Skip to content

Commit

Permalink
Merge pull request #661 from j-santander/master
Browse files Browse the repository at this point in the history 
Initial working Android Code
  • Loading branch information
@corrados
corrados authored Oct 5, 2020
2 parents 9f4432b + d2aa4ee commit f62e058
Show file tree
Hide file tree
Showing 3 changed files with 90 additions and 110 deletions.
2 changes: 2 additions & 0 deletions Jamulus.pro
View file
Original file line number Diff line number Diff line change
Expand Up @@ -1019,3 +1019,5 @@ contains(CONFIG, "disable_version_check") {
message(The version check is disabled.)
DEFINES += DISABLE_VERSION_CHECK
}

ANDROID_ABIS = armeabi-v7a arm64-v8a
179 changes: 84 additions & 95 deletions android/sound.cpp
View file
Original file line number Diff line number Diff line change
Expand Up @@ -32,10 +32,8 @@ CSound::CSound ( void (*fpNewProcessCallback) ( CVector<short>& psData
const int iCtrlMIDIChannel,
const bool ,
const QString& ) :
CSoundBase ( "OpenSL", fpNewProcessCallback, arg, iCtrlMIDIChannel )
CSoundBase ( "Oboe", fpNewProcessCallback, arg, iCtrlMIDIChannel )
{
pSound = this;

#ifdef ANDROIDDEBUG
qInstallMessageHandler(myMessageHandler);
#endif
Expand All @@ -45,17 +43,36 @@ void CSound::setupCommonStreamParams(oboe::AudioStreamBuilder *builder)
{
// We request EXCLUSIVE mode since this will give us the lowest possible
// latency. If EXCLUSIVE mode isn't available the builder will fall back to SHARED mode
builder->setCallback(this)
builder
->setFormat(oboe::AudioFormat::Float)
->setSharingMode(oboe::SharingMode::Shared)
->setChannelCount(oboe::ChannelCount::Mono)
// ->setSampleRate(48000)
// ->setSampleRateConversionQuality(oboe::SampleRateConversionQuality::Medium)
->setPerformanceMode(oboe::PerformanceMode::None);
->setSharingMode(oboe::SharingMode::Exclusive)
->setChannelCount(oboe::ChannelCount::Stereo)
->setSampleRate(SYSTEM_SAMPLE_RATE_HZ)
->setFramesPerCallback(iOpenSLBufferSizeMono)
->setSampleRateConversionQuality(oboe::SampleRateConversionQuality::Medium)
->setPerformanceMode(oboe::PerformanceMode::LowLatency);

return;
}

void CSound::closeStream ( oboe::ManagedStream& stream )
{
if ( stream )
{
oboe::Result requestStopRes = stream->requestStop();
oboe::Result result = stream->close();

if ( result != oboe::Result::OK )
{
throw CGenErr ( tr ( "Error closing stream: $s",
oboe::convertToText ( result ) ) );
}

stream.reset();
}
}


void CSound::openStreams()
{
// Create callback
Expand All @@ -73,14 +90,17 @@ void CSound::openStreams()
{
return;
}

mPlayStream->setBufferSizeInFrames ( pSound->iOpenSLBufferSizeStereo );
mPlayStream->setBufferSizeInFrames ( iOpenSLBufferSizeStereo );

warnIfNotLowLatency ( mPlayStream, "PlayStream" );
printStreamDetails ( mPlayStream );

// Setup input stream
inBuilder.setDirection ( oboe::Direction::Input );

// Only set callback for the input direction
// the output will be handled writing directly on the stream
inBuilder.setCallback(this);
setupCommonStreamParams ( &inBuilder );

result = inBuilder.openManagedStream ( mRecordingStream );
Expand All @@ -91,7 +111,7 @@ void CSound::openStreams()
return;
}

mRecordingStream->setBufferSizeInFrames ( pSound->iOpenSLBufferSizeStereo );
mRecordingStream->setBufferSizeInFrames ( iOpenSLBufferSizeStereo );

warnIfNotLowLatency ( mRecordingStream, "RecordStream" );
printStreamDetails ( mRecordingStream );
Expand All @@ -110,10 +130,7 @@ void CSound::printStreamDetails ( oboe::ManagedStream& stream )
QString sDeviceID = QString::number ( stream->getDeviceId() );
QString sSampleRate = QString::number ( stream->getSampleRate() );
QString sAudioFormat = ( stream->getFormat()==oboe::AudioFormat::I16 ? "I16" : "Float" );

QString sFramesPerCallback = QString::number ( stream->getFramesPerCallback() );
//QString sSampleRateConversionQuality = (stream.getSampleRateConversionQuality()==oboe::SampleRateConversionQuality::

qInfo() << "Stream details: [sDirection: " << sDirection <<
", FramesPerBurst: " << sFramesPerBurst <<
", BufferSizeInFrames: " << sBufferSizeInFrames <<
Expand All @@ -136,23 +153,6 @@ void CSound::warnIfNotLowLatency ( oboe::ManagedStream& stream, QString streamNa
}
}

void CSound::closeStream ( oboe::ManagedStream& stream )
{
if ( stream )
{
oboe::Result requestStopRes = stream->requestStop();
oboe::Result result = stream->close();

if ( result != oboe::Result::OK )
{
throw CGenErr ( tr ( "Error closing stream: $s",
oboe::convertToText ( result ) ) );
}

stream.reset();
}
}

void CSound::closeStreams()
{
// clean up
Expand All @@ -170,7 +170,6 @@ void CSound::Start()
// finally start the streams so the callback begins, start with inputstream first.
mRecordingStream->requestStart();
mPlayStream->requestStart();

}

void CSound::Stop()
Expand All @@ -184,7 +183,7 @@ void CSound::Stop()
int CSound::Init ( const int iNewPrefMonoBufferSize )
{
// store buffer size
iOpenSLBufferSizeMono = 512; // iNewPrefMonoBufferSize;
iOpenSLBufferSizeMono = iNewPrefMonoBufferSize; // 512

// init base class
CSoundBase::Init ( iOpenSLBufferSizeMono );
Expand All @@ -193,106 +192,96 @@ int CSound::Init ( const int iNewPrefMonoBufferSize )
iOpenSLBufferSizeStereo = 2 * iOpenSLBufferSizeMono;

// create memory for intermediate audio buffer
vecsTmpAudioSndCrdStereo.Init ( iOpenSLBufferSizeStereo );

// TEST
#if ( SYSTEM_SAMPLE_RATE_HZ != 48000 )
# error "Only a system sample rate of 48 kHz is supported by this module"
#endif
// audio interface number of channels is 1 and the sample rate
// is 16 kHz -> just copy samples and perform no filtering as a
// first simple solution
// 48 kHz / 16 kHz = factor 3 (note that the buffer size mono might
// be divisible by three, therefore we will get a lot of drop outs)
iModifiedInBufSize = iOpenSLBufferSizeMono / 3;
vecsTmpAudioInSndCrd.Init ( iModifiedInBufSize );
vecsTmpInputAudioSndCrdStereo.Init ( iOpenSLBufferSizeStereo );
vecsTmpOutAudioSndCrdStereo.Init ( iOpenSLBufferSizeStereo );

return iOpenSLBufferSizeMono;
}

// This is the main callback method for when an audio stream is ready to publish data to an output stream
// or has received data on an input stream. As per manual much be very careful not to do anything in this back that
// can cause delays such as sleeping, file processing, allocate memory, etc
oboe::DataCallbackResult CSound::onAudioReady ( oboe::AudioStream* oboeStream, void* audioData, int32_t numFrames )
// or has received data on an input stream. As per manual much be very careful not to do anything in this back that
// can cause delays such as sleeping, file processing, allocate memory, etc.
oboe::DataCallbackResult CSound::onAudioReady ( oboe::AudioStream* oboeStream,
void* audioData,
int32_t numFrames )
{
// only process if we are running
if ( ! pSound->bRun )
if ( ! bRun )
{
return oboe::DataCallbackResult::Continue;
}

// Need to modify the size of the buffer based on the numFrames requested in this callback.
// Buffer size can change regularly by android devices
int& iBufferSizeMono = pSound->iOpenSLBufferSizeMono;
if ( oboeStream == mRecordingStream.get() && audioData )
{
// First things first, we need to discard the input queue a little for 500ms or so
if ( mCountCallbacksToDrain > 0 )
{
// discard the input buffer
int32_t numBytes = numFrames * oboeStream->getBytesPerFrame();
memset ( audioData, 0 /* value */, numBytes );
mCountCallbacksToDrain--;
}

// perform the processing for input and output
// QMutexLocker locker ( &pSound->Mutex );
// locker.mutex();
// We're good to start recording now
// Take the data from the recording device output buffer and move
// it to the vector ready to send up to the server
float* floatData = static_cast<float*> ( audioData );

// This can be called from both input and output at different times
if ( oboeStream == pSound->mPlayStream.get() && audioData )
{
float *floatData = static_cast<float*> ( audioData );
// Copy recording data to internal vector
for ( int frmNum = 0; frmNum < numFrames; ++frmNum )
{
for ( int channelNum = 0; channelNum < oboeStream->getChannelCount(); channelNum++ )
{
vecsTmpInputAudioSndCrdStereo[frmNum * oboeStream->getChannelCount() + channelNum] =
static_cast<int16_t>(floatData[frmNum * oboeStream->getChannelCount() + channelNum] * _MAXSHORT);
}
}

// Tell parent class that we've put some data ready to send to the server
ProcessCallback ( vecsTmpInputAudioSndCrdStereo );

// Zero out the incoming container array
memset ( audioData, 0, sizeof(float) * numFrames * oboeStream->getChannelCount() );
// The callback has placed in the vector the samples to play

// Only copy data if we have data to copy, otherwise fill with silence
if ( !pSound->vecsTmpAudioSndCrdStereo.empty() )
if ( !vecsTmpInputAudioSndCrdStereo.empty() )
{
for ( int frmNum = 0; frmNum < numFrames; ++frmNum )
for ( int frmNum = 0; frmNum < iOpenSLBufferSizeMono; ++frmNum )
{
for ( int channelNum = 0; channelNum < oboeStream->getChannelCount(); channelNum++ )
{
// copy sample received from server into output buffer

// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
pSound->vecsTmpAudioSndCrdStereo [frmNum * oboeStream->getChannelCount() + channelNum] );
vecsTmpInputAudioSndCrdStereo[frmNum * oboeStream->getChannelCount() + channelNum] );

floatData[frmNum * oboeStream->getChannelCount() + channelNum] = (float) iCurSam / _MAXSHORT;
vecsTmpOutAudioSndCrdStereo[frmNum * oboeStream->getChannelCount() + channelNum] = ((float) iCurSam) / _MAXSHORT;
}
}
}
else
{
// prime output stream buffer with silence
memset(static_cast<float*> ( audioData ) + numFrames * oboeStream->getChannelCount(), 0,
( numFrames ) * oboeStream->getBytesPerFrame() );
}
}
else if ( oboeStream == pSound->mRecordingStream.get() && audioData )
{
// First things first, we need to discard the input queue a little for 500ms or so
if ( pSound->mCountCallbacksToDrain > 0 )
{
// discard the input buffer
int32_t numBytes = numFrames * oboeStream->getBytesPerFrame();
memset ( audioData, 0 /* value */, numBytes );
pSound->mCountCallbacksToDrain--;
vecsTmpOutAudioSndCrdStereo.Reset(0);
}

// We're good to start recording now
// Take the data from the recording device output buffer and move
// it to the vector ready to send up to the server
float *floatData = static_cast<float*> ( audioData );

// Copy recording data to internal vector
for ( int frmNum = 0; frmNum < numFrames; ++frmNum )
// Write without blocking.
oboe::ResultWithValue<int32_t> r = mPlayStream->write(vecsTmpOutAudioSndCrdStereo.data(),iOpenSLBufferSizeMono,0);
if(r)
{
for ( int channelNum = 0; channelNum < oboeStream->getChannelCount(); channelNum++ )
// Note: this could result in not writing the requested number of frames.
// We could buffer them and wait for the next occasion, but for now we just discard them.
if(r.value() < iOpenSLBufferSizeMono)
{
pSound->vecsTmpAudioSndCrdStereo[frmNum * oboeStream->getChannelCount() + channelNum] =
(short) floatData[frmNum * oboeStream->getChannelCount() + channelNum] * _MAXSHORT;
qDebug() << "Could only write " << r.value() << " frames of the " << iOpenSLBufferSizeMono << " requested.";
}
}

// Tell parent class that we've put some data ready to send to the server
pSound->ProcessCallback ( pSound->vecsTmpAudioSndCrdStereo );
else
{
qDebug() << "Could not write to output stream: " << oboe::convertToText(r.error());
}
}

// locker.unlock();

return oboe::DataCallbackResult::Continue;
}

Expand Down
19 changes: 4 additions & 15 deletions android/sound.h
View file
Original file line number Diff line number Diff line change
Expand Up @@ -25,15 +25,13 @@
#pragma once

#include <oboe/Oboe.h>
#include <QMutex>
#include "soundbase.h"
#include "global.h"
#include <QDebug>
#include <android/log.h>


/* Classes ********************************************************************/
class CSound : public CSoundBase, public oboe::AudioStreamCallback//, public IRenderableAudio, public IRestartable
class CSound : public CSoundBase, public oboe::AudioStreamCallback
{
public:
CSound ( void (*fpNewProcessCallback) ( CVector<short>& psData, void* arg ),
Expand All @@ -52,9 +50,9 @@ class CSound : public CSoundBase, public oboe::AudioStreamCallback//, public IRe
virtual void onErrorAfterClose ( oboe::AudioStream* oboeStream, oboe::Result result );
virtual void onErrorBeforeClose ( oboe::AudioStream* oboeStream, oboe::Result result );

// these variables should be protected but cannot since we want
// to access them from the callback function
CVector<short> vecsTmpAudioSndCrdStereo;
protected:
CVector<short> vecsTmpInputAudioSndCrdStereo;
CVector<float> vecsTmpOutAudioSndCrdStereo;

static void android_message_handler ( QtMsgType type,
const QMessageLogContext& context,
Expand All @@ -73,10 +71,6 @@ class CSound : public CSoundBase, public oboe::AudioStreamCallback//, public IRe
__android_log_print ( priority, "Qt", "%s", qPrintable ( message ) );
};

// TEST
CVector<short> vecsTmpAudioInSndCrd;
int iModifiedInBufSize;

int iOpenSLBufferSizeMono;
int iOpenSLBufferSizeStereo;

Expand All @@ -96,9 +90,4 @@ int iModifiedInBufSize;
// empty and the garbage in the first 500ms or so is discarded
static constexpr int32_t kNumCallbacksToDrain = 10;
int32_t mCountCallbacksToDrain = kNumCallbacksToDrain;

// Used to reference this instance of class from within the static callback
CSound *pSound;

QMutex Mutex;
};

0 comments on commit f62e058

Please sign in to comment.