Update code to use new AudioData interleaving/deinterleaving helper methods

modules/juce_audio_devices/native/juce_android_Audio.cpp

@@ -339,20 +339,12 @@ public:
 
                 jshort* const src = env->GetShortArrayElements (audioBuffer, nullptr);
 
-                for (int chan = 0; chan < inputChannelBuffer.getNumChannels(); ++chan)
-                {
-                    AudioData::Pointer <AudioData::Float32, AudioData::NativeEndian, AudioData::NonInterleaved, AudioData::NonConst> d (inputChannelBuffer.getWritePointer (chan));
-
-                    if (chan < numDeviceInputChannels)
-                    {
-                        AudioData::Pointer <AudioData::Int16, AudioData::NativeEndian, AudioData::Interleaved, AudioData::Const> s (src + chan, numDeviceInputChannels);
-                        d.convertSamples (s, actualBufferSize);
-                    }
-                    else
-                    {
-                        d.clearSamples (actualBufferSize);
-                    }
-                }
+                AudioData::deinterleaveSamples<AudioData::Int16,   AudioData::NativeEndian,
+                                               AudioData::Float32, AudioData::NativeEndian> (reinterpret_cast<const uint16*> (src),
+                                                                                             numDeviceInputChannels,
+                                                                                             inputChannelBuffer.getArrayOfWritePointers(),
+                                                                                             inputChannelBuffer.getNumChannels(),
+                                                                                             actualBufferSize);
 
                 env->ReleaseShortArrayElements (audioBuffer, src, 0);
             }
@@ -382,14 +374,12 @@ public:
 
                 jshort* const dest = env->GetShortArrayElements (audioBuffer, nullptr);
 
-                for (int chan = 0; chan < numDeviceOutputChannels; ++chan)
-                {
-                    AudioData::Pointer <AudioData::Int16, AudioData::NativeEndian, AudioData::Interleaved, AudioData::NonConst> d (dest + chan, numDeviceOutputChannels);
-
-                    const float* const sourceChanData = outputChannelBuffer.getReadPointer (jmin (chan, outputChannelBuffer.getNumChannels() - 1));
-                    AudioData::Pointer <AudioData::Float32, AudioData::NativeEndian, AudioData::NonInterleaved, AudioData::Const> s (sourceChanData);
-                    d.convertSamples (s, actualBufferSize);
-                }
+                AudioData::interleaveSamples<AudioData::Float32, AudioData::NativeEndian,
+                                             AudioData::Int16,   AudioData::NativeEndian> (outputChannelBuffer.getArrayOfReadPointers(),
+                                                                                           outputChannelBuffer.getNumChannels(),
+                                                                                           reinterpret_cast<uint16*> (dest),
+                                                                                           numDeviceOutputChannels,
+                                                                                           actualBufferSize);
 
                 env->ReleaseShortArrayElements (audioBuffer, dest, 0);
                 jint numWritten = env->CallIntMethod (outputDevice, AudioTrack.write, audioBuffer, 0, actualBufferSize * numDeviceOutputChannels);

modules/juce_audio_devices/native/juce_android_Oboe.cpp

@@ -46,28 +46,26 @@ struct OboeAudioIODeviceBufferHelpers<int16>
 
     static void convertFromOboe (const int16* srcInterleaved, AudioBuffer<float>& audioBuffer, int numSamples)
     {
-        for (int i = 0; i < audioBuffer.getNumChannels(); ++i)
-        {
-            using DstSampleType = AudioData::Pointer<AudioData::Float32, AudioData::NativeEndian, AudioData::NonInterleaved, AudioData::NonConst>;
-            using SrcSampleType = AudioData::Pointer<AudioData::Int16,   AudioData::NativeEndian, AudioData::Interleaved,    AudioData::Const>;
-
-            DstSampleType dstData (audioBuffer.getWritePointer (i));
-            SrcSampleType srcData (srcInterleaved + i, audioBuffer.getNumChannels());
-            dstData.convertSamples (srcData, numSamples);
-        }
+        const auto numChannels = audioBuffer.getNumChannels();
+
+        AudioData::deinterleaveSamples<AudioData::Int16,   AudioData::NativeEndian,
+                                       AudioData::Float32, AudioData::NativeEndian> (reinterpret_cast<const uint16*> (srcInterleaved),
+                                                                                     numChannels,
+                                                                                     audioBuffer.getArrayOfWritePointers(),
+                                                                                     numChannels,
+                                                                                     numSamples);
     }
 
     static void convertToOboe (const AudioBuffer<float>& audioBuffer, int16* dstInterleaved, int numSamples)
     {
-        for (int i = 0; i < audioBuffer.getNumChannels(); ++i)
-        {
-            using DstSampleType = AudioData::Pointer<AudioData::Int16,   AudioData::NativeEndian, AudioData::Interleaved,    AudioData::NonConst>;
-            using SrcSampleType = AudioData::Pointer<AudioData::Float32, AudioData::NativeEndian, AudioData::NonInterleaved, AudioData::Const>;
-
-            DstSampleType dstData (dstInterleaved + i, audioBuffer.getNumChannels());
-            SrcSampleType srcData (audioBuffer.getReadPointer (i));
-            dstData.convertSamples (srcData, numSamples);
-        }
+        const auto numChannels = audioBuffer.getNumChannels();
+
+        AudioData::interleaveSamples<AudioData::Float32, AudioData::NativeEndian,
+                                     AudioData::Int16,   AudioData::NativeEndian> (audioBuffer.getArrayOfReadPointers(),
+                                                                                   numChannels,
+                                                                                   reinterpret_cast<uint16*> (dstInterleaved),
+                                                                                   numChannels,
+                                                                                   numSamples);
     }
 };
 
@@ -98,15 +96,12 @@ struct OboeAudioIODeviceBufferHelpers<float>
             // No need to convert, we instructed the buffer to point to the src data directly already
             jassert (audioBuffer.getWritePointer (0) != srcInterleaved);
 
-            for (int i = 0; i < numChannels; ++i)
-            {
-                using DstSampleType = AudioData::Pointer<AudioData::Float32, AudioData::NativeEndian, AudioData::NonInterleaved, AudioData::NonConst>;
-                using SrcSampleType = AudioData::Pointer<AudioData::Float32, AudioData::NativeEndian, AudioData::Interleaved,    AudioData::Const>;
-
-                DstSampleType dstData (audioBuffer.getWritePointer (i));
-                SrcSampleType srcData (srcInterleaved + i, audioBuffer.getNumChannels());
-                dstData.convertSamples (srcData, numSamples);
-            }
+            AudioData::deinterleaveSamples<AudioData::Float32, AudioData::NativeEndian,
+                                           AudioData::Float32, AudioData::NativeEndian> (srcInterleaved,
+                                                                                         numChannels,
+                                                                                         audioBuffer.getArrayOfWritePointers(),
+                                                                                         numChannels,
+                                                                                         numSamples);
         }
     }
 
@@ -119,15 +114,12 @@ struct OboeAudioIODeviceBufferHelpers<float>
             // No need to convert, we instructed the buffer to point to the src data directly already
             jassert (audioBuffer.getReadPointer (0) != dstInterleaved);
 
-            for (int i = 0; i < numChannels; ++i)
-            {
-                using DstSampleType = AudioData::Pointer<AudioData::Float32, AudioData::NativeEndian, AudioData::Interleaved,    AudioData::NonConst>;
-                using SrcSampleType = AudioData::Pointer<AudioData::Float32, AudioData::NativeEndian, AudioData::NonInterleaved, AudioData::Const>;
-
-                DstSampleType dstData (dstInterleaved + i, audioBuffer.getNumChannels());
-                SrcSampleType srcData (audioBuffer.getReadPointer (i));
-                dstData.convertSamples (srcData, numSamples);
-            }
+            AudioData::interleaveSamples<AudioData::Float32, AudioData::NativeEndian,
+                                         AudioData::Float32, AudioData::NativeEndian> (audioBuffer.getArrayOfReadPointers(),
+                                                                                       numChannels,
+                                                                                       dstInterleaved,
+                                                                                       numChannels,
+                                                                                       numSamples);
         }
     }
 };

modules/juce_audio_devices/native/juce_android_OpenSL.cpp

@@ -196,29 +196,26 @@ struct BufferHelpers<int16>
 
     static void convertFromOpenSL (const int16* srcInterleaved, AudioBuffer<float>& audioBuffer)
     {
-        for (int i = 0; i < audioBuffer.getNumChannels(); ++i)
-        {
-            using DstSampleType = AudioData::Pointer<AudioData::Float32, AudioData::NativeEndian, AudioData::NonInterleaved, AudioData::NonConst>;
-            using SrcSampleType = AudioData::Pointer<AudioData::Int16,   AudioData::LittleEndian, AudioData::Interleaved,    AudioData::Const>;
-
-            DstSampleType dstData (audioBuffer.getWritePointer (i));
-            SrcSampleType srcData (srcInterleaved + i, audioBuffer.getNumChannels());
-            dstData.convertSamples (srcData, audioBuffer.getNumSamples());
-        }
+        const auto numChannels = audioBuffer.getNumChannels();
+
+        AudioData::deinterleaveSamples<AudioData::Int16,   AudioData::LittleEndian,
+                                       AudioData::Float32, AudioData::NativeEndian> (reinterpret_cast<const uint16*> (srcInterleaved),
+                                                                                     numChannels,
+                                                                                     audioBuffer.getArrayOfWritePointers(),
+                                                                                     numChannels,
+                                                                                     audioBuffer.getNumSamples());
     }
 
     static void convertToOpenSL (const AudioBuffer<float>& audioBuffer, int16* dstInterleaved)
     {
-        for (int i = 0; i < audioBuffer.getNumChannels(); ++i)
-        {
-            using DstSampleType = AudioData::Pointer<AudioData::Int16,   AudioData::LittleEndian, AudioData::Interleaved, AudioData::NonConst>;
-            using SrcSampleType = AudioData::Pointer<AudioData::Float32, AudioData::NativeEndian, AudioData::NonInterleaved, AudioData::Const>;
-
-            DstSampleType dstData (dstInterleaved + i, audioBuffer.getNumChannels());
-            SrcSampleType srcData (audioBuffer.getReadPointer (i));
-
-            dstData.convertSamples (srcData, audioBuffer.getNumSamples());
-        }
+        const auto numChannels = audioBuffer.getNumChannels();
+
+        AudioData::interleaveSamples<AudioData::Float32, AudioData::NativeEndian,
+                                     AudioData::Int16,   AudioData::LittleEndian> (audioBuffer.getArrayOfReadPointers(),
+                                                                                   numChannels,
+                                                                                   reinterpret_cast<uint16*> (dstInterleaved),
+                                                                                   numChannels,
+                                                                                   audioBuffer.getNumSamples());
     }
 
 };
@@ -249,41 +246,38 @@ struct BufferHelpers<float>
 
     static void convertFromOpenSL (const float* srcInterleaved, AudioBuffer<float>& audioBuffer)
     {
-        if (audioBuffer.getNumChannels() == 1)
+        const auto numChannels = audioBuffer.getNumChannels();
+
+        if (numChannels == 1)
         {
             jassert (srcInterleaved == audioBuffer.getWritePointer (0));
             return;
         }
 
-        for (int i = 0; i < audioBuffer.getNumChannels(); ++i)
-        {
-            using DstSampleType = AudioData::Pointer<AudioData::Float32, AudioData::NativeEndian, AudioData::NonInterleaved, AudioData::NonConst>;
-            using SrcSampleType = AudioData::Pointer<AudioData::Float32, AudioData::LittleEndian, AudioData::Interleaved,    AudioData::Const>;
-
-            DstSampleType dstData (audioBuffer.getWritePointer (i));
-            SrcSampleType srcData (srcInterleaved + i, audioBuffer.getNumChannels());
-            dstData.convertSamples (srcData, audioBuffer.getNumSamples());
-        }
+        AudioData::deinterleaveSamples<AudioData::Float32, AudioData::LittleEndian,
+                                       AudioData::Float32, AudioData::NativeEndian> (srcInterleaved,
+                                                                                     numChannels,
+                                                                                     audioBuffer.getArrayOfWritePointers(),
+                                                                                     numChannels,
+                                                                                     audioBuffer.getNumSamples());
     }
 
     static void convertToOpenSL (const AudioBuffer<float>& audioBuffer, float* dstInterleaved)
     {
-        if (audioBuffer.getNumChannels() == 1)
+        const auto numChannels = audioBuffer.getNumChannels();
+
+        if (numChannels == 1)
         {
             jassert (dstInterleaved == audioBuffer.getReadPointer (0));
             return;
         }
 
-        for (int i = 0; i < audioBuffer.getNumChannels(); ++i)
-        {
-            using DstSampleType = AudioData::Pointer<AudioData::Float32, AudioData::LittleEndian, AudioData::Interleaved,    AudioData::NonConst>;
-            using SrcSampleType = AudioData::Pointer<AudioData::Float32, AudioData::NativeEndian, AudioData::NonInterleaved, AudioData::Const>;
-
-            DstSampleType dstData (dstInterleaved + i, audioBuffer.getNumChannels());
-            SrcSampleType srcData (audioBuffer.getReadPointer (i));
-
-            dstData.convertSamples (srcData, audioBuffer.getNumSamples());
-        }
+        AudioData::interleaveSamples<AudioData::Float32, AudioData::NativeEndian,
+                                     AudioData::Float32, AudioData::LittleEndian> (audioBuffer.getArrayOfReadPointers(),
+                                                                                   numChannels,
+                                                                                   dstInterleaved,
+                                                                                   numChannels,
+                                                                                   audioBuffer.getNumSamples());
     }
 };
 

modules/juce_audio_utils/native/juce_mac_AudioCDBurner.mm

@@ -132,13 +132,10 @@ private:
 
                 source->source->getNextAudioBlock (info);
 
-                typedef AudioData::Pointer <AudioData::Int16,   AudioData::LittleEndian, AudioData::Interleaved,    AudioData::NonConst> CDSampleFormat;
-                typedef AudioData::Pointer <AudioData::Float32, AudioData::NativeEndian, AudioData::NonInterleaved, AudioData::Const> SourceSampleFormat;
-
-                CDSampleFormat left (buffer, 2);
-                left.convertSamples (SourceSampleFormat (tempBuffer.getReadPointer (0)), numSamples);
-                CDSampleFormat right (buffer + 2, 2);
-                right.convertSamples (SourceSampleFormat (tempBuffer.getReadPointer (1)), numSamples);
+                AudioData::interleaveSamples<AudioData::Float32, AudioData::NativeEndian,
+                                             AudioData::Int16,   AudioData::LittleEndian> (tempBuffer.getArrayOfReadPointers(), 2,
+                                                                                           reinterpret_cast<uint16*> (buffer), 2,
+                                                                                           numSamples);
 
                 source->readPosition += numSamples;
             }

modules/juce_audio_utils/native/juce_win32_AudioCDBurner.cpp

@@ -388,16 +388,10 @@ bool AudioCDBurner::addAudioTrack (AudioSource* audioSource, int numSamples)
 
         buffer.clear (bytesPerBlock);
 
-        typedef AudioData::Pointer <AudioData::Int16, AudioData::LittleEndian,
-                                    AudioData::Interleaved, AudioData::NonConst> CDSampleFormat;
-
-        typedef AudioData::Pointer <AudioData::Float32, AudioData::NativeEndian,
-                                    AudioData::NonInterleaved, AudioData::Const> SourceSampleFormat;
-
-        CDSampleFormat left (buffer, 2);
-        left.convertSamples (SourceSampleFormat (sourceBuffer.getReadPointer (0)), samplesPerBlock);
-        CDSampleFormat right (buffer + 2, 2);
-        right.convertSamples (SourceSampleFormat (sourceBuffer.getReadPointer (1)), samplesPerBlock);
+        AudioData::interleaveSamples<AudioData::Float32, AudioData::NativeEndian,
+                                     AudioData::Int16,   AudioData::LittleEndian> (sourceBuffer.getArrayOfReadPointers(), 2,
+                                                                                   reinterpret_cast<uint16*> (buffer), 2,
+                                                                                   samplesPerBlock);
 
         hr = pimpl->redbook->AddAudioTrackBlocks (buffer, bytesPerBlock);