LICENSE=	GPLv3

LICENSE_FILE=	${WRKSRC}/COPYING

USE_AUTOTOOLS=	aclocal autoconf

--- 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] )

--- 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]

--- src/aacPlusEncoder.cpp.orig	2010-05-10 07:18:48.000000000 +0900

+++ src/aacPlusEncoder.cpp	2010-10-13 07:42:25.000000000 +0900

@@ -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();

--- src/aacPlusEncoder.h.orig	2010-05-10 07:18:48.000000000 +0900

+++ src/aacPlusEncoder.h	2010-10-13 07:42:25.000000000 +0900

@@ -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;