X-Git-Url: https://git.carlh.net/gitweb/?a=blobdiff_plain;f=RtAudio.cpp;h=e7c3ad12cce4542e6aa5663c7feaae279ba92cac;hb=50e9bba7559500f1ba01f3d8bc29ce15969d49ca;hp=7c87572fd79964819720eab69ccbce4cbaf1dd45;hpb=fa8b676a577de663b8622d0e8f6ce3605576b977;p=rtaudio.git

diff --git a/RtAudio.cpp b/RtAudio.cpp
index 7c87572..e7c3ad1 100644
--- a/RtAudio.cpp
+++ b/RtAudio.cpp
@@ -7,10 +7,11 @@
     and OSS), Macintosh OS X (CoreAudio and Jack), and Windows
     (DirectSound, ASIO and WASAPI) operating systems.
 
+    RtAudio GitHub site: https://github.com/thestk/rtaudio
     RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
 
     RtAudio: realtime audio i/o C++ classes
-    Copyright (c) 2001-2017 Gary P. Scavone
+    Copyright (c) 2001-2019 Gary P. Scavone
 
     Permission is hereby granted, free of charge, to any person
     obtaining a copy of this software and associated documentation files
@@ -38,7 +39,7 @@
 */
 /************************************************************************/
 
-// RtAudio: Version 5.0.0
+// RtAudio: Version 5.1.0
 
 #include "RtAudio.h"
 #include <iostream>
@@ -1541,6 +1542,10 @@ void RtApiCore :: startStream( void )
     return;
   }
 
+  #if defined( HAVE_GETTIMEOFDAY )
+  gettimeofday( &stream_.lastTickTimestamp, NULL );
+  #endif
+
   OSStatus result = noErr;
   CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
   if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
@@ -1901,7 +1906,10 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,
  unlock:
   //MUTEX_UNLOCK( &stream_.mutex );
 
-  RtApi::tickStreamTime();
+  // Make sure to only tick duplex stream time once if using two devices
+  if ( stream_.mode != DUPLEX || (stream_.mode == DUPLEX && handle->id[0] != handle->id[1] && deviceId == handle->id[0] ) )
+    RtApi::tickStreamTime();
+  
   return SUCCESS;
 }
 
@@ -2499,6 +2507,10 @@ void RtApiJack :: startStream( void )
     return;
   }
 
+  #if defined( HAVE_GETTIMEOFDAY )
+  gettimeofday( &stream_.lastTickTimestamp, NULL );
+  #endif
+
   JackHandle *handle = (JackHandle *) stream_.apiHandle;
   int result = jack_activate( handle->client );
   if ( result ) {
@@ -3378,6 +3390,10 @@ void RtApiAsio :: startStream()
     return;
   }
 
+  #if defined( HAVE_GETTIMEOFDAY )
+  gettimeofday( &stream_.lastTickTimestamp, NULL );
+  #endif
+
   AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
   ASIOError result = ASIOStart();
   if ( result != ASE_OK ) {
@@ -3829,8 +3845,9 @@ public:
       relOutIndex += bufferSize_;
     }
 
-    // "in" index can end on the "out" index but cannot begin at it
-    if ( inIndex_ <= relOutIndex && inIndexEnd > relOutIndex ) {
+    // the "IN" index CAN BEGIN at the "OUT" index
+    // the "IN" index CANNOT END at the "OUT" index
+    if ( inIndex_ < relOutIndex && inIndexEnd >= relOutIndex ) {
       return false; // not enough space between "in" index and "out" index
     }
 
@@ -3890,8 +3907,9 @@ public:
       relInIndex += bufferSize_;
     }
 
-    // "out" index can begin at and end on the "in" index
-    if ( outIndex_ < relInIndex && outIndexEnd > relInIndex ) {
+    // the "OUT" index CANNOT BEGIN at the "IN" index
+    // the "OUT" index CAN END at the "IN" index
+    if ( outIndex_ <= relInIndex && outIndexEnd > relInIndex ) {
       return false; // not enough space between "out" index and "in" index
     }
 
@@ -4551,6 +4569,10 @@ void RtApiWasapi::startStream( void )
     return;
   }
 
+  #if defined( HAVE_GETTIMEOFDAY )
+  gettimeofday( &stream_.lastTickTimestamp, NULL );
+  #endif
+
   // update stream state
   stream_.state = STREAM_RUNNING;
 
@@ -4590,26 +4612,6 @@ void RtApiWasapi::stopStream( void )
   // Wait for the last buffer to play before stopping.
   Sleep( 1000 * stream_.bufferSize / stream_.sampleRate );
 
-  // stop capture client if applicable
-  if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
-    HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
-    if ( FAILED( hr ) ) {
-      errorText_ = "RtApiWasapi::stopStream: Unable to stop capture stream.";
-      error( RtAudioError::DRIVER_ERROR );
-      return;
-    }
-  }
-
-  // stop render client if applicable
-  if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
-    HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
-    if ( FAILED( hr ) ) {
-      errorText_ = "RtApiWasapi::stopStream: Unable to stop render stream.";
-      error( RtAudioError::DRIVER_ERROR );
-      return;
-    }
-  }
-
   // close thread handle
   if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
     errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread.";
@@ -4640,26 +4642,6 @@ void RtApiWasapi::abortStream( void )
     Sleep( 1 );
   }
 
-  // stop capture client if applicable
-  if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
-    HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
-    if ( FAILED( hr ) ) {
-      errorText_ = "RtApiWasapi::abortStream: Unable to stop capture stream.";
-      error( RtAudioError::DRIVER_ERROR );
-      return;
-    }
-  }
-
-  // stop render client if applicable
-  if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
-    HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
-    if ( FAILED( hr ) ) {
-      errorText_ = "RtApiWasapi::abortStream: Unable to stop render stream.";
-      error( RtAudioError::DRIVER_ERROR );
-      return;
-    }
-  }
-
   // close thread handle
   if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
     errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread.";
@@ -4860,15 +4842,14 @@ bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
   stream_.doConvertBuffer[mode] = false;
   if ( stream_.userFormat != stream_.deviceFormat[mode] ||
        stream_.nUserChannels[0] != stream_.nDeviceChannels[0] ||
-       stream_.nUserChannels[1] != stream_.nDeviceChannels[1] ||
-       stream_.userInterleaved )
+       stream_.nUserChannels[1] != stream_.nDeviceChannels[1] )
     stream_.doConvertBuffer[mode] = true;
   else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
             stream_.nUserChannels[mode] > 1 )
     stream_.doConvertBuffer[mode] = true;
 
   if ( stream_.doConvertBuffer[mode] )
-    setConvertInfo( mode, 0 );
+    setConvertInfo( mode, firstChannel );
 
   // Allocate necessary internal buffers
   bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat );
@@ -4982,7 +4963,8 @@ void RtApiWasapi::wasapiThread()
   HMODULE AvrtDll = LoadLibrary( (LPCTSTR) "AVRT.dll" );
   if ( AvrtDll ) {
     DWORD taskIndex = 0;
-    TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" );
+    TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr =
+      ( TAvSetMmThreadCharacteristicsPtr ) (void(*)()) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" );
     AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex );
     FreeLibrary( AvrtDll );
   }
@@ -5042,6 +5024,20 @@ void RtApiWasapi::wasapiThread()
       }
 
       ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient;
+
+      // reset the capture stream
+      hr = captureAudioClient->Reset();
+      if ( FAILED( hr ) ) {
+        errorText = "RtApiWasapi::wasapiThread: Unable to reset capture stream.";
+        goto Exit;
+      }
+
+      // start the capture stream
+      hr = captureAudioClient->Start();
+      if ( FAILED( hr ) ) {
+        errorText = "RtApiWasapi::wasapiThread: Unable to start capture stream.";
+        goto Exit;
+      }
     }
 
     unsigned int inBufferSize = 0;
@@ -5057,20 +5053,6 @@ void RtApiWasapi::wasapiThread()
 
     // set captureBuffer size
     captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) );
-
-    // reset the capture stream
-    hr = captureAudioClient->Reset();
-    if ( FAILED( hr ) ) {
-      errorText = "RtApiWasapi::wasapiThread: Unable to reset capture stream.";
-      goto Exit;
-    }
-
-    // start the capture stream
-    hr = captureAudioClient->Start();
-    if ( FAILED( hr ) ) {
-      errorText = "RtApiWasapi::wasapiThread: Unable to start capture stream.";
-      goto Exit;
-    }
   }
 
   // start render stream if applicable
@@ -5123,6 +5105,20 @@ void RtApiWasapi::wasapiThread()
 
       ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient;
       ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent;
+
+      // reset the render stream
+      hr = renderAudioClient->Reset();
+      if ( FAILED( hr ) ) {
+        errorText = "RtApiWasapi::wasapiThread: Unable to reset render stream.";
+        goto Exit;
+      }
+
+      // start the render stream
+      hr = renderAudioClient->Start();
+      if ( FAILED( hr ) ) {
+        errorText = "RtApiWasapi::wasapiThread: Unable to start render stream.";
+        goto Exit;
+      }
     }
 
     unsigned int outBufferSize = 0;
@@ -5138,20 +5134,6 @@ void RtApiWasapi::wasapiThread()
 
     // set renderBuffer size
     renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) );
-
-    // reset the render stream
-    hr = renderAudioClient->Reset();
-    if ( FAILED( hr ) ) {
-      errorText = "RtApiWasapi::wasapiThread: Unable to reset render stream.";
-      goto Exit;
-    }
-
-    // start the render stream
-    hr = renderAudioClient->Start();
-    if ( FAILED( hr ) ) {
-      errorText = "RtApiWasapi::wasapiThread: Unable to start render stream.";
-      goto Exit;
-    }
   }
 
   // malloc buffer memory
@@ -5175,8 +5157,8 @@ void RtApiWasapi::wasapiThread()
   }
 
   convBuffSize *= 2; // allow overflow for *SrRatio remainders
-  convBuffer = ( char* ) malloc( convBuffSize );
-  stream_.deviceBuffer = ( char* ) malloc( deviceBuffSize );
+  convBuffer = ( char* ) calloc( convBuffSize, 1 );
+  stream_.deviceBuffer = ( char* ) calloc( deviceBuffSize, 1 );
   if ( !convBuffer || !stream_.deviceBuffer ) {
     errorType = RtAudioError::MEMORY_ERROR;
     errorText = "RtApiWasapi::wasapiThread: Error allocating device buffer memory.";
@@ -5263,6 +5245,9 @@ void RtApiWasapi::wasapiThread()
                                    captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0,
                                    stream_.callbackInfo.userData );
 
+        // tick stream time
+        RtApi::tickStreamTime();
+
         // Handle return value from callback
         if ( callbackResult == 1 ) {
           // instantiate a thread to stop this thread
@@ -5318,6 +5303,12 @@ void RtApiWasapi::wasapiThread()
                          stream_.convertInfo[OUTPUT] );
 
         }
+        else {
+          // no further conversion, simple copy userBuffer to deviceBuffer
+          memcpy( stream_.deviceBuffer,
+                  stream_.userBuffer[OUTPUT],
+                  stream_.bufferSize * stream_.nUserChannels[OUTPUT] * formatBytes( stream_.userFormat ) );
+        }
 
         // Convert callback buffer to stream sample rate
         renderResampler->Convert( convBuffer,
@@ -5465,9 +5456,6 @@ void RtApiWasapi::wasapiThread()
       // unsetting the callbackPulled flag lets the stream know that
       // the audio device is ready for another callback output buffer.
       callbackPulled = false;
-
-      // tick stream time
-      RtApi::tickStreamTime();
     }
 
   }
@@ -6387,6 +6375,10 @@ void RtApiDs :: startStream()
     return;
   }
 
+  #if defined( HAVE_GETTIMEOFDAY )
+  gettimeofday( &stream_.lastTickTimestamp, NULL );
+  #endif
+
   DsHandle *handle = (DsHandle *) stream_.apiHandle;
 
   // Increase scheduler frequency on lesser windows (a side-effect of
@@ -7154,7 +7146,7 @@ unsigned int RtApiAlsa :: getDeviceCount( void )
   unsigned nDevices = 0;
   int result, subdevice, card;
   char name[64];
-  snd_ctl_t *handle;
+  snd_ctl_t *handle = 0;
 
   // Count cards and devices
   card = -1;
@@ -7163,6 +7155,7 @@ unsigned int RtApiAlsa :: getDeviceCount( void )
     sprintf( name, "hw:%d", card );
     result = snd_ctl_open( &handle, name, 0 );
     if ( result < 0 ) {
+      handle = 0;
       errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";
       errorText_ = errorStream_.str();
       error( RtAudioError::WARNING );
@@ -7182,7 +7175,8 @@ unsigned int RtApiAlsa :: getDeviceCount( void )
       nDevices++;
     }
   nextcard:
-    snd_ctl_close( handle );
+    if ( handle )
+        snd_ctl_close( handle );
     snd_card_next( &card );
   }
 
@@ -7203,7 +7197,7 @@ RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
   unsigned nDevices = 0;
   int result, subdevice, card;
   char name[64];
-  snd_ctl_t *chandle;
+  snd_ctl_t *chandle = 0;
 
   // Count cards and devices
   card = -1;
@@ -7213,6 +7207,7 @@ RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
     sprintf( name, "hw:%d", card );
     result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
     if ( result < 0 ) {
+      chandle = 0;
       errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";
       errorText_ = errorStream_.str();
       error( RtAudioError::WARNING );
@@ -7235,7 +7230,8 @@ RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
       nDevices++;
     }
   nextcard:
-    snd_ctl_close( chandle );
+    if ( chandle )
+        snd_ctl_close( chandle );
     snd_card_next( &card );
   }
 
@@ -8079,6 +8075,10 @@ void RtApiAlsa :: startStream()
 
   MUTEX_LOCK( &stream_.mutex );
 
+  #if defined( HAVE_GETTIMEOFDAY )
+  gettimeofday( &stream_.lastTickTimestamp, NULL );
+  #endif
+
   int result = 0;
   snd_pcm_state_t state;
   AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
@@ -8650,6 +8650,10 @@ void RtApiPulse::startStream( void )
 
   MUTEX_LOCK( &stream_.mutex );
 
+  #if defined( HAVE_GETTIMEOFDAY )
+  gettimeofday( &stream_.lastTickTimestamp, NULL );
+  #endif
+
   stream_.state = STREAM_RUNNING;
 
   pah->runnable = true;
@@ -9634,6 +9638,10 @@ void RtApiOss :: startStream()
 
   MUTEX_LOCK( &stream_.mutex );
 
+  #if defined( HAVE_GETTIMEOFDAY )
+  gettimeofday( &stream_.lastTickTimestamp, NULL );
+  #endif
+
   stream_.state = STREAM_RUNNING;
 
   // No need to do anything else here ... OSS automatically starts