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/*
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This class handles both RX and TX audio , each is created as a seperate instance of the class
but as the setup / handling if output ( RX ) and input ( TX ) devices is so similar I have combined them .
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*/
# include "audiohandler.h"
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# include "logcategories.h"
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# include "ulaw.h"
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audioHandler : : audioHandler ( QObject * parent ) :
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isInitialized ( false ) ,
isUlaw ( false ) ,
audioLatency ( 0 ) ,
isInput ( 0 ) ,
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chunkAvailable ( false )
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{
}
audioHandler : : ~ audioHandler ( )
{
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//stop();
if ( resampler ! = Q_NULLPTR ) {
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speex_resampler_destroy ( resampler ) ;
}
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if ( audio . isStreamRunning ( ) )
{
audio . stopStream ( ) ;
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audio . closeStream ( ) ;
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}
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if ( ringBuf ! = Q_NULLPTR )
delete ringBuf ;
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}
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bool audioHandler : : init ( const quint8 bits , const quint8 channels , const quint16 samplerate , const quint16 latency , const bool ulaw , const bool isinput , int port , quint8 resampleQuality )
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{
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if ( isInitialized ) {
return false ;
}
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this - > audioLatency = latency ;
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this - > isUlaw = ulaw ;
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this - > isInput = isinput ;
this - > radioSampleBits = bits ;
this - > radioSampleRate = samplerate ;
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this - > radioChannels = channels ;
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// chunk size is always relative to Internal Sample Rate.
this - > chunkSize = ( INTERNAL_SAMPLE_RATE / 25 ) * radioChannels ;
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ringBuf = new wilt : : Ring < audioPacket > ( 100 ) ; // Should be customizable.
tempBuf . sent = 0 ;
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if ( port > 0 ) {
aParams . deviceId = port ;
}
else if ( isInput ) {
aParams . deviceId = audio . getDefaultInputDevice ( ) ;
}
else {
aParams . deviceId = audio . getDefaultOutputDevice ( ) ;
}
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aParams . nChannels = 2 ; // Internally this is always 2 channels
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aParams . firstChannel = 0 ;
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try {
info = audio . getDeviceInfo ( aParams . deviceId ) ;
}
catch ( RtAudioError & e ) {
qInfo ( logAudio ( ) ) < < " Device error: " < < aParams . deviceId < < " : " < < QString : : fromStdString ( e . getMessage ( ) ) ;
return false ;
}
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if ( info . probed )
{
qInfo ( logAudio ( ) ) < < ( isInput ? " Input " : " Output " ) < < QString : : fromStdString ( info . name ) < < " ( " < < aParams . deviceId < < " ) successfully probed " ;
if ( info . nativeFormats = = 0 )
qInfo ( logAudio ( ) ) < < " No natively supported data formats! " ;
else {
qDebug ( logAudio ( ) ) < < " Supported formats: " < <
( info . nativeFormats & RTAUDIO_SINT8 ? " 8-bit int, " : " " ) < <
( info . nativeFormats & RTAUDIO_SINT16 ? " 16-bit int, " : " " ) < <
( info . nativeFormats & RTAUDIO_SINT24 ? " 24-bit int, " : " " ) < <
( info . nativeFormats & RTAUDIO_SINT32 ? " 32-bit int, " : " " ) < <
( info . nativeFormats & RTAUDIO_FLOAT32 ? " 32-bit float, " : " " ) < <
( info . nativeFormats & RTAUDIO_FLOAT64 ? " 64-bit float, " : " " ) ;
qInfo ( logAudio ( ) ) < < " Preferred sample rate: " < < info . preferredSampleRate ;
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}
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qInfo ( logAudio ( ) ) < < " chunkSize: " < < chunkSize ;
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}
else
{
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qCritical ( logAudio ( ) ) < < ( isInput ? " Input " : " Output " ) < < QString : : fromStdString ( info . name ) < < " ( " < < aParams . deviceId < < " ) could not be probed, check audio configuration! " ;
}
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int resample_error = 0 ;
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if ( isInput ) {
resampler = wf_resampler_init ( radioChannels , INTERNAL_SAMPLE_RATE , samplerate , resampleQuality , & resample_error ) ;
try {
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audio . openStream ( NULL , & aParams , RTAUDIO_SINT16 , INTERNAL_SAMPLE_RATE , & this - > chunkSize , & staticWrite , this ) ;
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audio . startStream ( ) ;
}
catch ( RtAudioError & e ) {
qInfo ( logAudio ( ) ) < < " Error opening: " < < QString : : fromStdString ( e . getMessage ( ) ) ;
return false ;
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}
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}
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else
{
resampler = wf_resampler_init ( radioChannels , samplerate , INTERNAL_SAMPLE_RATE , resampleQuality , & resample_error ) ;
try {
unsigned int length = chunkSize / 2 ;
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audio . openStream ( & aParams , NULL , RTAUDIO_SINT16 , INTERNAL_SAMPLE_RATE , & length , & staticRead , this ) ;
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audio . startStream ( ) ;
}
catch ( RtAudioError & e ) {
qInfo ( logAudio ( ) ) < < " Error opening: " < < QString : : fromStdString ( e . getMessage ( ) ) ;
return false ;
}
}
qInfo ( logAudio ( ) ) < < ( isInput ? " Input " : " Output " ) < < " device successfully opened " ;
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wf_resampler_get_ratio ( resampler , & ratioNum , & ratioDen ) ;
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qInfo ( logAudio ( ) ) < < ( isInput ? " Input " : " Output " ) < < " wf_resampler_init() returned: " < < resample_error < < " ratioNum " < < ratioNum < < " ratioDen " < < ratioDen ;
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return isInitialized ;
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}
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void audioHandler : : setVolume ( unsigned char volume )
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{
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qInfo ( logAudio ( ) ) < < ( isInput ? " Input " : " Output " ) < < " setVolume: " < < volume < < " ( " < < ( qreal ) ( volume / 255.0 ) < < " ) " ;
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}
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/// <summary>
/// This function processes the incoming audio FROM the radio and pushes it into the playback buffer *data
/// </summary>
/// <param name="data"></param>
/// <param name="maxlen"></param>
/// <returns></returns>
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int audioHandler : : readData ( void * outputBuffer , void * inputBuffer , unsigned int nFrames , double streamTime , RtAudioStreamStatus status )
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{
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// Calculate output length, always full samples
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int sentlen = 0 ;
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quint8 * buffer = ( quint8 * ) outputBuffer ;
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if ( status = = RTAUDIO_OUTPUT_UNDERFLOW )
qDebug ( logAudio ( ) ) < < " Underflow detected " ;
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unsigned int nBytes = nFrames * 2 * 2 ; // This is ALWAYS 2 bytes per sample and 2 channels
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if ( ringBuf - > size ( ) > 0 )
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{
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// Output buffer is ALWAYS 16 bit.
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while ( sentlen < nBytes )
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{
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audioPacket packet ;
if ( ! ringBuf - > try_read ( packet ) )
{
qDebug ( ) < < " No more data available but buffer is not full! sentlen: " < < sentlen < < " nBytes: " < < nBytes ;
return 0 ;
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}
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currentLatency = packet . time . msecsTo ( QTime : : currentTime ( ) ) ;
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// This shouldn't be required but if we did output a partial packet
// This will add the remaining packet data to the output buffer.
if ( tempBuf . sent ! = tempBuf . data . length ( ) )
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{
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int send = qMin ( ( int ) nBytes - sentlen , tempBuf . data . length ( ) - tempBuf . sent ) ;
memcpy ( buffer + sentlen , tempBuf . data . constData ( ) + tempBuf . sent , send ) ;
tempBuf . sent = tempBuf . sent + send ;
sentlen = sentlen + send ;
qDebug ( logAudio ( ) ) < < " Adding partial: " < < send ;
}
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while ( currentLatency > ( int ) audioLatency ) {
qInfo ( logAudio ( ) ) < < ( isInput ? " Input " : " Output " ) < < " Packet " < < hex < < packet . seq < <
" arrived too late (increase output latency!) " < <
dec < < packet . time . msecsTo ( QTime : : currentTime ( ) ) < < " ms " ;
lastSeq = packet . seq ;
if ( ! ringBuf - > try_read ( packet ) )
return sentlen ;
currentLatency = packet . time . msecsTo ( QTime : : currentTime ( ) ) ;
}
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int send = qMin ( ( int ) nBytes - sentlen , packet . data . length ( ) ) ;
memcpy ( buffer + sentlen , packet . data . constData ( ) , send ) ;
sentlen = sentlen + send ;
if ( send < packet . data . length ( ) )
{
qDebug ( logAudio ( ) ) < < " Asking for partial, sent: " < < send < < " packet length " < < packet . data . length ( ) ;
tempBuf = packet ;
tempBuf . sent = tempBuf . sent + send ;
//lastSeq = packet.seq;
//break;
}
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if ( packet . seq < = lastSeq ) {
qInfo ( logAudio ( ) ) < < ( isInput ? " Input " : " Output " ) < < " Duplicate/early audio packet: " < < hex < < lastSeq < < " got " < < hex < < packet . seq ;
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}
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else if ( packet . seq ! = lastSeq + 1 ) {
qInfo ( logAudio ( ) ) < < ( isInput ? " Input " : " Output " ) < < " Missing audio packet(s) from: " < < hex < < lastSeq + 1 < < " to " < < hex < < packet . seq - 1 ;
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}
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lastSeq = packet . seq ;
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}
}
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//qDebug(logAudio()) << "looking for: " << nBytes << " got: " << sentlen;
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return 0 ;
}
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int audioHandler : : writeData ( void * outputBuffer , void * inputBuffer , unsigned int nFrames , double streamTime , RtAudioStreamStatus status )
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{
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int sentlen = 0 ;
/*
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QMutexLocker locker ( & mutex ) ;
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audioPacket * current ;
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while ( sentlen < len ) {
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if ( ! audioBuffer . isEmpty ( ) )
{
if ( audioBuffer . last ( ) . sent = = chunkSize )
{
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audioBuffer . append ( audioPacket ( ) ) ;
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audioBuffer . last ( ) . sent = 0 ;
}
}
else
{
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audioBuffer . append ( audioPacket ( ) ) ;
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audioBuffer . last ( ) . sent = 0 ;
}
current = & audioBuffer . last ( ) ;
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int send = qMin ( ( int ) ( len - sentlen ) , ( int ) chunkSize - current - > sent ) ;
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current - > data . append ( QByteArray : : fromRawData ( data + sentlen , send ) ) ;
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sentlen = sentlen + send ;
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current - > seq = 0 ; // Not used in TX
current - > time = QTime : : currentTime ( ) ;
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current - > sent = current - > data . length ( ) ;
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if ( current - > sent = = chunkSize )
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{
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chunkAvailable = true ;
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}
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else if ( audioBuffer . length ( ) < = 1 & & current - > sent ! = chunkSize ) {
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chunkAvailable = false ;
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}
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}
*/
return ( sentlen ) ; // Always return the same number as we received
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}
qint64 audioHandler : : bytesAvailable ( ) const
{
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return 0 ;
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}
bool audioHandler : : isSequential ( ) const
{
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return true ;
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}
void audioHandler : : notified ( )
{
}
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void audioHandler : : stateChanged ( QAudio : : State state )
{
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}
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int audioHandler : : incomingAudio ( audioPacket data )
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{
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// No point buffering audio until stream is actually running.
if ( ! audio . isStreamRunning ( ) )
{
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qDebug ( logAudio ( ) ) < < " Packet received before stream was started " ;
return currentLatency ;
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}
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// Incoming data is 8bits?
if ( radioSampleBits = = 8 )
{
QByteArray outPacket ( ( int ) data . data . length ( ) * 2 , ( char ) 0xff ) ;
qint16 * out = ( qint16 * ) outPacket . data ( ) ;
for ( int f = 0 ; f < data . data . length ( ) ; f + + )
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{
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if ( isUlaw )
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{
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out [ f ] = ulaw_decode [ ( quint8 ) data . data [ f ] ] ;
}
else
{
// Convert 8-bit sample to 16-bit
out [ f ] = ( qint16 ) ( ( ( quint8 ) data . data [ f ] < < 8 ) - 32640 ) ;
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}
}
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data . data . clear ( ) ;
data . data = outPacket ; // Replace incoming data with converted.
}
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//qInfo(logAudio()) << "Adding packet to buffer:" << data.seq << ": " << data.data.length();
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/* We now have an array of 16bit samples in the NATIVE samplerate of the radio
If the radio sample rate is below 48000 , we need to resample .
*/
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if ( ratioDen ! = 1 ) {
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// We need to resample
quint32 outFrames = ( ( data . data . length ( ) / 2 ) * ratioDen ) / radioChannels ;
quint32 inFrames = ( data . data . length ( ) / 2 ) / radioChannels ;
QByteArray outPacket ( outFrames * 2 * radioChannels , 0xff ) ; // Preset the output buffer size.
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int err = 0 ;
if ( this - > radioChannels = = 1 ) {
err = wf_resampler_process_int ( resampler , 0 , ( const qint16 * ) data . data . constData ( ) , & inFrames , ( qint16 * ) outPacket . data ( ) , & outFrames ) ;
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}
else {
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err = wf_resampler_process_interleaved_int ( resampler , ( const qint16 * ) data . data . constData ( ) , & inFrames , ( qint16 * ) outPacket . data ( ) , & outFrames ) ;
}
if ( err ) {
qInfo ( logAudio ( ) ) < < ( isInput ? " Input " : " Output " ) < < " Resampler error " < < err < < " inFrames: " < < inFrames < < " outFrames: " < < outFrames ;
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}
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data . data . clear ( ) ;
data . data = outPacket ; // Replace incoming data with converted.
}
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if ( radioChannels = = 1 )
{
// Convert to stereo
QByteArray outPacket ( data . data . length ( ) * 2 , 0xff ) ; // Preset the output buffer size.
qint16 * in = ( qint16 * ) data . data . data ( ) ;
qint16 * out = ( qint16 * ) outPacket . data ( ) ;
for ( int f = 0 ; f < data . data . length ( ) / 2 ; f + + )
{
* out + + = * in ;
* out + + = * in + + ;
}
data . data . clear ( ) ;
data . data = outPacket ; // Replace incoming data with converted.
}
if ( ! ringBuf - > try_write ( data ) )
{
qDebug ( logAudio ( ) ) < < " Buffer full! capacity: " < < ringBuf - > capacity ( ) < < " length " < < ringBuf - > size ( ) ;
}
return currentLatency ;
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}
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void audioHandler : : changeLatency ( const quint16 newSize )
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{
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qInfo ( logAudio ( ) ) < < ( isInput ? " Input " : " Output " ) < < " Changing latency to: " < < newSize < < " from " < < audioLatency ;
audioLatency = newSize ;
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}
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void audioHandler : : getLatency ( )
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{
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emit sendLatency ( audioLatency ) ;
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}
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bool audioHandler : : isChunkAvailable ( )
{
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return ( chunkAvailable ) ;
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}
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void audioHandler : : getNextAudioChunk ( QByteArray & ret )
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{
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/*
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if ( ! audioBuffer . isEmpty ( ) & & chunkAvailable )
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{
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QMutexLocker locker ( & mutex ) ;
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// Skip through audio buffer deleting any old entry.
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auto packet = audioBuffer . begin ( ) ;
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while ( packet ! = audioBuffer . end ( ) )
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{
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if ( packet - > time . msecsTo ( QTime : : currentTime ( ) ) > 100 ) {
//qInfo(logAudio()) << "TX Packet too old " << dec << packet->time.msecsTo(QTime::currentTime()) << "ms";
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packet = audioBuffer . erase ( packet ) ; // returns next packet
}
else {
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if ( packet - > data . length ( ) = = chunkSize & & ret . length ( ) = = 0 )
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{
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// We now have an array of samples in the computer native format (48000)
// If the radio sample rate is below 48000, we need to resample.
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if ( ratioNum ! = 1 )
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{
// We need to resample (we are STILL 16 bit!)
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quint32 outFrames = ( ( packet - > data . length ( ) / 2 ) / ratioNum ) / radioChannels ;
quint32 inFrames = ( packet - > data . length ( ) / 2 ) / radioChannels ;
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packet - > dataout . resize ( outFrames * 2 * radioChannels ) ; // Preset the output buffer size.
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const qint16 * in = ( qint16 * ) packet - > data . constData ( ) ;
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qint16 * out = ( qint16 * ) packet - > dataout . data ( ) ;
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int err = 0 ;
if ( this - > radioChannels = = 1 ) {
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err = wf_resampler_process_int ( resampler , 0 , in , & inFrames , out , & outFrames ) ;
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}
else {
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err = wf_resampler_process_interleaved_int ( resampler , in , & inFrames , out , & outFrames ) ;
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}
if ( err ) {
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qInfo ( logAudio ( ) ) < < ( isInput ? " Input " : " Output " ) < < " Resampler error " < < err < < " inFrames: " < < inFrames < < " outFrames: " < < outFrames ;
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}
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//qInfo(logAudio()) << "Resampler run " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
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//qInfo(logAudio()) << "Resampler run inLen:" << packet->data.length() << " outLen:" << packet->dataout.length();
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if ( radioSampleBits = = 8 )
{
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packet - > data = packet - > dataout ; // Copy output packet back to input buffer.
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packet - > dataout . clear ( ) ; // Buffer MUST be cleared ready to be re-filled by the upsampling below.
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}
}
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else if ( radioSampleBits = = 16 ) {
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// Only copy buffer if radioSampleBits is 16, as it will be handled below otherwise.
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packet - > dataout = packet - > data ;
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}
// Do we need to convert 16-bit to 8-bit?
if ( radioSampleBits = = 8 ) {
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packet - > dataout . resize ( packet - > data . length ( ) / 2 ) ;
qint16 * in = ( qint16 * ) packet - > data . data ( ) ;
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for ( int f = 0 ; f < packet - > dataout . length ( ) ; f + + )
{
quint8 outdata = 0 ;
if ( isUlaw ) {
qint16 enc = qFromLittleEndian < quint16 > ( in + f ) ;
if ( enc > = 0 )
outdata = ulaw_encode [ enc ] ;
else
outdata = 0x7f & ulaw_encode [ - enc ] ;
}
else {
outdata = ( quint8 ) ( ( ( qFromLittleEndian < qint16 > ( in + f ) > > 8 ) ^ 0x80 ) & 0xff ) ;
}
packet - > dataout [ f ] = ( char ) outdata ;
}
}
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ret = packet - > dataout ;
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packet = audioBuffer . erase ( packet ) ; // returns next packet
}
else {
packet + + ;
}
}
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}
}
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*/
return ;
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}
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