Various changes for true 24-bit support, changes regarding Jack get_latency, and Linux ALSA realtime threading (gps).

1 files changed, 56 insertions, 30 deletions

diff --git a/RtAudio.cpp b/RtAudio.cpp
index 823faaf..8477ef9 100644
--- a/RtAudio.cpp
+++ b/RtAudio.cpp
@@ -2094,8 +2094,17 @@ bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
 

   // Get the latency of the JACK port.

   ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );

-  if ( ports[ firstChannel ] )

-    stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );

+  if ( ports[ firstChannel ] ) {

+    // Added by Ge Wang

+    jack_latency_callback_mode_t cbmode = (mode == INPUT ? JackCaptureLatency : JackPlaybackLatency);

+    // the range (usually the min and max are equal)

+    jack_latency_range_t latrange; latrange.min = latrange.max = 0;

+    // get the latency range

+    jack_port_get_latency_range( jack_port_by_name( client, ports[firstChannel] ), cbmode, &latrange );

+    // be optimistic, use the min!

+    stream_.latency[mode] = latrange.min;

+    //stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );

+  }

   free( ports );

 

   // The jack server always uses 32-bit floating-point data.

@@ -2708,6 +2717,8 @@ RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device )
     info.nativeFormats |= RTAUDIO_FLOAT32;

   else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB )

     info.nativeFormats |= RTAUDIO_FLOAT64;

+  else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB )

+    info.nativeFormats |= RTAUDIO_SINT24;

 

   if ( info.outputChannels > 0 )

     if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;

@@ -2860,6 +2871,10 @@ bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
     stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;

     if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true;

   }

+  else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) {

+    stream_.deviceFormat[mode] = RTAUDIO_SINT24;

+    if ( channelInfo.type == ASIOSTInt24MSB ) stream_.doByteSwap[mode] = true;

+  }

 

   if ( stream_.deviceFormat[mode] == 0 ) {

     drivers.removeCurrentDriver();

@@ -3879,7 +3894,7 @@ bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned
   // Determine the device buffer size. By default, we'll use the value

   // defined above (32K), but we will grow it to make allowances for

   // very large software buffer sizes.

-  DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE;;

+  DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE;

   DWORD dsPointerLeadTime = 0;

 

   void *ohandle = 0, *bhandle = 0;

@@ -5875,8 +5890,8 @@ bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
       if ( priority < min ) priority = min;

       else if ( priority > max ) priority = max;

       param.sched_priority = priority;

-      pthread_attr_setschedparam( &attr, &param );

       pthread_attr_setschedpolicy( &attr, SCHED_RR );

+      pthread_attr_setschedparam( &attr, &param );

     }

     else

       pthread_attr_setschedpolicy( &attr, SCHED_OTHER );

@@ -7741,11 +7756,12 @@ unsigned int RtApi :: formatBytes( RtAudioFormat format )
 {

   if ( format == RTAUDIO_SINT16 )

     return 2;

-  else if ( format == RTAUDIO_SINT24 || format == RTAUDIO_SINT32 ||

-            format == RTAUDIO_FLOAT32 )

+  else if ( format == RTAUDIO_SINT32 || format == RTAUDIO_FLOAT32 )

     return 4;

   else if ( format == RTAUDIO_FLOAT64 )

     return 8;

+  else if ( format == RTAUDIO_SINT24 )

+    return 3;

   else if ( format == RTAUDIO_SINT8 )

     return 1;

 

@@ -7878,11 +7894,11 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info
       }

     }

     else if (info.inFormat == RTAUDIO_SINT24) {

-      Int32 *in = (Int32 *)inBuffer;

+      Int24 *in = (Int24 *)inBuffer;

       scale = 1.0 / 8388607.5;

       for (unsigned int i=0; i<stream_.bufferSize; i++) {

         for (j=0; j<info.channels; j++) {

-          out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]] & 0x00ffffff);

+          out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]].asInt());

           out[info.outOffset[j]] += 0.5;

           out[info.outOffset[j]] *= scale;

         }

@@ -7956,11 +7972,11 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info
       }

     }

     else if (info.inFormat == RTAUDIO_SINT24) {

-      Int32 *in = (Int32 *)inBuffer;

+      Int24 *in = (Int24 *)inBuffer;

       scale = (Float32) ( 1.0 / 8388607.5 );

       for (unsigned int i=0; i<stream_.bufferSize; i++) {

         for (j=0; j<info.channels; j++) {

-          out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]] & 0x00ffffff);

+          out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]].asInt());

           out[info.outOffset[j]] += 0.5;

           out[info.outOffset[j]] *= scale;

         }

@@ -8027,11 +8043,11 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info
         out += info.outJump;

       }

     }

-    else if (info.inFormat == RTAUDIO_SINT24) { // Hmmm ... we could just leave it in the lower 3 bytes

-      Int32 *in = (Int32 *)inBuffer;

+    else if (info.inFormat == RTAUDIO_SINT24) {

+      Int24 *in = (Int24 *)inBuffer;

       for (unsigned int i=0; i<stream_.bufferSize; i++) {

         for (j=0; j<info.channels; j++) {

-          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];

+          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]].asInt();

           out[info.outOffset[j]] <<= 8;

         }

         in += info.inJump;

@@ -8071,13 +8087,13 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info
     }

   }

   else if (info.outFormat == RTAUDIO_SINT24) {

-    Int32 *out = (Int32 *)outBuffer;

+    Int24 *out = (Int24 *)outBuffer;

     if (info.inFormat == RTAUDIO_SINT8) {

       signed char *in = (signed char *)inBuffer;

       for (unsigned int i=0; i<stream_.bufferSize; i++) {

         for (j=0; j<info.channels; j++) {

-          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];

-          out[info.outOffset[j]] <<= 16;

+          out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 16);

+          //out[info.outOffset[j]] <<= 16;

         }

         in += info.inJump;

         out += info.outJump;

@@ -8087,8 +8103,8 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info
       Int16 *in = (Int16 *)inBuffer;

       for (unsigned int i=0; i<stream_.bufferSize; i++) {

         for (j=0; j<info.channels; j++) {

-          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];

-          out[info.outOffset[j]] <<= 8;

+          out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 8);

+          //out[info.outOffset[j]] <<= 8;

         }

         in += info.inJump;

         out += info.outJump;

@@ -8096,7 +8112,7 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info
     }

     else if (info.inFormat == RTAUDIO_SINT24) {

       // Channel compensation and/or (de)interleaving only.

-      Int32 *in = (Int32 *)inBuffer;

+      Int24 *in = (Int24 *)inBuffer;

       for (unsigned int i=0; i<stream_.bufferSize; i++) {

         for (j=0; j<info.channels; j++) {

           out[info.outOffset[j]] = in[info.inOffset[j]];

@@ -8109,8 +8125,8 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info
       Int32 *in = (Int32 *)inBuffer;

       for (unsigned int i=0; i<stream_.bufferSize; i++) {

         for (j=0; j<info.channels; j++) {

-          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];

-          out[info.outOffset[j]] >>= 8;

+          out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] >> 8);

+          //out[info.outOffset[j]] >>= 8;

         }

         in += info.inJump;

         out += info.outJump;

@@ -8162,10 +8178,10 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info
       }

     }

     else if (info.inFormat == RTAUDIO_SINT24) {

-      Int32 *in = (Int32 *)inBuffer;

+      Int24 *in = (Int24 *)inBuffer;

       for (unsigned int i=0; i<stream_.bufferSize; i++) {

         for (j=0; j<info.channels; j++) {

-          out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 8) & 0x0000ffff);

+          out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]].asInt() >> 8);

         }

         in += info.inJump;

         out += info.outJump;

@@ -8226,10 +8242,10 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info
       }

     }

     else if (info.inFormat == RTAUDIO_SINT24) {

-      Int32 *in = (Int32 *)inBuffer;

+      Int24 *in = (Int24 *)inBuffer;

       for (unsigned int i=0; i<stream_.bufferSize; i++) {

         for (j=0; j<info.channels; j++) {

-          out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 16) & 0x000000ff);

+          out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]].asInt() >> 16);

         }

         in += info.inJump;

         out += info.outJump;

@@ -8268,9 +8284,9 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info
   }

 }

 

-  //static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }

-  //static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }

-  //static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }

+//static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }

+//static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }

+//static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }

 

 void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )

 {

@@ -8289,8 +8305,7 @@ void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat
       ptr += 2;

     }

   }

-  else if ( format == RTAUDIO_SINT24 ||

-            format == RTAUDIO_SINT32 ||

+  else if ( format == RTAUDIO_SINT32 ||

             format == RTAUDIO_FLOAT32 ) {

     for ( unsigned int i=0; i<samples; i++ ) {

       // Swap 1st and 4th bytes.

@@ -8308,6 +8323,17 @@ void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat
       ptr += 3;

     }

   }

+  else if ( format == RTAUDIO_SINT24 ) {

+    for ( unsigned int i=0; i<samples; i++ ) {

+      // Swap 1st and 3rd bytes.

+      val = *(ptr);

+      *(ptr) = *(ptr+2);

+      *(ptr+2) = val;

+

+      // Increment 2 more bytes.

+      ptr += 2;

+    }

+  }

   else if ( format == RTAUDIO_FLOAT64 ) {

     for ( unsigned int i=0; i<samples; i++ ) {

       // Swap 1st and 8th bytes