Fix conversion warnings for the Audio module

include/SFML/Audio/SoundStream.hpp

@@ -327,7 +327,7 @@ class SFML_AUDIO_API SoundStream : public SoundSource
     unsigned int  m_buffers[BufferCount];     //!< Sound buffers used to store temporary audio data
     unsigned int  m_channelCount;             //!< Number of channels (1 = mono, 2 = stereo, ...)
     unsigned int  m_sampleRate;               //!< Frequency (samples / second)
-    Uint32        m_format;                   //!< Format of the internal sound buffers
+    Int32         m_format;                   //!< Format of the internal sound buffers
     bool          m_loop;                     //!< Loop flag (true to loop, false to play once)
     Uint64        m_samplesProcessed;         //!< Number of samples processed since beginning of the stream
     Int64         m_bufferSeeks[BufferCount]; //!< If buffer is an "end buffer", holds next seek position, else NoLoop. For play offset calculation.

src/SFML/Audio/InputSoundFile.cpp

@@ -202,7 +202,7 @@ Time InputSoundFile::getDuration() const
     if (m_channelCount == 0 || m_sampleRate == 0)
         return Time::Zero;
 
-    return seconds(static_cast<float>(m_sampleCount) / m_channelCount / m_sampleRate);
+    return seconds(static_cast<float>(m_sampleCount) / static_cast<float>(m_channelCount) / static_cast<float>(m_sampleRate));
 }
 
 
@@ -213,7 +213,7 @@ Time InputSoundFile::getTimeOffset() const
     if (m_channelCount == 0 || m_sampleRate == 0)
         return Time::Zero;
 
-    return seconds(static_cast<float>(m_sampleOffset) / m_channelCount / m_sampleRate);
+    return seconds(static_cast<float>(m_sampleOffset) / static_cast<float>(m_channelCount) / static_cast<float>(m_sampleRate));
 }
 
 
@@ -240,7 +240,7 @@ void InputSoundFile::seek(Uint64 sampleOffset)
 ////////////////////////////////////////////////////////////
 void InputSoundFile::seek(Time timeOffset)
 {
-    seek(static_cast<Uint64>(timeOffset.asSeconds() * m_sampleRate) * m_channelCount);
+    seek(static_cast<Uint64>(timeOffset.asSeconds() * static_cast<float>(m_sampleRate)) * m_channelCount);
 }
 
 

src/SFML/Audio/Music.cpp

@@ -220,7 +220,7 @@ Int64 Music::onLoop()
         // Looping is enabled, and either we're at the loop end, or we're at the EOF
         // when it's equivalent to the loop end (loop end takes priority). Send us to loop begin
         m_file.seek(m_loopSpan.offset);
-        return m_file.getSampleOffset();
+        return static_cast<Int64>(m_file.getSampleOffset());
     }
     else if (getLoop() && (currentOffset >= m_file.getSampleCount()))
     {
@@ -253,7 +253,7 @@ Uint64 Music::timeToSamples(Time position) const
     // This avoids most precision errors arising from "samples => Time => samples" conversions
     // Original rounding calculation is ((Micros * Freq * Channels) / 1000000) + 0.5
     // We refactor it to keep Int64 as the data type throughout the whole operation.
-    return ((position.asMicroseconds() * getSampleRate() * getChannelCount()) + 500000) / 1000000;
+    return ((static_cast<Uint64>(position.asMicroseconds()) * getSampleRate() * getChannelCount()) + 500000) / 1000000;
 }
 
 
@@ -264,7 +264,7 @@ Time Music::samplesToTime(Uint64 samples) const
 
     // Make sure we don't divide by 0
     if (getSampleRate() != 0 && getChannelCount() != 0)
-        position = microseconds((samples * 1000000) / (getChannelCount() * getSampleRate()));
+        position = microseconds(static_cast<Int64>((samples * 1000000) / (getChannelCount() * getSampleRate())));
 
     return position;
 }

src/SFML/Audio/Sound.cpp

@@ -101,7 +101,7 @@ void Sound::setBuffer(const SoundBuffer& buffer)
     // Assign and use the new buffer
     m_buffer = &buffer;
     m_buffer->attachSound(this);
-    alCheck(alSourcei(m_source, AL_BUFFER, m_buffer->m_buffer));
+    alCheck(alSourcei(m_source, AL_BUFFER, static_cast<ALint>(m_buffer->m_buffer)));
 }
 
 

src/SFML/Audio/SoundBuffer.cpp

@@ -179,7 +179,7 @@ unsigned int SoundBuffer::getSampleRate() const
     ALint sampleRate;
     alCheck(alGetBufferi(m_buffer, AL_FREQUENCY, &sampleRate));
 
-    return sampleRate;
+    return static_cast<unsigned int>(sampleRate);
 }
 
 
@@ -189,7 +189,7 @@ unsigned int SoundBuffer::getChannelCount() const
     ALint channelCount;
     alCheck(alGetBufferi(m_buffer, AL_CHANNELS, &channelCount));
 
-    return channelCount;
+    return static_cast<unsigned int>(channelCount);
 }
 
 
@@ -261,11 +261,11 @@ bool SoundBuffer::update(unsigned int channelCount, unsigned int sampleRate)
         (*it)->resetBuffer();
 
     // Fill the buffer
-    ALsizei size = static_cast<ALsizei>(m_samples.size()) * sizeof(Int16);
-    alCheck(alBufferData(m_buffer, format, &m_samples[0], size, sampleRate));
+    ALsizei size = static_cast<ALsizei>(m_samples.size() * sizeof(Int16));
+    alCheck(alBufferData(m_buffer, format, &m_samples[0], size, static_cast<ALsizei>(sampleRate)));
 
     // Compute the duration
-    m_duration = seconds(static_cast<float>(m_samples.size()) / sampleRate / channelCount);
+    m_duration = seconds(static_cast<float>(m_samples.size()) / static_cast<float>(sampleRate) / static_cast<float>(channelCount));
 
     // Now reattach the buffer to the sounds that use it
     for (SoundList::const_iterator it = sounds.begin(); it != sounds.end(); ++it)

src/SFML/Audio/SoundFileReaderFlac.cpp

@@ -37,7 +37,7 @@ namespace
     {
         sf::priv::SoundFileReaderFlac::ClientData* data = static_cast<sf::priv::SoundFileReaderFlac::ClientData*>(clientData);
 
-        sf::Int64 count = data->stream->read(buffer, *bytes);
+        sf::Int64 count = data->stream->read(buffer, static_cast<sf::Int64>(*bytes));
         if (count > 0)
         {
             *bytes = static_cast<std::size_t>(count);
@@ -57,7 +57,7 @@ namespace
     {
         sf::priv::SoundFileReaderFlac::ClientData* data = static_cast<sf::priv::SoundFileReaderFlac::ClientData*>(clientData);
 
-        sf::Int64 position = data->stream->seek(absoluteByteOffset);
+        sf::Int64 position = data->stream->seek(static_cast<sf::Int64>(absoluteByteOffset));
         if (position >= 0)
             return FLAC__STREAM_DECODER_SEEK_STATUS_OK;
         else
@@ -71,7 +71,7 @@ namespace
         sf::Int64 position = data->stream->tell();
         if (position >= 0)
         {
-            *absoluteByteOffset = position;
+            *absoluteByteOffset = static_cast<FLAC__uint64>(position);
             return FLAC__STREAM_DECODER_TELL_STATUS_OK;
         }
         else
@@ -87,7 +87,7 @@ namespace
         sf::Int64 count = data->stream->getSize();
         if (count >= 0)
         {
-            *streamLength = count;
+            *streamLength = static_cast<FLAC__uint64>(count);
             return FLAC__STREAM_DECODER_LENGTH_STATUS_OK;
         }
         else
@@ -290,8 +290,8 @@ Uint64 SoundFileReaderFlac::read(Int16* samples, Uint64 maxCount)
         if (left > maxCount)
         {
             // There are more leftovers than needed
-            std::copy(m_clientData.leftovers.begin(), m_clientData.leftovers.begin() + static_cast<std::size_t>(maxCount), samples);
-            std::vector<Int16> leftovers(m_clientData.leftovers.begin() + static_cast<std::size_t>(maxCount), m_clientData.leftovers.end());
+            std::copy(m_clientData.leftovers.begin(), m_clientData.leftovers.begin() + static_cast<std::vector<Int16>::difference_type>(maxCount), samples);
+            std::vector<Int16> leftovers(m_clientData.leftovers.begin() + static_cast<std::vector<Int16>::difference_type>(maxCount), m_clientData.leftovers.end());
             m_clientData.leftovers.swap(leftovers);
             return maxCount;
         }

src/SFML/Audio/SoundFileReaderOgg.cpp

@@ -38,7 +38,7 @@ namespace
     size_t read(void* ptr, size_t size, size_t nmemb, void* data)
     {
         sf::InputStream* stream = static_cast<sf::InputStream*>(data);
-        return static_cast<std::size_t>(stream->read(ptr, size * nmemb));
+        return static_cast<std::size_t>(stream->read(ptr, static_cast<sf::Int64>(size * nmemb)));
     }
 
     int seek(void* data, ogg_int64_t offset, int whence)
@@ -115,8 +115,8 @@ bool SoundFileReaderOgg::open(InputStream& stream, Info& info)
 
     // Retrieve the music attributes
     vorbis_info* vorbisInfo = ov_info(&m_vorbis, -1);
-    info.channelCount = vorbisInfo->channels;
-    info.sampleRate = vorbisInfo->rate;
+    info.channelCount = static_cast<unsigned int>(vorbisInfo->channels);
+    info.sampleRate = static_cast<unsigned int>(vorbisInfo->rate);
     info.sampleCount = static_cast<std::size_t>(ov_pcm_total(&m_vorbis, -1) * vorbisInfo->channels);
 
     // We must keep the channel count for the seek function
@@ -131,7 +131,7 @@ void SoundFileReaderOgg::seek(Uint64 sampleOffset)
 {
     assert(m_vorbis.datasource);
 
-    ov_pcm_seek(&m_vorbis, sampleOffset / m_channelCount);
+    ov_pcm_seek(&m_vorbis, static_cast<ogg_int64_t>(sampleOffset / m_channelCount));
 }
 
 
@@ -144,12 +144,12 @@ Uint64 SoundFileReaderOgg::read(Int16* samples, Uint64 maxCount)
     Uint64 count = 0;
     while (count < maxCount)
     {
-        int bytesToRead = static_cast<int>(maxCount - count) * sizeof(Int16);
+        int bytesToRead = static_cast<int>(maxCount - count) * static_cast<int>(sizeof(Int16));
         long bytesRead = ov_read(&m_vorbis, reinterpret_cast<char*>(samples), bytesToRead, 0, 2, 1, NULL);
         if (bytesRead > 0)
         {
-            long samplesRead = bytesRead / sizeof(Int16);
-            count += samplesRead;
+            long samplesRead = bytesRead / static_cast<long>(sizeof(Int16));
+            count += static_cast<Uint64>(samplesRead);
             samples += samplesRead;
         }
         else

src/SFML/Audio/SoundFileReaderWav.cpp

@@ -50,7 +50,7 @@ namespace
         if (stream.read(bytes, sizeof(bytes)) != sizeof(bytes))
             return false;
 
-        value = bytes[0] | (bytes[1] << 8);
+        value = static_cast<sf::Uint8>(bytes[0] | (bytes[1] << 8));
 
         return true;
     }
@@ -61,7 +61,7 @@ namespace
         if (stream.read(bytes, sizeof(bytes)) != sizeof(bytes))
             return false;
 
-        value = bytes[0] | (bytes[1] << 8);
+        value = static_cast<sf::Uint16>(bytes[0] | (bytes[1] << 8));
 
         return true;
     }
@@ -72,7 +72,7 @@ namespace
         if (stream.read(bytes, sizeof(bytes)) != sizeof(bytes))
             return false;
 
-        value = bytes[0] | (bytes[1] << 8) | (bytes[2] << 16);
+        value = static_cast<sf::Uint32>(bytes[0] | (bytes[1] << 8) | (bytes[2] << 16));
 
         return true;
     }
@@ -83,7 +83,7 @@ namespace
         if (stream.read(bytes, sizeof(bytes)) != sizeof(bytes))
             return false;
 
-        value = bytes[0] | (bytes[1] << 8) | (bytes[2] << 16) | (bytes[3] << 24);
+        value = static_cast<sf::Uint32>(bytes[0] | (bytes[1] << 8) | (bytes[2] << 16) | (bytes[3] << 24));
 
         return true;
     }
@@ -145,7 +145,7 @@ void SoundFileReaderWav::seek(Uint64 sampleOffset)
 {
     assert(m_stream);
 
-    m_stream->seek(m_dataStart + sampleOffset * m_bytesPerSample);
+    m_stream->seek(static_cast<Int64>(m_dataStart + sampleOffset * m_bytesPerSample));
 }
 
 
@@ -155,7 +155,7 @@ Uint64 SoundFileReaderWav::read(Int16* samples, Uint64 maxCount)
     assert(m_stream);
 
     Uint64 count = 0;
-    Uint64 startPos = m_stream->tell();
+    Uint64 startPos = static_cast<Uint64>(m_stream->tell());
 
     // Tracking of m_dataEnd is important to prevent sf::Music from reading
     // data until EOF, as WAV files may have metadata at the end.
@@ -167,7 +167,7 @@ Uint64 SoundFileReaderWav::read(Int16* samples, Uint64 maxCount)
             {
                 Uint8 sample = 0;
                 if (decode(*m_stream, sample))
-                    *samples++ = (static_cast<Int16>(sample) - 128) << 8;
+                    *samples++ = static_cast<Int16>((static_cast<Int16>(sample) - 128) << 8);
                 else
                     return count;
                 break;
@@ -336,7 +336,7 @@ bool SoundFileReaderWav::parseHeader(Info& info)
             info.sampleCount = subChunkSize / m_bytesPerSample;
 
             // Store the start and end position of samples in the file
-            m_dataStart = subChunkStart;
+            m_dataStart = static_cast<Uint64>(subChunkStart);
             m_dataEnd = m_dataStart + info.sampleCount * m_bytesPerSample;
 
             dataChunkFound = true;

src/SFML/Audio/SoundFileWriterWav.cpp

@@ -162,7 +162,7 @@ bool SoundFileWriterWav::writeHeader(unsigned int sampleRate, unsigned int chann
 
     // Write the sound attributes
     encode(m_file, static_cast<Uint16>(channelCount));
-    encode(m_file, static_cast<Uint32>(sampleRate));
+    encode(m_file, sampleRate);
     Uint32 byteRate = sampleRate * channelCount * 2;
     encode(m_file, byteRate);
     Uint16 blockAlign = static_cast<Uint16>(channelCount * 2);

src/SFML/Audio/SoundRecorder.cpp

@@ -91,7 +91,7 @@ bool SoundRecorder::start(unsigned int sampleRate)
     ALCenum format = (m_channelCount == 1) ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16;
 
     // Open the capture device for capturing 16 bits samples
-    captureDevice = alcCaptureOpenDevice(m_deviceName.c_str(), sampleRate, format, sampleRate);
+    captureDevice = alcCaptureOpenDevice(m_deviceName.c_str(), sampleRate, format, static_cast<ALCsizei>(sampleRate));
     if (!captureDevice)
     {
         err() << "Failed to open the audio capture device with the name: " << m_deviceName << std::endl;
@@ -188,7 +188,7 @@ bool SoundRecorder::setDevice(const std::string& name)
         ALCenum format = (m_channelCount == 1) ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16;
 
         // Open the requested capture device for capturing 16 bits samples
-        captureDevice = alcCaptureOpenDevice(m_deviceName.c_str(), m_sampleRate, format, m_sampleRate);
+        captureDevice = alcCaptureOpenDevice(m_deviceName.c_str(), m_sampleRate, format, static_cast<ALCsizei>(m_sampleRate));
         if (!captureDevice)
         {
             // Notify derived class
@@ -300,7 +300,7 @@ void SoundRecorder::processCapturedSamples()
     if (samplesAvailable > 0)
     {
         // Get the recorded samples
-        m_samples.resize(samplesAvailable * getChannelCount());
+        m_samples.resize(static_cast<unsigned int>(samplesAvailable) * getChannelCount());
         alcCaptureSamples(captureDevice, &m_samples[0], samplesAvailable);
 
         // Forward them to the derived class

src/SFML/Audio/SoundStream.cpp

@@ -216,7 +216,7 @@ void SoundStream::setPlayingOffset(Time timeOffset)
     onSeek(timeOffset);
 
     // Restart streaming
-    m_samplesProcessed = static_cast<Uint64>(timeOffset.asSeconds() * m_sampleRate * m_channelCount);
+    m_samplesProcessed = static_cast<Uint64>(timeOffset.asSeconds() * static_cast<float>(m_sampleRate)) * m_channelCount;
 
     if (oldStatus == Stopped)
         return;
@@ -235,7 +235,7 @@ Time SoundStream::getPlayingOffset() const
         ALfloat secs = 0.f;
         alCheck(alGetSourcef(m_source, AL_SEC_OFFSET, &secs));
 
-        return seconds(secs + static_cast<float>(m_samplesProcessed) / m_sampleRate / m_channelCount);
+        return seconds(secs + static_cast<float>(m_samplesProcessed) / static_cast<float>(m_sampleRate) / static_cast<float>(m_channelCount));
     }
     else
     {
@@ -342,7 +342,7 @@ void SoundStream::streamData()
 
             // Find its number
             unsigned int bufferNum = 0;
-            for (int i = 0; i < BufferCount; ++i)
+            for (unsigned int i = 0; i < BufferCount; ++i)
                 if (m_buffers[i] == buffer)
                 {
                     bufferNum = i;
@@ -353,7 +353,7 @@ void SoundStream::streamData()
             if (m_bufferSeeks[bufferNum] != NoLoop)
             {
                 // This was the last buffer before EOF or Loop End: reset the sample count
-                m_samplesProcessed = m_bufferSeeks[bufferNum];
+                m_samplesProcessed = static_cast<Uint64>(m_bufferSeeks[bufferNum]);
                 m_bufferSeeks[bufferNum] = NoLoop;
             }
             else
@@ -376,7 +376,7 @@ void SoundStream::streamData()
                 }
                 else
                 {
-                    m_samplesProcessed += size / (bits / 8);
+                    m_samplesProcessed += static_cast<Uint64>(size / (bits / 8));
                 }
             }
 
@@ -439,7 +439,7 @@ bool SoundStream::fillAndPushBuffer(unsigned int bufferNum, bool immediateLoop)
         if (immediateLoop && (m_bufferSeeks[bufferNum] != NoLoop))
         {
             // We just tried to begin preloading at EOF or Loop End: reset the sample count
-            m_samplesProcessed = m_bufferSeeks[bufferNum];
+            m_samplesProcessed = static_cast<Uint64>(m_bufferSeeks[bufferNum]);
             m_bufferSeeks[bufferNum] = NoLoop;
         }
 
@@ -452,8 +452,8 @@ bool SoundStream::fillAndPushBuffer(unsigned int bufferNum, bool immediateLoop)
         unsigned int buffer = m_buffers[bufferNum];
 
         // Fill the buffer
-        ALsizei size = static_cast<ALsizei>(data.sampleCount) * sizeof(Int16);
-        alCheck(alBufferData(buffer, m_format, data.samples, size, m_sampleRate));
+        ALsizei size = static_cast<ALsizei>(data.sampleCount * sizeof(Int16));
+        alCheck(alBufferData(buffer, m_format, data.samples, size, static_cast<ALsizei>(m_sampleRate)));
 
         // Push it into the sound queue
         alCheck(alSourceQueueBuffers(m_source, 1, &buffer));
@@ -473,7 +473,7 @@ bool SoundStream::fillQueue()
 {
     // Fill and enqueue all the available buffers
     bool requestStop = false;
-    for (int i = 0; (i < BufferCount) && !requestStop; ++i)
+    for (unsigned int i = 0; (i < BufferCount) && !requestStop; ++i)
     {
         // Since no sound has been loaded yet, we can't schedule loop seeks preemptively,
         // So if we start on EOF or Loop End, we let fillAndPushBuffer() adjust the sample count