update to 0.2.7

changes:
-Fix decoding of multi-channel ADPCM WAVE files (was patched in pkgsrc)
-Reduce unshared data in library
-Fix handling of audio files with more than 2^24 frames
-Add support for writing double-precision floating-point WAVE files
-Add support for reading certain uncompressed AIFF-C files created
 by Mac OS X
-Write fact chunk in floating-point WAVE files

7 files changed, 56 insertions, 159 deletions

diff --git a/audio/libaudiofile/Makefile b/audio/libaudiofile/Makefile
index e8cf56539d6..fdb67902522 100644
--- a/audio/libaudiofile/Makefile
+++ b/audio/libaudiofile/Makefile
@@ -1,8 +1,7 @@
-# $NetBSD: Makefile,v 1.44 2009/12/01 10:44:02 drochner Exp $
+# $NetBSD: Makefile,v 1.45 2010/04/12 19:04:58 drochner Exp $
 
-DISTNAME=		audiofile-0.2.6
+DISTNAME=		audiofile-0.2.7
 PKGNAME=		lib${DISTNAME}
-PKGREVISION=		3
 CATEGORIES=		audio
 MASTER_SITES=		http://www.68k.org/~michael/audiofile/ \
 			${MASTER_SITE_GNOME:=sources/audiofile/0.2/}
diff --git a/audio/libaudiofile/distinfo b/audio/libaudiofile/distinfo
index 4fe8b47ee83..6e8f9177729 100644
--- a/audio/libaudiofile/distinfo
+++ b/audio/libaudiofile/distinfo
@@ -1,9 +1,8 @@
-$NetBSD: distinfo,v 1.13 2009/12/01 10:44:02 drochner Exp $
+$NetBSD: distinfo,v 1.14 2010/04/12 19:04:58 drochner Exp $
 
-SHA1 (audiofile-0.2.6.tar.gz) = 94a6ab8e5122bc1971cf186e5a52d032811c9bc5
-RMD160 (audiofile-0.2.6.tar.gz) = b61fbc856768a5838ca5c0fad602f35205e8d037
-Size (audiofile-0.2.6.tar.gz) = 374688 bytes
-SHA1 (patch-aa) = 7c498a63fe055f1f7e16f45f655341a9b6708f71
+SHA1 (audiofile-0.2.7.tar.gz) = 1d8a5bb0ca63c7047436f136fb1fb98ea886e920
+RMD160 (audiofile-0.2.7.tar.gz) = 70327d38c6a9057045f5c59f67f45e2f6f9bf677
+Size (audiofile-0.2.7.tar.gz) = 502448 bytes
 SHA1 (patch-ab) = cd292a827aa18b9839305b2c62c3dbd526384430
-SHA1 (patch-ac) = fd2b8bb374b263b34a0e724c3e7de498f3ff2b7c
-SHA1 (patch-ad) = 3558f3f83e322efcf2c2a4e7afbc144b084e99f6
+SHA1 (patch-ae) = 6dc99026b3448f436a5ccf638598344018329661
+SHA1 (patch-af) = d0d9b737bf6b3c5ea497ea16160fcee7aad4ffe2
diff --git a/audio/libaudiofile/patches/patch-aa b/audio/libaudiofile/patches/patch-aa
deleted file mode 100644
index aeb9599c840..00000000000
--- a/audio/libaudiofile/patches/patch-aa
+++ /dev/null
@@ -1,13 +0,0 @@
-$NetBSD: patch-aa,v 1.10 2005/04/27 08:59:41 wiz Exp $
-
---- audiofile.m4.orig	2004-03-06 07:39:23.000000000 +0100
-+++ audiofile.m4
-@@ -9,7 +9,7 @@
- dnl AM_PATH_AUDIOFILE([MINIMUM-VERSION, [ACTION-IF-FOUND [, ACTION-IF-NOT-FOUND]]])
- dnl Test for Audio File Library, and define AUDIOFILE_CFLAGS and AUDIOFILE_LIBS.
- dnl
--AC_DEFUN(AM_PATH_AUDIOFILE,
-+AC_DEFUN([AM_PATH_AUDIOFILE],
- [dnl 
- dnl Get compiler flags and libraries from the audiofile-config script.
- dnl
diff --git a/audio/libaudiofile/patches/patch-ac b/audio/libaudiofile/patches/patch-ac
deleted file mode 100644
index 643a56c0fa2..00000000000
--- a/audio/libaudiofile/patches/patch-ac
+++ /dev/null
@@ -1,25 +0,0 @@
-$NetBSD: patch-ac,v 1.2 2009/12/01 10:44:02 drochner Exp $
-
---- libaudiofile/modules/msadpcm.c.orig	2004-03-06 07:39:23.000000000 +0100
-+++ libaudiofile/modules/msadpcm.c
-@@ -129,8 +129,7 @@ static int ms_adpcm_decode_block (ms_adp
- 	ms_adpcm_state	*state[2];
- 
- 	/* Calculate the number of bytes needed for decoded data. */
--	outputLength = msadpcm->samplesPerBlock * sizeof (int16_t) *
--		msadpcm->track->f.channelCount;
-+	outputLength = msadpcm->samplesPerBlock * sizeof (int16_t);

- 
- 	channelCount = msadpcm->track->f.channelCount;
- 
-@@ -180,8 +179,8 @@ static int ms_adpcm_decode_block (ms_adp
- 		The first two samples have already been 'decoded' in
- 		the block header.
- 	*/
--	samplesRemaining = (msadpcm->samplesPerBlock - 2) *
--		msadpcm->track->f.channelCount;
-+	samplesRemaining = msadpcm->samplesPerBlock

-+		- 2 * msadpcm->track->f.channelCount;

- 
- 	while (samplesRemaining > 0)
- 	{
diff --git a/audio/libaudiofile/patches/patch-ad b/audio/libaudiofile/patches/patch-ad
deleted file mode 100644
index 7c7fcfda01f..00000000000
--- a/audio/libaudiofile/patches/patch-ad
+++ /dev/null
@@ -1,111 +0,0 @@
-$NetBSD: patch-ad,v 1.2 2009/12/01 10:44:02 drochner Exp $
-
---- libaudiofile/wave.c.orig	2004-03-06 07:39:23.000000000 +0100
-+++ libaudiofile/wave.c
-@@ -199,11 +199,13 @@ static status ParseFormat (AFfilehandle 
- 		case WAVE_FORMAT_ADPCM:
- 		{
- 			u_int16_t	bitsPerSample, extraByteCount,
--					samplesPerBlock, numCoefficients;
-+					samplesPerBlock, numCoefficients,

-+					framesPerBlock;

- 			int		i;
- 			AUpvlist	pv;
- 			long		l;
- 			void		*v;
-+			int		minBlockLength;

- 
- 			if (track->f.channelCount != 1 &&
- 				track->f.channelCount != 2)
-@@ -216,11 +218,33 @@ static status ParseFormat (AFfilehandle 
- 			af_fread(&bitsPerSample, 1, 2, fp);
- 			bitsPerSample = LENDIAN_TO_HOST_INT16(bitsPerSample);
- 
-+			if (bitsPerSample != 4)

-+			{

-+				_af_error(AF_BAD_WIDTH,

-+					"bad sample width of %hd bits",

-+					bitsPerSample);

-+				return AF_FAIL;

-+			}

-+

- 			af_fread(&extraByteCount, 1, 2, fp);
- 			extraByteCount = LENDIAN_TO_HOST_INT16(extraByteCount);
- 
--			af_fread(&samplesPerBlock, 1, 2, fp);
--			samplesPerBlock = LENDIAN_TO_HOST_INT16(samplesPerBlock);
-+			af_fread(&framesPerBlock, 1, 2, fp);

-+			framesPerBlock = LENDIAN_TO_HOST_INT16(framesPerBlock);

-+

-+			minBlockLength = 7 * channelCount; /* header */

-+			if (framesPerBlock > 2)

-+				minBlockLength += ( ( framesPerBlock - 2 ) * channelCount + 1) / 2;

-+

-+			if (blockAlign < minBlockLength)

-+			{

-+				_af_error(AF_BAD_FRAMECNT,

-+					"blockAlign %hd too small for %hd samplesPerBlock",

-+					blockAlign, samplesPerBlock);

-+				return AF_FAIL;

-+			}

-+

-+			samplesPerBlock = framesPerBlock *channelCount;

- 
- 			af_fread(&numCoefficients, 1, 2, fp);
- 			numCoefficients = LENDIAN_TO_HOST_INT16(numCoefficients);
-@@ -242,6 +266,7 @@ static status ParseFormat (AFfilehandle 
- 				wave->msadpcmCoefficients[i][1] = a1;
- 			}
- 
-+

- 			track->f.sampleWidth = 16;
- 			track->f.sampleFormat = AF_SAMPFMT_TWOSCOMP;
- 			track->f.compressionType = AF_COMPRESSION_MS_ADPCM;
-@@ -277,18 +302,44 @@ static status ParseFormat (AFfilehandle 
- 		{
- 			AUpvlist	pv;
- 			long		l;
-+			int		minBlockLength;

- 
- 			u_int16_t	bitsPerSample, extraByteCount,
--					samplesPerBlock;
-+					samplesPerBlock, framesPerBlock;

- 
- 			af_fread(&bitsPerSample, 1, 2, fp);
- 			bitsPerSample = LENDIAN_TO_HOST_INT16(bitsPerSample);
- 
-+			if (bitsPerSample != 4)

-+			{

-+				_af_error(AF_BAD_WIDTH,

-+					"bad sample width of %hd bits",

-+					bitsPerSample);

-+				return AF_FAIL;

-+			}

-+

- 			af_fread(&extraByteCount, 1, 2, fp);
- 			extraByteCount = LENDIAN_TO_HOST_INT16(extraByteCount);
- 
--			af_fread(&samplesPerBlock, 1, 2, fp);
--			samplesPerBlock = LENDIAN_TO_HOST_INT16(samplesPerBlock);
-+			af_fread(&framesPerBlock, 1, 2, fp);

-+			framesPerBlock = LENDIAN_TO_HOST_INT16(framesPerBlock);

-+			samplesPerBlock = framesPerBlock * channelCount;

-+

-+			/* per channel, ima has blocks of len 4, the 1st has 1st sample, the others

-+			 * up to 8 samples per block,

-+			 * so number of later blocks is (nsamp-1 + 7)/8, total blocks/chan is

-+			 * (nsamp-1+7)/8 + 1 = (nsamp+14)/8

-+			 */

-+

-+			minBlockLength = ( framesPerBlock + 14 )/8 * 4 * channelCount;

-+

-+			if (blockAlign < minBlockLength)

-+			{

-+				_af_error(AF_BAD_FRAMECNT,

-+					"blockAlign %hd too small for %hd samplesPerBlock",

-+					blockAlign, samplesPerBlock);

-+				return AF_FAIL;

-+			}

- 
- 			track->f.sampleWidth = 16;
- 			track->f.sampleFormat = AF_SAMPFMT_TWOSCOMP;
diff --git a/audio/libaudiofile/patches/patch-ae b/audio/libaudiofile/patches/patch-ae
new file mode 100644
index 00000000000..18fb5c788b9
--- /dev/null
+++ b/audio/libaudiofile/patches/patch-ae
@@ -0,0 +1,32 @@
+$NetBSD: patch-ae,v 1.1 2010/04/12 19:04:58 drochner Exp $
+
+see Ubuntu bug 327018
+
+--- libaudiofile/util.c.orig	2010-01-26 19:56:24.000000000 +0000
++++ libaudiofile/util.c
+@@ -343,10 +343,10 @@ int _af_format_sample_size_uncompressed 
+ 	return size;
+ }
+ 
+-float _af_format_sample_size (_AudioFormat *fmt, bool stretch3to4)
++double _af_format_sample_size (_AudioFormat *fmt, bool stretch3to4)
+ {
+ 	int	compressionIndex;
+-	float	squishFactor;
++	double	squishFactor;
+ 
+ 	compressionIndex = _af_compression_index_from_id(fmt->compressionType);
+ 	squishFactor = _af_compression[compressionIndex].squishFactor;
+@@ -361,10 +361,10 @@ int _af_format_frame_size_uncompressed (
+ 		fmt->channelCount;
+ }
+ 
+-float _af_format_frame_size (_AudioFormat *fmt, bool stretch3to4)
++double _af_format_frame_size (_AudioFormat *fmt, bool stretch3to4)
+ {
+ 	int	compressionIndex;
+-	float	squishFactor;
++	double	squishFactor;
+ 
+ 	compressionIndex = _af_compression_index_from_id(fmt->compressionType);
+ 	squishFactor = _af_compression[compressionIndex].squishFactor;
diff --git a/audio/libaudiofile/patches/patch-af b/audio/libaudiofile/patches/patch-af
new file mode 100644
index 00000000000..6f824106027
--- /dev/null
+++ b/audio/libaudiofile/patches/patch-af
@@ -0,0 +1,16 @@
+$NetBSD: patch-af,v 1.1 2010/04/12 19:04:58 drochner Exp $
+
+--- libaudiofile/util.h.orig	2010-01-26 19:55:15.000000000 +0000
++++ libaudiofile/util.h
+@@ -55,9 +55,9 @@ _Track *_af_filehandle_get_track (AFfile
+ 
+ bool _af_unique_ids (int *ids, int nids, char *idname, int iderr);
+ 
+-float _af_format_frame_size (_AudioFormat *format, bool stretch3to4);
++double _af_format_frame_size (_AudioFormat *format, bool stretch3to4);
+ int _af_format_frame_size_uncompressed (_AudioFormat *format, bool stretch3to4);
+-float _af_format_sample_size (_AudioFormat *format, bool stretch3to4);
++double _af_format_sample_size (_AudioFormat *format, bool stretch3to4);
+ int _af_format_sample_size_uncompressed (_AudioFormat *format, bool stretch3to4);
+ 
+ status _af_set_sample_format (_AudioFormat *f, int sampleFormat, int sampleWidth);