- Update to verion 1.1 [1]

- Update project homepage [1]
- Add check for options sanity (at least one codec must be choosen)
- Install darkice.cfg and remove it when it was not edited

PR:		ports/162290 [1]
Submitted by:	Takefu <takefu@airport.fm> [1]

8 files changed, 20 insertions, 551 deletions

diff --git a/audio/darkice/Makefile b/audio/darkice/Makefile
index daf890c43814..2a1ea436caf6 100644
--- a/audio/darkice/Makefile
+++ b/audio/darkice/Makefile
@@ -7,8 +7,7 @@
 #
 
 PORTNAME=	darkice
-PORTVERSION=	1.0
-PORTREVISION=	3
+PORTVERSION=	1.1
 CATEGORIES=	audio net
 MASTER_SITES=	GOOGLE_CODE
 
@@ -36,9 +35,8 @@ USE_RC_SUBR=	${PORTNAME}
 
 MAN1=		darkice.1
 MAN5=		darkice.cfg.5
-PLIST_FILES=	bin/darkice etc/darkice.cfg.dist
 
-.include <bsd.port.pre.mk>
+.include <bsd.port.options.mk>
 
 .if defined(WITHOUT_VORBIS)
 CONFIGURE_ARGS+=	--without-vorbis
@@ -83,6 +81,11 @@ CONFIGURE_ARGS+=	--with-aacplus-prefix=${LOCALBASE} --with-samplerate-prefix=${L
 CONFIGURE_ARGS+=	--without-aacplus --without-samplerate
 .endif
 
+.if defined(WITHOUT_VORBIS) && defined(WITHOUT_LAME) && defined(WITHOUT_TWOLAME) && \
+	defined(WITHOUT_FAAC) && defined(WITHOUT_AACPLUS)
+IGNORE=at least one of VORBIS,LAME,TWOLAME,FAAC,AACPLUS options must be set
+.endif
+
 post-patch:
 	@${REINPLACE_CMD} -e '/test/s|==|=|g'\
 	 -e 's/sbr_main.h/libaacplus\/sbr_main.h/' ${WRKSRC}/configure
@@ -95,4 +98,9 @@ do-install:
 	@${INSTALL_MAN} ${WRKSRC}/man/${PORTNAME}.cfg.5 ${MAN5PREFIX}/man/man5
 	@${CAT} ${PKGMESSAGE}
 
-.include <bsd.port.post.mk>
+post-install:
+	@if [ ! -f ${PREFIX}/etc/darkice.cfg ]; then \
+		${CP} -p ${PREFIX}/etc/darkice.cfg.dist ${PREFIX}/etc/darkice.cfg ; \
+	fi
+
+.include <bsd.port.mk>
diff --git a/audio/darkice/distinfo b/audio/darkice/distinfo
index 6b18851d1d25..ed974f3567eb 100644
--- a/audio/darkice/distinfo
+++ b/audio/darkice/distinfo
@@ -1,2 +1,2 @@
-SHA256 (darkice-1.0.tar.gz) = 61a05c4dab206c22c3e3d5570ee4841f9c8875241098adf687717e7dcc6df332
-SIZE (darkice-1.0.tar.gz) = 311567
+SHA256 (darkice-1.1.tar.gz) = 170342cb4dbb0b44a62e37d0db1515fa7799c410fc4995bf8f32aaa6614f5f79
+SIZE (darkice-1.1.tar.gz) = 344568
diff --git a/audio/darkice/files/patch-configure.in b/audio/darkice/files/patch-configure.in
deleted file mode 100644
index 2e48f4b16bbe..000000000000
--- a/audio/darkice/files/patch-configure.in
+++ /dev/null
@@ -1,11 +0,0 @@
---- configure.in.orig	2010-05-10 06:38:57.000000000 +0900
-+++ configure.in	2010-12-29 19:11:40.000000000 +0900
-@@ -166,7 +166,7 @@
-
- if test "x${USE_AACPLUS}" = "xyes" ; then
-     AC_MSG_CHECKING( [for aacplus library at ${CONFIG_AACPLUS_PREFIX}] )
--    LA_SEARCH_LIB( AACPLUS_LIB_LOC, AACPLUS_INC_LOC, libaacplus.a libaacplus.so, sbr_main.h,
-+    LA_SEARCH_LIB( AACPLUS_LIB_LOC, AACPLUS_INC_LOC, libaacplus.a libaacplus.so, aacplus.h,
-                    ${CONFIG_AACPLUS_PREFIX})
-     if test "x${AACPLUS_LIB_LOC}" != "x" ; then
-         AC_DEFINE( HAVE_AACPLUS_LIB, 1, [build with aacplus library] )
diff --git a/audio/darkice/files/patch-darkice.cfg b/audio/darkice/files/patch-darkice.cfg
deleted file mode 100644
index 59354fe2410d..000000000000
--- a/audio/darkice/files/patch-darkice.cfg
+++ /dev/null
@@ -1,10 +0,0 @@
---- darkice.cfg.orig	2010-05-10 05:26:19.000000000 +0900
-+++ darkice.cfg	2010-12-29 19:17:57.000000000 +0900
-@@ -6,6 +6,7 @@
- duration        = 60        # duration of encoding, in seconds. 0 means forever
- bufferSecs      = 5         # size of internal slip buffer, in seconds
- reconnect       = yes       # reconnect to the server(s) if disconnected
-+realtime        = yes       # run the encoder with POSIX realtime priority
-
- # this section describes the audio input that will be streamed
- [input]
diff --git a/audio/darkice/files/patch-src_aacPlusEncoder.cpp b/audio/darkice/files/patch-src_aacPlusEncoder.cpp
deleted file mode 100644
index 2093b16877f7..000000000000
--- a/audio/darkice/files/patch-src_aacPlusEncoder.cpp
+++ /dev/null
@@ -1,328 +0,0 @@
---- src/aacPlusEncoder.cpp.orig	2010-05-10 00:18:48.000000000 +0200
-+++ src/aacPlusEncoder.cpp	2011-01-20 13:39:21.000000000 +0100
-@@ -5,8 +5,8 @@
-    Tyrell DarkIce
- 
-    File     : aacPlusEncoder.cpp
--   Version  : $Revision: 474 $
--   Author   : $Author: rafael@riseup.net $
-+   Version  : $Revision$
-+   Author   : $Author$
-    Location : $HeadURL$
- 
-    Copyright notice:
-@@ -51,7 +51,7 @@
- /*------------------------------------------------------------------------------
-  *  File identity
-  *----------------------------------------------------------------------------*/
--static const char fileid[] = "$Id: aacPlusEncoder.cpp 474 2010-05-10 01:18:15Z rafael@riseup.net $";
-+static const char fileid[] = "$Id$";
- 
- 
- /* ===============================================  local function prototypes */
-@@ -76,82 +76,27 @@
-                          "aacplus lib opening underlying sink error");
-     }
- 
--    reportEvent(1, "Using aacplus codec version", "720 3gpp");
-+    reportEvent(1, "Using aacplus codec");
-     
--    bitrate = getOutBitrate() * 1000;
--    bandwidth = 0;
--    useParametricStereo = 0;
--    numAncDataBytes=0;
--    coreWriteOffset = 0;
--    envReadOffset = 0;
--    writeOffset = INPUT_DELAY*MAX_CHANNELS;
--    writtenSamples = 0;
--    aacEnc = NULL;
--    hEnvEnc=NULL;
--
--    /* set up basic parameters for aacPlus codec */
--    AacInitDefaultConfig(&config);
--    nChannelsAAC = nChannelsSBR = getOutChannel();
--    
--    if ( (getInChannel() == 2) && (bitrate >= 16000) && (bitrate < 44001) ) {
--        useParametricStereo = 1;
--        nChannelsAAC = 1;
--        nChannelsSBR = 2;
--        
--        reportEvent(10, "use Parametric Stereo");
--        
--        envReadOffset = (MAX_DS_FILTER_DELAY + INPUT_DELAY)*MAX_CHANNELS;
--        coreWriteOffset = CORE_INPUT_OFFSET_PS;
--        writeOffset = envReadOffset;
--    } else {
--    	/* set up 2:1 downsampling */
--    	InitIIR21_Resampler(&(IIR21_reSampler[0]));
--    	InitIIR21_Resampler(&(IIR21_reSampler[1]));
--    	
--    	if(IIR21_reSampler[0].delay > MAX_DS_FILTER_DELAY)
--    		throw Exception(__FILE__, __LINE__, "IIR21 resampler delay is bigger then MAX_DS_FILTER_DELAY");
--        writeOffset += IIR21_reSampler[0].delay*MAX_CHANNELS;
-+    encoderHandle = aacplusEncOpen(getOutSampleRate(),
-+                                getInChannel(),
-+                                &inputSamples,
-+                                &maxOutputBytes);
-+
-+    aacplusEncConfiguration      * aacplusConfig;
-+
-+    aacplusConfig = aacplusEncGetCurrentConfiguration(encoderHandle);
-+
-+    aacplusConfig->bitRate       = getOutBitrate() * 1000;
-+    aacplusConfig->bandWidth     = lowpass;
-+    aacplusConfig->outputFormat  = 1;
-+    aacplusConfig->inputFormat   = AACPLUS_INPUT_16BIT;
-+    aacplusConfig->nChannelsOut  = getOutChannel();
-+
-+    if (!aacplusEncSetConfiguration(encoderHandle, aacplusConfig)) {
-+        throw Exception(__FILE__, __LINE__,
-+                        "error configuring libaacplus library");
-     }
--    
--    sampleRateAAC = getOutSampleRate();
--    config.bitRate = bitrate;
--    config.nChannelsIn=getInChannel();
--    config.nChannelsOut=nChannelsAAC;
--    config.bandWidth=bandwidth;
--    
--    /* set up SBR configuration    */
--    if(!IsSbrSettingAvail(bitrate, nChannelsAAC, sampleRateAAC, &sampleRateAAC))
--        throw Exception(__FILE__, __LINE__, "No valid SBR configuration found");
--    
--    InitializeSbrDefaults (&sbrConfig);
--    sbrConfig.usePs = useParametricStereo;
--    
--    AdjustSbrSettings( &sbrConfig,
--                       bitrate,
--                       nChannelsAAC,
--                       sampleRateAAC,
--                       AACENC_TRANS_FAC,
--                       24000);
--    
--    EnvOpen( &hEnvEnc,
--             inBuf + coreWriteOffset,
--             &sbrConfig,
--             &config.bandWidth);
--        
--    /* set up AAC encoder, now that samling rate is known */
--    config.sampleRate = sampleRateAAC;
--    if (AacEncOpen(&aacEnc, config) != 0){
--        AacEncClose(aacEnc);
--        throw Exception(__FILE__, __LINE__, "Initialisation of AAC failed !");
--    }
--    
--    init_plans();
--    
--    /* create the ADTS header */
--    adts_hdr(outBuf, &config);
--    
--    inSamples = AACENC_BLOCKSIZE * getInChannel() * 2;
--    
- 
-     // initialize the resampling coverter if needed
-     if ( converter ) {
-@@ -159,8 +104,8 @@
-         converterData.input_frames   = 4096/((getInBitsPerSample() / 8) * getInChannel());
-         converterData.data_in        = new float[converterData.input_frames*getInChannel()];
-         converterData.output_frames  = (int) (converterData.input_frames * resampleRatio + 1);
--        if ((int) inSamples >  getInChannel() * converterData.output_frames) {
--            resampledOffset       = new float[2 * inSamples];
-+        if ((int) inputSamples >  getInChannel() * converterData.output_frames) {
-+            resampledOffset       = new float[2 * inputSamples];
-         } else {
-             resampledOffset       = new float[2 * getInChannel() * converterData.input_frames];
-         }
-@@ -178,13 +123,9 @@
-     }
- 
-     aacplusOpen = true;
--    reportEvent(10, "bitrate=", bitrate);
--    reportEvent(10, "nChannelsIn", getInChannel());
--    reportEvent(10, "nChannelsOut", getOutChannel());
--    reportEvent(10, "nChannelsSBR", nChannelsSBR);
--    reportEvent(10, "nChannelsAAC", nChannelsAAC);
--    reportEvent(10, "sampleRateAAC", sampleRateAAC);
--    reportEvent(10, "inSamples", inSamples);
-+    reportEvent(10, "nChannelsAAC", aacplusConfig->nChannelsOut);
-+    reportEvent(10, "sampleRateAAC", aacplusConfig->sampleRate);
-+    reportEvent(10, "inSamples", inputSamples);
-     return true;
- }
- 
-@@ -199,21 +140,23 @@
-     if ( !isOpen() || len == 0) {
-         return 0;
-     }
--    
-+
-     unsigned int    channels         = getInChannel();
-     unsigned int    bitsPerSample    = getInBitsPerSample();
-     unsigned int    sampleSize       = (bitsPerSample / 8) * channels;
-+    unsigned char * b                = (unsigned char*) buf;
-     unsigned int    processed        = len - (len % sampleSize);
-     unsigned int    nSamples         = processed / sampleSize;
--    unsigned int    samples          = (unsigned int) nSamples * channels;
--    int processedSamples = 0;
--    
--    
-+    unsigned char * aacplusBuf          = new unsigned char[maxOutputBytes];
-+    int             samples          = (int) nSamples * channels;
-+    int             processedSamples = 0;
-+
-+
- 
-     if ( converter ) {
-         unsigned int         converted;
- #ifdef HAVE_SRC_LIB
--        src_short_to_float_array ((short *) buf, converterData.data_in, samples);
-+        src_short_to_float_array ((short *) b, converterData.data_in, samples);
-         converterData.input_frames   = nSamples;
-         converterData.data_out = resampledOffset + (resampledOffsetSize * channels);
-         int srcError = src_process (converter, &converterData);
-@@ -224,7 +167,6 @@
-         int         inCount  = nSamples;
-         short int     * shortBuffer  = new short int[samples];
-         int         outCount = (int) (inCount * resampleRatio);
--        unsigned char * b = (unsigned char*) buf;
-         Util::conv( bitsPerSample, b, processed, shortBuffer, isInBigEndian());
-         converted = converter->resample( inCount,
-                                          outCount+1,
-@@ -235,18 +177,27 @@
-         resampledOffsetSize += converted;
- 
-         // encode samples (if enough)
--        while(resampledOffsetSize - processedSamples >= inSamples/channels) {
-+        while(resampledOffsetSize - processedSamples >= inputSamples/channels) {
-+            int outputBytes;
- #ifdef HAVE_SRC_LIB
--            short *shortData = new short[inSamples];
-+            short *shortData = new short[inputSamples];
-             src_float_to_short_array(resampledOffset + (processedSamples * channels),
--                                     shortData, inSamples) ;
--
--            encodeAacSamples (shortData, inSamples, channels);
-+                                     shortData, inputSamples) ;
-+            outputBytes = aacplusEncEncode(encoderHandle,
-+                                       (int32_t*) shortData,
-+                                        inputSamples,
-+                                        aacplusBuf,
-+                                        maxOutputBytes);
-             delete [] shortData;
- #else
--            encodeAacSamples (&resampledOffset[processedSamples*channels], inSamples, channels);
-+            outputBytes = aacplusEncEncode(encoderHandle,
-+                                       (int32_t*) &resampledOffset[processedSamples*channels],
-+                                        inputSamples,
-+                                        aacplusBuf,
-+                                        maxOutputBytes);
- #endif
--            processedSamples+=inSamples/channels;
-+            getSink()->write(aacplusBuf, outputBytes);
-+            processedSamples+=inputSamples/channels;
-         }
- 
-         if (processedSamples && (int) resampledOffsetSize >= processedSamples) {
-@@ -262,70 +213,27 @@
- #endif
-         }
-     } else {
--        encodeAacSamples ((short *) buf, samples, channels);
--    }
-+        while (processedSamples < samples) {
-+            int     outputBytes;
-+            int     inSamples = samples - processedSamples < (int) inputSamples
-+                              ? samples - processedSamples
-+                              : inputSamples;
-+
-+            outputBytes = aacplusEncEncode(encoderHandle,
-+                                       (int32_t*) (b + processedSamples/sampleSize),
-+                                        inSamples,
-+                                        aacplusBuf,
-+                                        maxOutputBytes);
-+            getSink()->write(aacplusBuf, outputBytes);
- 
--    return samples;
--}
--
--void
--aacPlusEncoder :: encodeAacSamples (short *TimeDataPcm, unsigned int samples, int channels)
--                                                                               throw ( Exception )
--{
--    unsigned int i;
--    int ch, outSamples, numOutBytes;
--
--    for (i=0; i<samples; i++)
--        inBuf[(2/channels)*i+writeOffset+writtenSamples] = (float) TimeDataPcm[i];
--
--    writtenSamples+=samples;
--
--    if (writtenSamples < inSamples)
--        return;
--    
--    /* encode one SBR frame */
--    EnvEncodeFrame( hEnvEnc,
--                    inBuf + envReadOffset,
--                    inBuf + coreWriteOffset,
--                    MAX_CHANNELS,
--                    &numAncDataBytes,
--                    ancDataBytes);
--    
--    /* 2:1 downsampling for AAC core */
--    if (!useParametricStereo) {
--        for( ch=0; ch<nChannelsAAC; ch++ )
--            IIR21_Downsample( &(IIR21_reSampler[ch]),
--                              inBuf + writeOffset+ch,
--                              writtenSamples/channels,
--                              MAX_CHANNELS,
--                              inBuf+ch,
--                              &outSamples,
--                              MAX_CHANNELS);
--    }
--        
--    /* encode one AAC frame */
--    AacEncEncode( aacEnc,
--                  inBuf,
--                  useParametricStereo ? 1 : MAX_CHANNELS, /* stride (step) */
--                  ancDataBytes,
--                  &numAncDataBytes,
--                  (unsigned *) (outBuf+ADTS_HEADER_SIZE),
--                  &numOutBytes);
--    if (useParametricStereo) {
--        memcpy( inBuf,inBuf+AACENC_BLOCKSIZE,CORE_INPUT_OFFSET_PS*sizeof(float));
--    } else {
--        memmove( inBuf,inBuf+AACENC_BLOCKSIZE*2*MAX_CHANNELS,writeOffset*sizeof(float));
--    }
--    
--    /* Write one frame of encoded audio */
--    if (numOutBytes) {
--        adts_hdr_up(outBuf, numOutBytes);
--        sink->write(outBuf, numOutBytes+ADTS_HEADER_SIZE);
-+            processedSamples += inSamples;
-+        }
-     }
--    
--    writtenSamples=0;
- 
--    return;
-+    delete[] aacplusBuf;
-+
-+//    return processedSamples;
-+    return samples;
- }
- 
- /*------------------------------------------------------------------------------
-@@ -352,12 +260,7 @@
-     if ( isOpen() ) {
-         flush();
-     
--        destroy_plans();
--        AacEncClose(aacEnc);
--        if (hEnvEnc) {
--            EnvClose(hEnvEnc);
--        }
--    
-+        aacplusEncClose(encoderHandle);
-         aacplusOpen = false;
-     
-         sink->close();
diff --git a/audio/darkice/files/patch-src_aacPlusEncoder.h b/audio/darkice/files/patch-src_aacPlusEncoder.h
deleted file mode 100644
index b77150ce5a15..000000000000
--- a/audio/darkice/files/patch-src_aacPlusEncoder.h
+++ /dev/null
@@ -1,194 +0,0 @@
---- src/aacPlusEncoder.h.orig	2010-05-10 00:18:48.000000000 +0200
-+++ src/aacPlusEncoder.h	2011-01-20 13:41:06.000000000 +0100
-@@ -5,8 +5,8 @@
-    Tyrell DarkIce
- 
-    File     : aacPlusEncoder.h
--   Version  : $Revision: 474 $
--   Author   : $Author: rafael@riseup.net $
-+   Version  : $Revision$
-+   Author   : $Author$
-    Location : $HeadURL$
-    
-    Copyright notice:
-@@ -41,18 +41,7 @@
- #endif
- 
- #ifdef HAVE_AACPLUS_LIB
--extern "C" {
--#include <libaacplus/cfftn.h>
--#include <libaacplus/FloatFR.h>
--#include <libaacplus/aacenc.h>
--#include <libaacplus/resampler.h>
--
--#include <libaacplus/adts.h>
--
--#include <libaacplus/sbr_main.h>
--#include <libaacplus/aac_ram.h>
--#include <libaacplus/aac_rom.h>
--}
-+#include <aacplus.h>
- #else
- #error configure with aacplus
- #endif
-@@ -83,16 +72,10 @@
- /**
-  *  A class representing aacplus AAC+ encoder.
-  *
-- *  @author  $Author: rafael@riseup.net $
-- *  @version $Revision: 474 $
-+ *  @author  $Author$
-+ *  @version $Revision$
-  */
- 
--#define CORE_DELAY   (1600)
--#define INPUT_DELAY  ((CORE_DELAY)*2 +6*64-2048+1)     /* ((1600 (core codec)*2 (multi rate) + 6*64 (sbr dec delay) - 2048 (sbr enc delay) + magic*/
--#define MAX_DS_FILTER_DELAY 16                         /* the additional max resampler filter delay (source fs)*/
--
--#define CORE_INPUT_OFFSET_PS (0)  /* (96-64) makes AAC still some 64 core samples too early wrt SBR ... maybe -32 would be even more correct, but 1024-32 would need additional SBR bitstream delay by one frame */
--
- class aacPlusEncoder : public AudioEncoder, public virtual Reporter
- {
-     private:
-@@ -124,31 +107,26 @@
-          */
-         Ref<Sink>                   sink;
- 
--		float inBuf[(AACENC_BLOCKSIZE*2 + MAX_DS_FILTER_DELAY + INPUT_DELAY)*MAX_CHANNELS];
--		char outBuf[(6144/8)*MAX_CHANNELS+ADTS_HEADER_SIZE];
--		IIR21_RESAMPLER IIR21_reSampler[MAX_CHANNELS];
--
--		AACENC_CONFIG     config;
--		
--		int nChannelsAAC, nChannelsSBR;
--		unsigned int sampleRateAAC;
--
--		int bitrate;
--		int bandwidth;
--		
--		unsigned int numAncDataBytes;
--		unsigned char ancDataBytes[MAX_PAYLOAD_SIZE];
--		
--		bool useParametricStereo;
--		int coreWriteOffset;
--		int envReadOffset;
--		int writeOffset;
--		struct AAC_ENCODER *aacEnc;
--		unsigned int inSamples;
--		unsigned int writtenSamples;
--		
--		HANDLE_SBR_ENCODER hEnvEnc;
--		sbrConfiguration sbrConfig;
-+        /**
-+         *  The handle to the AAC+ encoder instance.
-+         */
-+        aacplusEncHandle               encoderHandle;
-+
-+        /**
-+         *  The maximum number of input samples to supply to the encoder.
-+         */
-+        unsigned long               inputSamples;
-+
-+        /**
-+         *  The maximum number of output bytes the encoder returns in one call.
-+         */
-+        unsigned long               maxOutputBytes;
-+
-+        /**
-+         *  Lowpass filter. Sound frequency in Hz, from where up the
-+         *  input is cut.
-+         */
-+        int                             lowpass;
- 
-         /**
-          *  Initialize the object.
-@@ -157,10 +135,11 @@
-          *  @exception Exception
-          */
-         inline void
--        init ( Sink           * sink)                throw (Exception)
-+        init ( Sink           * sink, int lowpass)                throw (Exception)
-         {
-             this->aacplusOpen        = false;
-             this->sink            = sink;
-+            this->lowpass         = lowpass;
- 	    
- 	    /* TODO: if we have float as input, we don't need conversion */
-             if ( getInBitsPerSample() != 16 && getInBitsPerSample() != 32 ) {
-@@ -179,11 +158,6 @@
-                         "unsupported number of output channels for the encoder",
-                                  getOutChannel() );
-             }
--	    /* TODO: this will be neede when we implement mono aac+ encoding */
--            if ( getInChannel() != getOutChannel() ) {
--                throw Exception( __FILE__, __LINE__,
--                             "input channels and output channels do not match");
--            }
- 
-             if ( getOutSampleRate() == getInSampleRate() ) {
-                 resampleRatio = 1;
-@@ -237,17 +211,6 @@
-                                  "specified bits per sample with samplerate conversion not supported",
-                                  getInBitsPerSample() );
-             }
--
--            bitrate = getOutBitrate() * 1000;
--            bandwidth = 0;
--            useParametricStereo = 0;
--            numAncDataBytes=0;
--            coreWriteOffset = 0;
--            envReadOffset = 0;
--            writeOffset = INPUT_DELAY*MAX_CHANNELS;
--            writtenSamples = 0;
--            aacEnc = NULL;
--            hEnvEnc=NULL;
-         }
- 
-         /**
-@@ -269,10 +232,6 @@
-             }
-         }
- 
--        void
--        encodeAacSamples (short *TimeDataPcm, unsigned int samples, int channels)
--                                                        throw ( Exception );
--
-     protected:
- 
-         /**
-@@ -335,7 +294,7 @@
-                                      outSampleRate,
-                                      outChannel )
-         {
--            init( sink);
-+            init( sink, lowpass);
-         }
- 
-         /**
-@@ -376,7 +335,7 @@
-                                      outSampleRate,
-                                      outChannel )
-         {
--            init( sink);
-+            init( sink, lowpass );
-         }
- 
-         /**
-@@ -389,7 +348,7 @@
-                                                             throw ( Exception )
-                     : AudioEncoder( encoder )
-         {
--            init( encoder.sink.get());
-+            init( encoder.sink.get(), encoder.lowpass);
-         }
- 
- 
-@@ -420,7 +379,7 @@
-             if ( this != &encoder ) {
-                 strip();
-                 AudioEncoder::operator=( encoder);
--                init( encoder.sink.get());
-+                init( encoder.sink.get(), encoder.lowpass);
-             }
- 
-             return *this;
diff --git a/audio/darkice/pkg-descr b/audio/darkice/pkg-descr
index 976cb4aa812b..c91e1319d1e2 100644
--- a/audio/darkice/pkg-descr
+++ b/audio/darkice/pkg-descr
@@ -17,4 +17,4 @@ DarkIce can send the encoded stream to the following streaming servers:
  Darwin Streaming Server
  archive the encoded audio in files
 
-WWW: http://code.google.com/p/darkice/
+WWW: http://darkice.org/
diff --git a/audio/darkice/pkg-plist b/audio/darkice/pkg-plist
new file mode 100644
index 000000000000..26096d569fae
--- /dev/null
+++ b/audio/darkice/pkg-plist
@@ -0,0 +1,4 @@
+bin/darkice
+@unexec if cmp -s %D/etc/darkice.cfg.dist %D/etc/darkice.cfg; then rm -f %D/etc/darkice.cfg; fi
+etc/darkice.cfg.dist
+@exec if [ ! -f %D/etc/darkice.cfg ] ; then cp -p %D/%F %B/darkice.cfg; fi