Revert "coreaudio: modern C++"

This reverts commit f3d677fb.

Change-Id: Iebfbb5554c1d910c7ea0d919b13669e760d512ca

src/media/audio/coreaudio/corelayer.cpp

@@ -27,7 +27,8 @@
 #include "audiodevice.h"
 
 #include <cmath>
-#include <vector>
+#include <thread>
+#include <atomic>
 
 namespace ring {
 
@@ -38,52 +39,55 @@ CoreLayer::CoreLayer(const AudioPreference &pref)
     , indexOut_(pref.getAlsaCardout())
     , indexRing_(pref.getAlsaCardring())
     , playbackBuff_(0, audioFormat_)
+    , captureBuff_(0)
     , mainRingBuffer_(Manager::instance().getRingBufferPool().getRingBuffer(RingBufferPool::DEFAULT_ID))
 {}
 
 CoreLayer::~CoreLayer()
-{}
+{
+    if (captureBuff_) {
+        for (UInt32 i = 0; i < captureBuff_->mNumberBuffers; ++i)
+            free(captureBuff_->mBuffers[i].mData);
+        free(captureBuff_);
+        captureBuff_ = 0;
+    }
+}
 
-std::vector<std::string>
-CoreLayer::getCaptureDeviceList() const
+std::vector<std::string> CoreLayer::getCaptureDeviceList() const
 {
     std::vector<std::string> ret;
 
 #if !TARGET_OS_IPHONE
-    for (const auto& x : getDeviceList(true))
+    for (auto x : getDeviceList(true))
         ret.push_back(x.name_);
 #endif
 
     return ret;
 }
 
-std::vector<std::string>
-CoreLayer::getPlaybackDeviceList() const
+std::vector<std::string> CoreLayer::getPlaybackDeviceList() const
 {
     std::vector<std::string> ret;
 
 #if !TARGET_OS_IPHONE
-    for (const auto& x : getDeviceList(false))
+    for (auto x : getDeviceList(false))
         ret.push_back(x.name_);
 #endif
 
     return ret;
 }
 
-int
-CoreLayer::getAudioDeviceIndex(const std::string& name, DeviceType type) const
+int CoreLayer::getAudioDeviceIndex(const std::string& name, DeviceType type) const
 {
     return 0;
 }
 
-std::string
-CoreLayer::getAudioDeviceName(int index, DeviceType type) const
+std::string CoreLayer::getAudioDeviceName(int index, DeviceType type) const
 {
     return "";
 }
 
-void
-CoreLayer::initAudioLayerIO()
+void CoreLayer::initAudioLayerIO()
 {
     // OS X uses Audio Units for output. Steps:
     // 1) Create a description.
@@ -103,8 +107,8 @@ CoreLayer::initAudioLayerIO()
     desc.componentSubType = kAudioUnitSubType_VoiceProcessingIO;
     desc.componentManufacturer = kAudioUnitManufacturer_Apple;
 
-    auto comp = AudioComponentFindNext(nullptr, &desc);
-    if (comp == nullptr) {
+    AudioComponent comp = AudioComponentFindNext(NULL, &desc);
+    if (comp == NULL) {
         RING_ERR("Can't find default output audio component.");
         return;
     }
@@ -121,8 +125,7 @@ CoreLayer::initAudioLayerIO()
             &info,
             &size));
 
-    audioFormat_ = {static_cast<unsigned int>(info.mSampleRate),
-                    static_cast<unsigned int>(info.mChannelsPerFrame)};
+    audioFormat_ = {(unsigned int)info.mSampleRate, (unsigned int)info.mChannelsPerFrame};
 
     checkErr(AudioUnitGetProperty(ioUnit_,
             kAudioUnitProperty_StreamFormat,
@@ -151,8 +154,7 @@ CoreLayer::initAudioLayerIO()
                 &info,
                 &size));
 
-    audioInputFormat_ = {static_cast<unsigned int>(info.mSampleRate),
-                         static_cast<unsigned int>(info.mChannelsPerFrame)};
+    audioInputFormat_ = {(unsigned int)info.mSampleRate, (unsigned int)info.mChannelsPerFrame};
     hardwareInputFormatAvailable(audioInputFormat_);
 
     // Set format on output *SCOPE* in input *BUS*.
@@ -194,15 +196,15 @@ CoreLayer::initAudioLayerIO()
 #endif
 
     UInt32 bufferSizeBytes = bufferSizeFrames * sizeof(Float32);
-    size = offsetof(AudioBufferList, mBuffers) + (sizeof(AudioBuffer) * info.mChannelsPerFrame);
-    rawBuff_.reset(new Byte[size + bufferSizeBytes * info.mChannelsPerFrame]);
-    captureBuff_ = reinterpret_cast<::AudioBufferList*>(rawBuff_.get());
+    size = offsetof(AudioBufferList, mBuffers[0]) +
+        (sizeof(AudioBuffer) * info.mChannelsPerFrame);
+    captureBuff_ = (AudioBufferList *)malloc(size);
     captureBuff_->mNumberBuffers = info.mChannelsPerFrame;
 
     for (UInt32 i = 0; i < captureBuff_->mNumberBuffers; ++i) {
         captureBuff_->mBuffers[i].mNumberChannels = 1;
         captureBuff_->mBuffers[i].mDataByteSize = bufferSizeBytes;
-        captureBuff_->mBuffers[i].mData = rawBuff_.get() + bufferSizeBytes * i;
+        captureBuff_->mBuffers[i].mData = malloc(bufferSizeBytes);
     }
 
     // Input callback setup.
@@ -230,8 +232,7 @@ CoreLayer::initAudioLayerIO()
                sizeof(AURenderCallbackStruct)));
 }
 
-void
-CoreLayer::startStream()
+void CoreLayer::startStream()
 {
     RING_DBG("START STREAM");
 
@@ -251,16 +252,14 @@ CoreLayer::startStream()
     checkErr(AudioOutputUnitStart(ioUnit_));
 }
 
-void
-CoreLayer::destroyAudioLayer()
+void CoreLayer::destroyAudioLayer()
 {
     AudioOutputUnitStop(ioUnit_);
     AudioUnitUninitialize(ioUnit_);
     AudioComponentInstanceDispose(ioUnit_);
 }
 
-void
-CoreLayer::stopStream()
+void CoreLayer::stopStream()
 {
     RING_DBG("STOP STREAM");
 
@@ -280,8 +279,7 @@ CoreLayer::stopStream()
 
 //// PRIVATE /////
 
-OSStatus
-CoreLayer::outputCallback(void* inRefCon,
+OSStatus CoreLayer::outputCallback(void* inRefCon,
     AudioUnitRenderActionFlags* ioActionFlags,
     const AudioTimeStamp* inTimeStamp,
     UInt32 inBusNumber,
@@ -292,8 +290,7 @@ CoreLayer::outputCallback(void* inRefCon,
     return kAudioServicesNoError;
 }
 
-void
-CoreLayer::write(AudioUnitRenderActionFlags* ioActionFlags,
+void CoreLayer::write(AudioUnitRenderActionFlags* ioActionFlags,
     const AudioTimeStamp* inTimeStamp,
     UInt32 inBusNumber,
     UInt32 inNumberFrames,
@@ -306,29 +303,26 @@ CoreLayer::write(AudioUnitRenderActionFlags* ioActionFlags,
 
     if (toPlay.frames() == 0) {
         for (int i = 0; i < audioFormat_.nb_channels; ++i)
-            std::fill_n(reinterpret_cast<Float32*>(ioData->mBuffers[i].mData),
-                        ioData->mBuffers[i].mDataByteSize, 0);
-    } else {
+            memset((Float32*)ioData->mBuffers[i].mData, 0, ioData->mBuffers[i].mDataByteSize);
+    }
+    else {
         for (int i = 0; i < audioFormat_.nb_channels; ++i)
-            toPlay.channelToFloat(reinterpret_cast<Float32*>(ioData->mBuffers[i].mData), i);
+            toPlay.channelToFloat((Float32*)ioData->mBuffers[i].mData, i);
     }
 }
 
-OSStatus
-CoreLayer::inputCallback(void* inRefCon,
+OSStatus CoreLayer::inputCallback(void* inRefCon,
     AudioUnitRenderActionFlags* ioActionFlags,
     const AudioTimeStamp* inTimeStamp,
     UInt32 inBusNumber,
     UInt32 inNumberFrames,
     AudioBufferList* ioData)
 {
-    static_cast<CoreLayer*>(inRefCon)->read(ioActionFlags, inTimeStamp, inBusNumber,
-                                            inNumberFrames, ioData);
+    static_cast<CoreLayer*>(inRefCon)->read(ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames, ioData);
     return kAudioServicesNoError;
 }
 
-void
-CoreLayer::read(AudioUnitRenderActionFlags* ioActionFlags,
+void CoreLayer::read(AudioUnitRenderActionFlags* ioActionFlags,
     const AudioTimeStamp* inTimeStamp,
     UInt32 inBusNumber,
     UInt32 inNumberFrames,
@@ -361,12 +355,12 @@ CoreLayer::read(AudioUnitRenderActionFlags* ioActionFlags,
     bool resample = info.mSampleRate != mainBufferFormat.sample_rate;
 
     // FIXME: Performance! There *must* be a better way. This is testing only.
-    auto inBuff = AudioBuffer {inNumberFrames, audioInputFormat_};
+    AudioBuffer inBuff(inNumberFrames, audioInputFormat_);
 
     for (int i = 0; i < info.mChannelsPerFrame; ++i) {
-        auto data = reinterpret_cast<Float32*>(captureBuff_->mBuffers[i].mData);
+        Float32* data = (Float32*)captureBuff_->mBuffers[i].mData;
         for (int j = 0; j < inNumberFrames; ++j) {
-            (*inBuff.getChannel(i))[j] = static_cast<AudioSample>(data[j] / .000030517578125f);
+            (*inBuff.getChannel(i))[j] = (AudioSample)((data)[j] / .000030517578125f);
         }
     }
 
@@ -375,8 +369,8 @@ CoreLayer::read(AudioUnitRenderActionFlags* ioActionFlags,
 
         //FIXME: May be a multiplication, check alsa vs pulse implementation.
 
-        UInt32 outSamples = inNumberFrames / (static_cast<double>(audioInputFormat_.sample_rate) / mainBufferFormat.sample_rate);
-        auto out = AudioBuffer {outSamples, mainBufferFormat};
+        int outSamples = inNumberFrames / (static_cast<double>(audioInputFormat_.sample_rate) / mainBufferFormat.sample_rate);
+        AudioBuffer out(outSamples, mainBufferFormat);
         inputResampler_->resample(inBuff, out);
         dcblocker_.process(out);
         mainRingBuffer_->put(out);
@@ -406,8 +400,7 @@ void CoreLayer::updatePreference(AudioPreference &preference, int index, DeviceT
     }
 }
 
-std::vector<AudioDevice>
-CoreLayer::getDeviceList(bool getCapture) const
+std::vector<AudioDevice> CoreLayer::getDeviceList(bool getCapture) const
 {
     std::vector<AudioDevice> ret;
 #if !TARGET_OS_IPHONE
@@ -416,31 +409,31 @@ CoreLayer::getDeviceList(bool getCapture) const
     AudioObjectPropertyAddress theAddress = {
         kAudioHardwarePropertyDevices,
         kAudioObjectPropertyScopeGlobal,
-        kAudioObjectPropertyElementMaster
-    };
+        kAudioObjectPropertyElementMaster };
 
     verify_noerr(AudioObjectGetPropertyDataSize(kAudioObjectSystemObject,
-                                                &theAddress,
-                                                0,
-                                                nullptr,
-                                                &propsize));
+                            &theAddress,
+                            0,
+                            NULL,
+                            &propsize));
 
-    std::size_t nDevices = propsize / sizeof(AudioDeviceID);
-    auto devids = std::vector<AudioDeviceID>(nDevices);
+    size_t nDevices = propsize / sizeof(AudioDeviceID);
+    AudioDeviceID *devids = new AudioDeviceID[nDevices];
 
     verify_noerr(AudioObjectGetPropertyData(kAudioObjectSystemObject,
-                                            &theAddress,
-                                            0,
-                                            nullptr,
-                                            &propsize,
-                                            devids.data()));
+                        &theAddress,
+                        0,
+                        NULL,
+                        &propsize,
+                        devids));
 
     for (int i = 0; i < nDevices; ++i) {
-        auto dev = AudioDevice {devids[i], getCapture};
+        AudioDevice dev(devids[i], getCapture);
         if (dev.channels_ > 0) { // Channels < 0 if inactive.
-            ret.push_back(std::move(dev));
+            ret.push_back(dev);
         }
     }
+    delete[] devids;
 #endif
     return ret;
 }

src/media/audio/coreaudio/corelayer.h

@@ -162,8 +162,7 @@ class CoreLayer : public AudioLayer {
 
         /** Non-interleaved audio buffers */
         AudioBuffer playbackBuff_;
-        ::AudioBufferList* captureBuff_ {nullptr}; // CoreAudio buffer (pointer is casted rawBuff_)
-        std::unique_ptr<Byte[]> rawBuff_; // raw allocation of captureBuff_
+        ::AudioBufferList* captureBuff_; // CoreAudio buffer.
 
         /** Interleaved buffer */
         std::vector<AudioSample> playbackIBuff_;