diff --git a/toonz/sources/common/tsound/tsound_mac.cpp b/toonz/sources/common/tsound/tsound_mac.cpp
index e801618..50caa62 100644
--- a/toonz/sources/common/tsound/tsound_mac.cpp
+++ b/toonz/sources/common/tsound/tsound_mac.cpp
@@ -10,10 +10,11 @@
#include <queue>
#include <set>
-#include <CoreServices/CoreServices.h>
-#include <AudioUnit/AudioUnit.h>
-#include <CoreAudio/CoreAudio.h>
-#include <AudioToolbox/AudioToolbox.h>
+#include <QByteArray>
+#include <QAudioFormat>
+#include <QBuffer>
+#include <QAudioOutput>
+
using namespace std;
//==============================================================================
@@ -29,11 +30,10 @@ public:
TSoundTrackFormat m_currentFormat;
std::set<int> m_supportedRate;
bool m_opened;
- AudioFileID musicFileID;
- AudioUnit theOutputUnit;
- AudioStreamBasicDescription fileASBD;
- AudioStreamBasicDescription outputASBD;
- AudioConverterRef converter;
+ double m_volume = 0.5;
+
+ QAudioOutput *m_audioOutput;
+ QBuffer *m_buffer;
TSoundOutputDeviceImp()
: m_isPlaying(false)
@@ -50,6 +50,7 @@ public:
bool doStopDevice();
void play(const TSoundTrackP &st, TINT32 s0, TINT32 s1, bool loop,
bool scrubbing);
+ void prepareVolume(double volume);
};
//-----------------------------------------------------------------------------
@@ -58,15 +59,14 @@ namespace {
struct MyData {
char *entireFileBuffer;
- UInt64 totalPacketCount;
- UInt64 fileByteCount;
- UInt32 maxPacketSize;
- UInt64 packetOffset;
- UInt64 byteOffset;
+ quint64 totalPacketCount;
+ quint64 fileByteCount;
+ quint32 maxPacketSize;
+ quint64 packetOffset;
+ quint64 byteOffset;
bool m_doNotify;
-
void *sourceBuffer;
- AudioConverterRef converter;
+ // AudioConverterRef converter;
std::shared_ptr<TSoundOutputDeviceImp> imp;
bool isLooping;
MyData()
@@ -110,218 +110,16 @@ public:
fflush(stdout); \
}
-extern "C" {
-// This is an example of a Input Procedure from a call to
-// AudioConverterFillComplexBuffer.
-// The total amount of data needed is "ioNumberDataPackets" when this method is
-// first called.
-// On exit, "ioNumberDataPackets" must be set to the actual amount of data
-// obtained.
-// Upon completion, all new input data must point to the AudioBufferList in the
-// parameter ( "ioData" )
-OSStatus MyACComplexInputProc(
- AudioConverterRef inAudioConverter, UInt32 *ioNumberDataPackets,
- AudioBufferList *ioData,
- AudioStreamPacketDescription **outDataPacketDescription, void *inUserData) {
- OSStatus err = noErr;
- UInt32 bytesCopied = 0;
-
- MyData *myData = static_cast<MyData *>(inUserData);
-
- // initialize in case of failure
- ioData->mBuffers[0].mData = NULL;
- ioData->mBuffers[0].mDataByteSize = 0;
-
- {
- // TThread::ScopedLock sl(MutexOut);
- if (myData->imp->m_isPlaying == false) return noErr;
- }
-
- // if there are not enough packets to satisfy request, then read what's left
- if (myData->packetOffset + *ioNumberDataPackets > myData->totalPacketCount)
- *ioNumberDataPackets = myData->totalPacketCount - myData->packetOffset;
-
- // do nothing if there are no packets available
- if (*ioNumberDataPackets) {
- if (myData->sourceBuffer != NULL) {
- free(myData->sourceBuffer);
- myData->sourceBuffer = NULL;
- }
-
- // the total amount of data requested by the AudioConverter
- bytesCopied = *ioNumberDataPackets * myData->maxPacketSize;
- // alloc a small buffer for the AudioConverter to use.
- myData->sourceBuffer = (void *)calloc(1, bytesCopied);
- // copy the amount of data needed (bytesCopied) from buffer of audio file
- memcpy(myData->sourceBuffer, myData->entireFileBuffer + myData->byteOffset,
- bytesCopied);
-
- // keep track of where we want to read from next time
- myData->byteOffset += *ioNumberDataPackets * myData->maxPacketSize;
- myData->packetOffset += *ioNumberDataPackets;
-
- ioData->mBuffers[0].mData = myData->sourceBuffer; // tell the Audio
- // Converter where it's
- // source data is
- ioData->mBuffers[0].mDataByteSize =
- bytesCopied; // tell the Audio Converter how much data in each buffer
- } else {
- // there aren't any more packets to read.
- // Set the amount of data read (mDataByteSize) to zero
- // and return noErr to signal the AudioConverter there are
- // no packets left.
-
- ioData->mBuffers[0].mData = NULL;
- ioData->mBuffers[0].mDataByteSize = 0;
- delete[] myData->entireFileBuffer;
- myData->entireFileBuffer = 0;
- err = noErr;
- /*
-{
-TThread::ScopedLock sl(MutexOut);
-*(myData->isPlaying) = false; //questo lo faccio nel main thread
-}
-*/
- PlayCompletedMsg(myData).send();
- }
-
- return err;
-}
-
-OSStatus MyFileRenderProc(void *inRefCon,
- AudioUnitRenderActionFlags *inActionFlags,
- const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber,
- UInt32 inNumFrames, AudioBufferList *ioData) {
- MyData *myData = static_cast<MyData *>(inRefCon);
- OSStatus err = noErr;
- void *inInputDataProcUserData = inRefCon;
- AudioStreamPacketDescription *outPacketDescription = NULL;
- // To obtain a data buffer of converted data from a complex input
- // source(compressed files, etc.)
- // use AudioConverterFillComplexBuffer. The total amount of data requested is
- // "inNumFrames" and
- // on return is set to the actual amount of data recieved.
- // All converted data is returned to "ioData" (AudioBufferList).
- err = AudioConverterFillComplexBuffer(myData->converter, MyACComplexInputProc,
- inInputDataProcUserData, &inNumFrames,
- ioData, outPacketDescription);
-
- /*Parameters for AudioConverterFillComplexBuffer()
-converter - the converter being used
-ACComplexInputProc() - input procedure to supply data to the Audio Converter
-inInputDataProcUserData - Used to hold any data that needs to be passed on. Not
-needed in this example.
-inNumFrames - The amount of requested data. On output, this
-number is the amount actually received.
-ioData - Buffer of the converted data recieved on return
-outPacketDescription - contains the format of the returned data. Not used in
-this example.
-*/
-
- // checkStatus(err);
- return err;
-}
-
-} // extern "C"
-
-void PrintStreamDesc(AudioStreamBasicDescription *inDesc) {
- if (!inDesc) {
- printf("Can't print a NULL desc!\n");
- return;
- }
-
- printf("- - - - - - - - - - - - - - - - - - - -\n");
- printf(" Sample Rate:%f\n", inDesc->mSampleRate);
- printf(" Format ID:%.*s\n", (int)sizeof(inDesc->mFormatID),
- (char *)&inDesc->mFormatID);
- printf(" Format Flags:%lX\n", inDesc->mFormatFlags);
- printf(" Bytes per Packet:%ld\n", inDesc->mBytesPerPacket);
- printf(" Frames per Packet:%ld\n", inDesc->mFramesPerPacket);
- printf(" Bytes per Frame:%ld\n", inDesc->mBytesPerFrame);
- printf(" Channels per Frame:%ld\n", inDesc->mChannelsPerFrame);
- printf(" Bits per Channel:%ld\n", inDesc->mBitsPerChannel);
- printf("- - - - - - - - - - - - - - - - - - - -\n");
-}
-
bool TSoundOutputDeviceImp::doOpenDevice() {
- m_opened = false;
- OSStatus err = noErr;
- ComponentDescription desc;
- Component comp;
-
- desc.componentType = kAudioUnitType_Output;
- desc.componentSubType = kAudioUnitSubType_DefaultOutput;
- // all Audio Units in AUComponent.h must use "kAudioUnitManufacturer_Apple" as
- // the Manufacturer
- desc.componentManufacturer = kAudioUnitManufacturer_Apple;
- desc.componentFlags = 0;
- desc.componentFlagsMask = 0;
-
- comp = FindNextComponent(
- NULL, &desc); // Finds an component that meets the desc spec's
- if (comp == NULL) return false;
- err = OpenAComponent(comp, &theOutputUnit); // gains access to the services
- // provided by the component
- if (err) return false;
-
- UInt32 size;
- Boolean outWritable;
- UInt32 theInputBus = 0;
- // Gets the size of the Stream Format Property and if it is writable
- err =
- AudioUnitGetPropertyInfo(theOutputUnit, kAudioUnitProperty_StreamFormat,
- kAudioUnitScope_Output, 0, &size, &outWritable);
- // Get the current stream format of the output
- err = AudioUnitGetProperty(theOutputUnit, kAudioUnitProperty_StreamFormat,
- kAudioUnitScope_Output, 0, &outputASBD, &size);
- checkStatus(err);
- // Set the stream format of the output to match the input
- err = AudioUnitSetProperty(theOutputUnit, kAudioUnitProperty_StreamFormat,
- kAudioUnitScope_Input, theInputBus, &outputASBD,
- size);
- checkStatus(err);
-
- // Initialize AudioUnit, alloc mem buffers for processing
- err = AudioUnitInitialize(theOutputUnit);
- checkStatus(err);
- if (err == noErr) m_opened = true;
+ m_opened = false;
+ m_audioOutput = NULL;
+ m_opened = true;
return m_opened;
}
bool TSoundOutputDeviceImp::doSetStreamFormat(const TSoundTrackFormat &format) {
if (!m_opened) doOpenDevice();
if (!m_opened) return false;
-
- fileASBD.mSampleRate = format.m_sampleRate;
- fileASBD.mFormatID = kAudioFormatLinearPCM;
- fileASBD.mFormatFlags = 14;
- /*
-Standard flags: kAudioFormatFlagIsFloat = (1L << 0)
-kAudioFormatFlagIsBigEndian = (1L << 1)
-kAudioFormatFlagIsSignedInteger = (1L << 2)
-kAudioFormatFlagIsPacked = (1L << 3)
-kAudioFormatFlagIsAlignedHigh = (1L << 4)
-kAudioFormatFlagIsNonInterleaved = (1L << 5)
-kAudioFormatFlagsAreAllClear = (1L << 31)
-
-Linear PCM flags:
-kLinearPCMFormatFlagIsFloat = kAudioFormatFlagIsFloat
-kLinearPCMFormatFlagIsBigEndian = kAudioFormatFlagIsBigEndian
-kLinearPCMFormatFlagIsSignedInteger = kAudioFormatFlagIsSignedInteger
-kLinearPCMFormatFlagIsPacked = kAudioFormatFlagIsPacked
-kLinearPCMFormatFlagIsAlignedHigh = kAudioFormatFlagIsAlignedHigh
-kLinearPCMFormatFlagIsNonInterleaved = kAudioFormatFlagIsNonInterleaved
-kLinearPCMFormatFlagsAreAllClear = kAudioFormatFlagsAreAllClear
-*/
- fileASBD.mBytesPerPacket =
- (format.m_bitPerSample >> 3) * format.m_channelCount;
- fileASBD.mFramesPerPacket = 1;
- fileASBD.mBytesPerFrame =
- (format.m_bitPerSample >> 3) * format.m_channelCount;
- fileASBD.mChannelsPerFrame = format.m_channelCount;
- fileASBD.mBitsPerChannel = format.m_bitPerSample;
- fileASBD.mReserved = 0;
- // PrintStreamDesc(&fileASBD);
m_opened = true;
return true;
}
@@ -365,18 +163,25 @@ bool TSoundOutputDevice::open(const TSoundTrackP &st) {
bool TSoundOutputDevice::close() {
stop();
m_imp->m_opened = false;
- AudioUnitUninitialize(
- m_imp->theOutputUnit); // release resources without closing the component
- CloseComponent(m_imp->theOutputUnit); // Terminates your application's access
- // to the services provided
return true;
}
//------------------------------------------------------------------------------
+void TSoundOutputDeviceImp::prepareVolume(double volume) {
+ m_volume = volume;
+}
+
+//------------------------------------------------------------------------------
+
+void TSoundOutputDevice::prepareVolume(double volume) {
+ m_imp->prepareVolume(volume);
+}
+
+//------------------------------------------------------------------------------
+
void TSoundOutputDevice::play(const TSoundTrackP &st, TINT32 s0, TINT32 s1,
bool loop, bool scrubbing) {
- // TThread::ScopedLock sl(MutexOut);
int lastSample = st->getSampleCount() - 1;
notLessThan(0, s0);
notLessThan(0, s1);
@@ -406,105 +211,72 @@ void TSoundOutputDeviceImp::play(const TSoundTrackP &st, TINT32 s0, TINT32 s1,
bool loop, bool scrubbing) {
if (!doSetStreamFormat(st->getFormat())) return;
- OSStatus err = noErr;
MyData *myData = new MyData();
- myData->imp = shared_from_this();
- UInt32 magicCookieSize = 0;
- // PrintStreamDesc(&outputASBD);
- err = AudioConverterNew(&fileASBD, &outputASBD, &converter);
- checkStatus(err);
- err = AudioFileGetPropertyInfo(musicFileID, kAudioFilePropertyMagicCookieData,
- &magicCookieSize, NULL);
-
- if (err == noErr) {
- void *magicCookie = calloc(1, magicCookieSize);
- if (magicCookie) {
- // Get Magic Cookie data from Audio File
- err = AudioFileGetProperty(musicFileID, kAudioFilePropertyMagicCookieData,
- &magicCookieSize, magicCookie);
-
- // Give the AudioConverter the magic cookie decompression params if there
- // are any
- if (err == noErr) {
- err = AudioConverterSetProperty(myData->converter,
- kAudioConverterDecompressionMagicCookie,
- magicCookieSize, magicCookie);
- }
- err = noErr;
- if (magicCookie) free(magicCookie);
- }
- } else // this is OK because some audio data doesn't need magic cookie data
- err = noErr;
-
- checkStatus(err);
- myData->converter = converter;
+ myData->imp = shared_from_this();
myData->totalPacketCount = s1 - s0;
myData->fileByteCount = (s1 - s0) * st->getSampleSize();
myData->entireFileBuffer = new char[myData->fileByteCount];
-#if defined(i386)
- if (st->getBitPerSample() == 16) {
- int i;
- USHORT *dst = (USHORT *)(myData->entireFileBuffer);
- USHORT *src = (USHORT *)(st->getRawData() + s0 * st->getSampleSize());
-
- for (i = 0; i < myData->fileByteCount / 2; i++) *dst++ = swapUshort(*src++);
- } else
- memcpy(myData->entireFileBuffer,
- st->getRawData() + s0 * st->getSampleSize(), myData->fileByteCount);
-#else
memcpy(myData->entireFileBuffer, st->getRawData() + s0 * st->getSampleSize(),
myData->fileByteCount);
-#endif
- myData->maxPacketSize = fileASBD.mFramesPerPacket * fileASBD.mBytesPerFrame;
- {
- // TThread::ScopedLock sl(MutexOut);
- m_isPlaying = true;
- }
+ m_isPlaying = true;
myData->isLooping = loop;
- // cout << "total packet count = " << myData->totalPacketCount <<endl;
- // cout << "filebytecount " << myData->fileByteCount << endl;
-
- AURenderCallbackStruct renderCallback;
- memset(&renderCallback, 0, sizeof(AURenderCallbackStruct));
-
- renderCallback.inputProc = MyFileRenderProc;
- renderCallback.inputProcRefCon = myData;
-
- // Sets the callback for the Audio Unit to the renderCallback
- err =
- AudioUnitSetProperty(theOutputUnit, kAudioUnitProperty_SetRenderCallback,
- kAudioUnitScope_Input, 0, &renderCallback,
- sizeof(AURenderCallbackStruct));
-
- checkStatus(err);
-
- err = AudioOutputUnitStart(theOutputUnit);
-
- checkStatus(err);
+ QAudioFormat format;
+ QAudioDeviceInfo info(QAudioDeviceInfo::defaultOutputDevice());
+
+ format.setSampleSize(st->getBitPerSample());
+ format.setCodec("audio/pcm");
+ format.setChannelCount(st->getChannelCount());
+ format.setByteOrder(QAudioFormat::LittleEndian);
+ format.setSampleType(st->getFormat().m_signedSample
+ ? QAudioFormat::SignedInt
+ : QAudioFormat::UnSignedInt);
+ format.setSampleRate(st->getSampleRate());
+ QList<QAudioFormat::Endian> sbos = info.supportedByteOrders();
+ QList<int> sccs = info.supportedChannelCounts();
+ QList<int> ssrs = info.supportedSampleRates();
+ QList<QAudioFormat::SampleType> sstypes = info.supportedSampleTypes();
+ QList<int> ssss = info.supportedSampleSizes();
+ QStringList supCodes = info.supportedCodecs();
+ if (!info.isFormatSupported((format))) {
+ format = info.nearestFormat(format);
+ int newChannels = format.channelCount();
+ int newBitsPerSample = format.sampleSize();
+ int newSampleRate = format.sampleRate();
+ QAudioFormat::SampleType newSampleType = format.sampleType();
+ QAudioFormat::Endian newBo = format.byteOrder();
+ }
+ int test = st->getSampleCount() / st->getSampleRate();
+ QByteArray *data =
+ new QByteArray(myData->entireFileBuffer, myData->fileByteCount);
+ QBuffer *newBuffer = new QBuffer;
+ newBuffer->setBuffer(data);
+ newBuffer->open(QIODevice::ReadOnly);
+ newBuffer->seek(0);
+ if (m_audioOutput == NULL) {
+ m_audioOutput = new QAudioOutput(format, NULL);
+ }
+ m_audioOutput->start(newBuffer);
+ m_audioOutput->setVolume(m_volume);
}
//------------------------------------------------------------------------------
bool TSoundOutputDeviceImp::doStopDevice() {
m_isPlaying = false;
- AudioOutputUnitStop(
- theOutputUnit); // you must stop the audio unit from processing
- AudioConverterDispose(
- converter); // deallocates the memory used by inAudioConverter
+ m_audioOutput->stop();
+
return true;
}
//------------------------------------------------------------------------------
void TSoundOutputDevice::stop() {
- // TThread::ScopedLock sl(MutexOut);
if (m_imp->m_opened == false) return;
- // TThread::ScopedLock sl(MutexOut);
m_imp->doStopDevice();
}
@@ -523,28 +295,16 @@ void TSoundOutputDevice::detach(TSoundOutputDeviceListener *listener) {
//------------------------------------------------------------------------------
double TSoundOutputDevice::getVolume() {
- if (!m_imp->m_opened) m_imp->doOpenDevice();
-
- Float32 leftVol, rightVol;
- AudioUnitGetParameter(m_imp->theOutputUnit, kHALOutputParam_Volume,
- kAudioUnitScope_Output, 0, &leftVol);
-
- AudioUnitGetParameter(m_imp->theOutputUnit, kHALOutputParam_Volume,
- kAudioUnitScope_Output, 0, &rightVol);
- double vol = (leftVol + rightVol) / 2;
-
- return (vol < 0. ? 0. : vol);
+ if (m_imp->m_audioOutput != NULL)
+ return m_imp->m_audioOutput->volume();
+ else return m_imp->m_volume;
}
//------------------------------------------------------------------------------
bool TSoundOutputDevice::setVolume(double volume) {
- Float32 vol = volume;
- AudioUnitSetParameter(m_imp->theOutputUnit, kHALOutputParam_Volume,
- kAudioUnitScope_Output, 0, vol, 0);
-
- AudioUnitSetParameter(m_imp->theOutputUnit, kHALOutputParam_Volume,
- kAudioUnitScope_Output, 0, vol, 0);
+ m_imp->m_volume = volume;
+ m_imp->m_audioOutput->setVolume(volume);
return true;
}
@@ -554,24 +314,15 @@ bool TSoundOutputDevice::supportsVolume() { return true; }
//------------------------------------------------------------------------------
-bool TSoundOutputDevice::isPlaying() const {
- // TThread::ScopedLock sl(MutexOut);
- return m_imp->m_isPlaying;
-}
+bool TSoundOutputDevice::isPlaying() const { return m_imp->m_isPlaying; }
//------------------------------------------------------------------------------
-bool TSoundOutputDevice::isLooping() {
- // TThread::ScopedLock sl(MutexOut);
- return m_imp->m_looped;
-}
+bool TSoundOutputDevice::isLooping() { return m_imp->m_looped; }
//------------------------------------------------------------------------------
-void TSoundOutputDevice::setLooping(bool loop) {
- // TThread::ScopedLock sl(MutexOut);
- m_imp->m_looped = loop;
-}
+void TSoundOutputDevice::setLooping(bool loop) { m_imp->m_looped = loop; }
//------------------------------------------------------------------------------
@@ -586,13 +337,8 @@ TSoundTrackFormat TSoundOutputDevice::getPreferredFormat(TUINT32 sampleRate,
TSoundTrackFormat TSoundOutputDevice::getPreferredFormat(
const TSoundTrackFormat &format) {
- // try {
return getPreferredFormat(format.m_sampleRate, format.m_channelCount,
format.m_bitPerSample);
- /*}
-catch (TSoundDeviceException &e) {
-throw TSoundDeviceException( e.getType(), e.getMessage());
-}*/
}
//==============================================================================
diff --git a/toonz/sources/include/tsound.h b/toonz/sources/include/tsound.h
index 75593ff..f635fa9 100644
--- a/toonz/sources/include/tsound.h
+++ b/toonz/sources/include/tsound.h
@@ -382,6 +382,7 @@ Returns true if on the machine there is an audio card installed correctly
//! Set the value of volume , between [0,1]
bool setVolume(double value);
+ void prepareVolume(double volume);
#endif
//! Open the device according to the features of soundtrack
diff --git a/toonz/sources/tnzcore/CMakeLists.txt b/toonz/sources/tnzcore/CMakeLists.txt
index 24820c4..a46db6d 100644
--- a/toonz/sources/tnzcore/CMakeLists.txt
+++ b/toonz/sources/tnzcore/CMakeLists.txt
@@ -353,7 +353,7 @@ elseif(BUILD_ENV_UNIXLIKE)
endif()
target_link_libraries(tnzcore
- Qt5::OpenGL Qt5::Network
+ Qt5::OpenGL Qt5::Network Qt5::Multimedia
${GL_LIB} ${GLUT_LIB} ${QT_LIB} ${Z_LIB} ${JPEG_LIB} ${LZ4_LIB}
${EXTRA_LIBS}
)
diff --git a/toonz/sources/toonzlib/CMakeLists.txt b/toonz/sources/toonzlib/CMakeLists.txt
index 9bc91f3..76b0a2a 100644
--- a/toonz/sources/toonzlib/CMakeLists.txt
+++ b/toonz/sources/toonzlib/CMakeLists.txt
@@ -351,7 +351,7 @@ include_directories(
if(BUILD_ENV_MSVC)
target_link_libraries(toonzlib
- Qt5::Core Qt5::Gui Qt5::OpenGL Qt5::Script
+ Qt5::Core Qt5::Gui Qt5::OpenGL Qt5::Script Qt5::Multimedia
${GLUT_LIB} ${GL_LIB} ${MYPAINT_LIB_LDFLAGS} vfw32.lib
tnzcore tnzbase tnzext
)
@@ -364,7 +364,7 @@ elseif(BUILD_ENV_APPLE)
${MYPAINT_LIB_LDFLAGS}
)
- target_link_libraries(toonzlib Qt5::Core Qt5::Gui Qt5::OpenGL Qt5::Script ${GLUT_LIB} ${GL_LIB} ${EXTRA_LIBS})
+ target_link_libraries(toonzlib Qt5::Core Qt5::Gui Qt5::OpenGL Qt5::Script Qt5::Multimedia ${GLUT_LIB} ${GL_LIB} ${EXTRA_LIBS})
elseif(BUILD_ENV_UNIXLIKE)
_find_toonz_library(EXTRA_LIBS "tnzcore;tnzbase;tnzext")
@@ -372,5 +372,5 @@ elseif(BUILD_ENV_UNIXLIKE)
set(EXTRA_LIBS ${EXTRA_LIBS} -lvfw32)
endif()
- target_link_libraries(toonzlib Qt5::Core Qt5::Gui Qt5::OpenGL Qt5::Script ${GLUT_LIB} ${GL_LIB} ${EXTRA_LIBS} ${MYPAINT_LIB_LDFLAGS})
+ target_link_libraries(toonzlib Qt5::Core Qt5::Gui Qt5::OpenGL Qt5::Script Qt5::Multimedia ${GLUT_LIB} ${GL_LIB} ${EXTRA_LIBS} ${MYPAINT_LIB_LDFLAGS})
endif()
diff --git a/toonz/sources/toonzlib/txshsoundcolumn.cpp b/toonz/sources/toonzlib/txshsoundcolumn.cpp
index 3a20c1b..98c2265 100644
--- a/toonz/sources/toonzlib/txshsoundcolumn.cpp
+++ b/toonz/sources/toonzlib/txshsoundcolumn.cpp
@@ -13,6 +13,9 @@
#include "tsop.h"
#include "tconvert.h"
+#include <QAudioFormat>
+#include <QAudioDeviceInfo>
+
//=============================================================================
// ColumnLevel
//=============================================================================
@@ -806,7 +809,7 @@ void TXshSoundColumn::play(TSoundTrackP soundtrack, int s0, int s1, bool loop) {
if (m_player) {
try {
#ifdef MACOSX
- m_player->setVolume(m_volume);
+ m_player->prepareVolume(m_volume);
TSoundTrackP mixedTrack = soundtrack;
#else
TSoundTrackP mixedTrack = TSoundTrack::create(
@@ -997,10 +1000,33 @@ TSoundTrackP TXshSoundColumn::getOverallSoundTrack(int fromFrame, int toFrame,
format.m_signedSample = true;
}
- // We prefer to have 22050 as a maximum sampleRate (to avoid crashes or
- // another issues)
+// We prefer to have 22050 as a maximum sampleRate (to avoid crashes or
+// another issues)
+#ifndef MACOSX
if (format.m_sampleRate >= 44100) format.m_sampleRate = 22050;
-
+#else
+ QAudioDeviceInfo info(QAudioDeviceInfo::defaultOutputDevice());
+ QList<int> ssrs = info.supportedSampleRates();
+ if (!ssrs.contains(format.m_sampleRate)) format.m_sampleRate = 44100;
+ QAudioFormat qFormat;
+ qFormat.setSampleRate(format.m_sampleRate);
+ qFormat.setSampleType(format.m_signedSample ? QAudioFormat::SignedInt
+ : QAudioFormat::UnSignedInt);
+ qFormat.setSampleSize(format.m_bitPerSample);
+ qFormat.setCodec("audio/pcm");
+ qFormat.setChannelCount(format.m_channelCount);
+ qFormat.setByteOrder(QAudioFormat::LittleEndian);
+ if (!info.isFormatSupported((qFormat))) {
+ qFormat = info.nearestFormat(qFormat);
+ format.m_bitPerSample = qFormat.sampleSize();
+ format.m_channelCount = qFormat.channelCount();
+ format.m_sampleRate = qFormat.sampleRate();
+ if (qFormat.sampleType() == QAudioFormat::SignedInt)
+ format.m_signedSample = true;
+ else
+ format.m_signedSample = false;
+ }
+#endif
// Create the soundTrack
double samplePerFrame = format.m_sampleRate / fps;