlags, m_bufferOut, m_bufferOutSize);
+    int rc = lame_encode_flush(m_lameFlags, m_bufferOut, static_cast<int>(m_bufferOutSize));
     if (rc < 0) {
         return;
     }
@@ -127,39 +127,42 @@ void EncoderMp3::flush() {
     // `lame_get_lametag_frame` returns the number of bytes copied into buffer,
     // or the required buffer size, if the provided buffer is too small.
     // Function failed, if the return value is larger than `m_bufferOutSize`!
-    int numBytes = static_cast<int>(
-            lame_get_lametag_frame(m_lameFlags, m_bufferOut, m_bufferOutSize));
+    std::size_t numBytes =
+            lame_get_lametag_frame(m_lameFlags, m_bufferOut, m_bufferOutSize);
     if (numBytes > m_bufferOutSize) {
         bufferOutGrow(numBytes);
-        numBytes = static_cast<int>(lame_get_lametag_frame(
-                m_lameFlags, m_bufferOut, m_bufferOutSize));
+        numBytes = lame_get_lametag_frame(
+                m_lameFlags, m_bufferOut, m_bufferOutSize);
     }
     // Write the lame/xing header.
     m_pCallback->seek(0);
-    m_pCallback->write(nullptr, m_bufferOut, 0, numBytes);
+    m_pCallback->write(nullptr, m_bufferOut, 0, static_cast<int>(numBytes));
 }
 
-void EncoderMp3::encodeBuffer(const CSAMPLE *samples, const int size) {
+void EncoderMp3::encodeBuffer(const CSAMPLE* samples, const std::size_t bufferSize) {
     if (m_lameFlags == nullptr) {
         return;
     }
-    int outsize = 0;
+    std::size_t outsize = 0;
     int rc = 0;
 
-    outsize = (int)((1.25 * size + 7200) + 1);
+    outsize = (int)((1.25 * bufferSize + 7200) + 1);
     bufferOutGrow(outsize);
 
-    bufferInGrow(size);
+    bufferInGrow(bufferSize);
 
     // Deinterleave samples. We use normalized floats in the engine [-1.0, 1.0]
     // but LAME expects samples in the range [SHRT_MIN, SHRT_MAX].
-    for (int i = 0; i < size/2; ++i) {
+    for (std::size_t i = 0; i < bufferSize / 2; ++i) {
         m_bufferIn[0][i] = samples[i*2] * SHRT_MAX;
         m_bufferIn[1][i] = samples[i*2+1] * SHRT_MAX;
     }
-
-    rc = lame_encode_buffer_float(m_lameFlags, m_bufferIn[0], m_bufferIn[1],
-                                  size/2, m_bufferOut, m_bufferOutSize);
+    rc = lame_encode_buffer_float(m_lameFlags,
+            m_bufferIn[0],
+            m_bufferIn[1],
+            static_cast<int>(bufferSize / 2),
+            m_bufferOut,
+            static_cast<int>(m_bufferOutSize));
     if (rc < 0) {
         return;
     }

src/encoder/encodermp3.h

@@ -15,15 +15,15 @@ class EncoderMp3 final : public Encoder {
     ~EncoderMp3() override;
 
     int initEncoder(mixxx::audio::SampleRate sampleRate, QString* pUserErrorMessage) override;
-    void encodeBuffer(const CSAMPLE *samples, const int size) override;
+    void encodeBuffer(const CSAMPLE* samples, const std::size_t bufferSize) override;
     void updateMetaData(const QString& artist, const QString& title, const QString& album) override;
     void flush() override;
     void setEncoderSettings(const EncoderSettings& settings) override;
 
   private:
     void initStream();
-    int bufferOutGrow(int size);
-    int bufferInGrow(int size);
+    int bufferOutGrow(std::size_t bufferSsize);
+    int bufferInGrow(std::size_t bufferSsize);
 
     lame_t m_lameFlags;
     QString m_metaDataTitle;
@@ -35,9 +35,9 @@ class EncoderMp3 final : public Encoder {
     vbr_mode m_encoding_mode;
     MPEG_mode_e m_stereo_mode;
     unsigned char *m_bufferOut;
-    int m_bufferOutSize;
+    std::size_t m_bufferOutSize;
     float* m_bufferIn[2];
-    int m_bufferInSize;
+    std::size_t m_bufferInSize;
 
     EncoderCallback* m_pCallback;
 };

src/encoder/encoderopus.cpp

@@ -368,12 +368,12 @@ void EncoderOpus::pushTagsPacket() {
     }
 }
 
-void EncoderOpus::encodeBuffer(const CSAMPLE *samples, const int size) {
+void EncoderOpus::encodeBuffer(const CSAMPLE* samples, const std::size_t bufferSize) {
     if (!m_pOpus) {
         return;
     }
 
-    int writeRequired = size;
+    int writeRequired = static_cast<int>(bufferSize);
     int writeAvailable = m_fifoBuffer.writeAvailable();
     if (writeRequired > writeAvailable) {
         kLogger.warning() << "FIFO buffer too small, losing samples!"

src/encoder/encoderopus.h

@@ -27,7 +27,7 @@ class EncoderOpus: public Encoder {
     ~EncoderOpus() override;
 
     int initEncoder(mixxx::audio::SampleRate sampleRate, QString* pUserErrorMessage) override;
-    void encodeBuffer(const CSAMPLE *samples, const int size) override;
+    void encodeBuffer(const CSAMPLE* samples, const std::size_t bufferSize) override;
     void updateMetaData(const QString& artist, const QString& title, const QString& album) override;
     void flush() override;
     void setEncoderSettings(const EncoderSettings& settings) override;

src/encoder/encodersndfileflac.cpp

@@ -58,10 +58,10 @@ void EncoderSndfileFlac::setEncoderSettings(const EncoderSettings& settings)
     m_compression = static_cast<double>(settings.getCompression()) / 8.0;
 }
 
-void EncoderSndfileFlac::encodeBuffer(const CSAMPLE* pBuffer, const int iBufferSize) {
+void EncoderSndfileFlac::encodeBuffer(const CSAMPLE* pBuffer, const std::size_t bufferSize) {
     if (m_pClampBuffer) {
-        SINT numSamplesLeft = iBufferSize;
-        while (numSamplesLeft > 0) {
+            SINT numSamplesLeft = bufferSize;
+            while (numSamplesLeft > 0) {
             const SINT numSamplesToWrite = math_min(numSamplesLeft, kEncBufferSize);
             convertFloat32ToIntFormat(m_pClampBuffer.get(),
                     pBuffer,
@@ -70,9 +70,9 @@ void EncoderSndfileFlac::encodeBuffer(const CSAMPLE* pBuffer, const int iBufferS
             sf_write_int(m_pSndfile, m_pClampBuffer.get(), numSamplesToWrite);
             pBuffer += numSamplesToWrite;
             numSamplesLeft -= numSamplesToWrite;
-        }
+            }
     } else {
-        sf_write_float(m_pSndfile, pBuffer, iBufferSize);
+            sf_write_float(m_pSndfile, pBuffer, bufferSize);
     }
 }
 

src/encoder/encodersndfileflac.h

@@ -19,7 +19,7 @@ class EncoderSndfileFlac : public EncoderWave {
     ~EncoderSndfileFlac() override = default;
 
     void setEncoderSettings(const EncoderSettings& settings) override;
-    void encodeBuffer(const CSAMPLE* samples, const int size) override;
+    void encodeBuffer(const CSAMPLE* samples, const std::size_t bufferSize) override;
 
   protected:
     void initStream() override;

src/encoder/encodervorbis.cpp

@@ -133,25 +133,25 @@ void EncoderVorbis::writePage() {
     }
 }
 
-void EncoderVorbis::encodeBuffer(const CSAMPLE *samples, const int size) {
-    float **buffer = vorbis_analysis_buffer(&m_vdsp, size);
+void EncoderVorbis::encodeBuffer(const CSAMPLE* samples, const std::size_t bufferSize) {
+    float** buffer = vorbis_analysis_buffer(&m_vdsp, static_cast<int>(bufferSize));
 
     // Deinterleave samples. We use normalized floats in the engine [-1.0, 1.0]
     // and libvorbis expects samples in the range [-1.0, 1.0] so no conversion
     // is required.
     if (m_channels == 2) {
-        for (int i = 0; i < size/2; ++i) {
+        for (std::size_t i = 0; i < bufferSize / 2; ++i) {
             buffer[0][i] = samples[i*2];
             buffer[1][i] = samples[i*2+1];
         }
     }
     else {
-        for (int i = 0; i < size/2; ++i) {
+        for (std::size_t i = 0; i < bufferSize / 2; ++i) {
             buffer[0][i] = (samples[i*2] + samples[i*2+1]) / 2.f;
         }
     }
     /** encodes audio **/
-    vorbis_analysis_wrote(&m_vdsp, size/2);
+    vorbis_analysis_wrote(&m_vdsp, static_cast<int>(bufferSize) / 2);
     /** writes the OGG page and sends it to file or stream **/
     writePage();
 }

src/encoder/encodervorbis.h

@@ -19,7 +19,7 @@ class EncoderVorbis : public Encoder {
     ~EncoderVorbis() override;
 
     int initEncoder(mixxx::audio::SampleRate sampleRate, QString* pUserErrorMessage) override;
-    void encodeBuffer(const CSAMPLE *samples, const int size) override;
+    void encodeBuffer(const CSAMPLE* samples, const std::size_t bufferSize) override;
     void updateMetaData(const QString& artist, const QString& title, const QString& album) override;
     void flush() override;
     void setEncoderSettings(const EncoderSettings& settings) override;

src/encoder/encoderwave.cpp

@@ -136,9 +136,8 @@ void EncoderWave::flush() {
     sf_write_sync(m_pSndfile);
 }
 
-
-void EncoderWave::encodeBuffer(const CSAMPLE *pBuffer, const int iBufferSize) {
-    sf_write_float(m_pSndfile, pBuffer, iBufferSize);
+void EncoderWave::encodeBuffer(const CSAMPLE* pBuffer, const std::size_t bufferSize) {
+    sf_write_float(m_pSndfile, pBuffer, bufferSize);
 }
 
 /* Originally called from enginebroadcast.cpp to update metadata information

src/encoder/encoderwave.h

@@ -21,7 +21,7 @@ class EncoderWave : public Encoder {
     ~EncoderWave() override;
 
     int initEncoder(mixxx::audio::SampleRate sampleRate, QString* pUserErrorMessage) override;
-    void encodeBuffer(const CSAMPLE *samples, const int size) override;
+    void encodeBuffer(const CSAMPLE* samples, const std::size_t bufferSize) override;
     void updateMetaData(const QString& artist, const QString& title, const QString& album) override;
     void flush() override;
     void setEncoderSettings(const EncoderSettings& settings) override;