apply patches

doc/bugs.html

@@ -31,7 +31,7 @@
 	<UL>
 	<LI>	Compilation problems on new platforms.
 	<LI>	Errors being detected during the `make check' process.
-	<LI>	Segmentation faults occuring inside libsndfile.
+	<LI>	Segmentation faults occurring inside libsndfile.
 	<LI>	libsndfile hanging when opening a file.
 	<LI>	Supported sound file types being incorrectly read or written.
 	<LI>	Omissions, errors or spelling mistakes in the documentation.

programs/sndfile-convert.c

@@ -317,7 +317,7 @@ main (int argc, char * argv [])
 	if ((sfinfo.format & SF_FORMAT_SUBMASK) == SF_FORMAT_GSM610 && sfinfo.samplerate != 8000)
 	{	printf (
 			"WARNING: GSM 6.10 data format only supports 8kHz sample rate. The converted\n"
-			"ouput file will contain the input data converted to the GSM 6.10 data format\n"
+			"output file will contain the input data converted to the GSM 6.10 data format\n"
 			"but not re-sampled.\n"
 			) ;
 		} ;

programs/sndfile-deinterleave.c

@@ -89,6 +89,13 @@ main (int argc, char **argv)
 		exit (1) ;
 		} ;
 
+	if (sfinfo.channels > MAX_CHANNELS)
+	{	printf ("\nError : Input file '%s' has too many (%d) channels. Limit is %d.\n",
+			argv [1], sfinfo.channels, MAX_CHANNELS) ;
+		exit (1) ;
+		} ;
+
+
 	state.channels = sfinfo.channels ;
 	sfinfo.channels = 1 ;
 

src/aiff.c

@@ -1905,7 +1905,7 @@ aiff_read_chanmap (SF_PRIVATE * psf, unsigned dword)
 		psf_binheader_readf (psf, "j", dword - bytesread) ;
 
 	if (map_info->channel_map != NULL)
-	{	size_t chanmap_size = psf->sf.channels * sizeof (psf->channel_map [0]) ;
+	{	size_t chanmap_size = SF_MIN (psf->sf.channels, layout_tag & 0xffff) * sizeof (psf->channel_map [0]) ;
 
 		free (psf->channel_map) ;
 

src/alaw.c

@@ -19,6 +19,7 @@
 #include	"sfconfig.h"
 
 #include	<math.h>
+#include	<limits.h>
 
 #include	"sndfile.h"
 #include	"common.h"
@@ -326,7 +327,9 @@ s2alaw_array (const short *ptr, int count, unsigned char *buffer)
 static inline void
 i2alaw_array (const int *ptr, int count, unsigned char *buffer)
 {	while (--count >= 0)
-	{	if (ptr [count] >= 0)
+	{	if (ptr [count] == INT_MIN)
+			buffer [count] = alaw_encode [INT_MAX >> (16 + 4)] ;
+		else if (ptr [count] >= 0)
 			buffer [count] = alaw_encode [ptr [count] >> (16 + 4)] ;
 		else
 			buffer [count] = 0x7F & alaw_encode [- ptr [count] >> (16 + 4)] ;
@@ -346,7 +349,9 @@ f2alaw_array (const float *ptr, int count, unsigned char *buffer, float normfact
 static inline void
 d2alaw_array (const double *ptr, int count, unsigned char *buffer, double normfact)
 {	while (--count >= 0)
-	{	if (ptr [count] >= 0)
+	{	if (!isfinite (ptr [count]))
+			buffer [count] = 0 ;
+		else if (ptr [count] >= 0)
 			buffer [count] = alaw_encode [lrint (normfact * ptr [count])] ;
 		else
 			buffer [count] = 0x7F & alaw_encode [- lrint (normfact * ptr [count])] ;

src/common.c

@@ -675,15 +675,15 @@ psf_binheader_writef (SF_PRIVATE *psf, const char *format, ...)
 					/* Write a C string (guaranteed to have a zero terminator). */
 					strptr = va_arg (argptr, char *) ;
 					size = strlen (strptr) + 1 ;
-					size += (size & 1) ;
 
-					if (psf->header.indx + (sf_count_t) size >= psf->header.len && psf_bump_header_allocation (psf, 16))
+					if (psf->header.indx + 4 + (sf_count_t) size + (sf_count_t) (size & 1) > psf->header.len && psf_bump_header_allocation (psf, 4 + size + (size & 1)))
 						return count ;
 
 					if (psf->rwf_endian == SF_ENDIAN_BIG)
-						header_put_be_int (psf, size) ;
+						header_put_be_int (psf, size + (size & 1)) ;
 					else
-						header_put_le_int (psf, size) ;
+						header_put_le_int (psf, size + (size & 1)) ;
+					size += (size & 1) ;
 					memcpy (&(psf->header.ptr [psf->header.indx]), strptr, size) ;
 					psf->header.indx += size ;
 					psf->header.ptr [psf->header.indx - 1] = 0 ;

src/common.h

@@ -725,6 +725,7 @@ enum
 	SFE_FLAC_INIT_DECODER,
 	SFE_FLAC_LOST_SYNC,
 	SFE_FLAC_BAD_SAMPLE_RATE,
+	SFE_FLAC_CHANNEL_COUNT_CHANGED,
 	SFE_FLAC_UNKOWN_ERROR,
 
 	SFE_WVE_NOT_WVE,

src/double64.c

@@ -91,7 +91,7 @@ int
 double64_init	(SF_PRIVATE *psf)
 {	static int double64_caps ;
 
-	if (psf->sf.channels < 1)
+	if (psf->sf.channels < 1 || psf->sf.channels > SF_MAX_CHANNELS)
 	{	psf_log_printf (psf, "double64_init : internal error : channels = %d\n", psf->sf.channels) ;
 		return SFE_INTERNAL ;
 		} ;

src/flac.c

@@ -169,6 +169,14 @@ flac_buffer_copy (SF_PRIVATE *psf)
 	const int32_t* const *buffer = pflac->wbuffer ;
 	unsigned i = 0, j, offset, channels, len ;
 
+	if (psf->sf.channels != (int) frame->header.channels)
+	{	psf_log_printf (psf, "Error: FLAC frame changed from %d to %d channels\n"
+									"Nothing to do but to error out.\n" ,
+									psf->sf.channels, frame->header.channels) ;
+		psf->error = SFE_FLAC_CHANNEL_COUNT_CHANGED ;
+		return 0 ;
+		} ;
+
 	/*
 	**	frame->header.blocksize is variable and we're using a constant blocksize
 	**	of FLAC__MAX_BLOCK_SIZE.
@@ -202,7 +210,6 @@ flac_buffer_copy (SF_PRIVATE *psf)
 		return 0 ;
 		} ;
 
-
 	len = SF_MIN (pflac->len, frame->header.blocksize) ;
 
 	if (pflac->remain % channels != 0)
@@ -430,11 +437,23 @@ sf_flac_meta_get_vorbiscomments (SF_PRIVATE *psf, const FLAC__StreamMetadata *me
 static void
 sf_flac_meta_callback (const FLAC__StreamDecoder * UNUSED (decoder), const FLAC__StreamMetadata *metadata, void *client_data)
 {	SF_PRIVATE *psf = (SF_PRIVATE*) client_data ;
-	FLAC_PRIVATE* pflac = (FLAC_PRIVATE*) psf->codec_data ;
+	int bitwidth = 0 ;
-	int bitwidth = 0, i ;
 
 	switch (metadata->type)
 	{	case FLAC__METADATA_TYPE_STREAMINFO :
+			if (psf->sf.channels > 0 && psf->sf.channels != (int) metadata->data.stream_info.channels)
+			{	psf_log_printf (psf, "Error: FLAC stream changed from %d to %d channels\n"
+									"Nothing to do but to error out.\n" ,
+									psf->sf.channels, metadata->data.stream_info.channels) ;
+				psf->error = SFE_FLAC_CHANNEL_COUNT_CHANGED ;
+				return ;
+				} ;
+
+			if (psf->sf.channels > 0 && psf->sf.samplerate != (int) metadata->data.stream_info.sample_rate)
+			{	psf_log_printf (psf, "Warning: FLAC stream changed sample rates from %d to %d.\n"
+									"Carrying on as if nothing happened.",
+									psf->sf.samplerate, metadata->data.stream_info.sample_rate) ;
+				} ;
 			psf->sf.channels = metadata->data.stream_info.channels ;
 			psf->sf.samplerate = metadata->data.stream_info.sample_rate ;
 			psf->sf.frames = metadata->data.stream_info.total_samples ;
@@ -468,12 +487,6 @@ sf_flac_meta_callback (const FLAC__StreamDecoder * UNUSED (decoder), const FLAC_
 
 			if (bitwidth > 0)
 				psf_log_printf (psf, "  Bit width   : %d\n", bitwidth) ;
-
-
-			for (i = 0 ; i < psf->sf.channels ; i++)
-				pflac->rbuffer [i] = calloc (FLAC__MAX_BLOCK_SIZE, sizeof (int32_t)) ;
-
-			pflac->wbuffer = (const int32_t* const*) pflac->rbuffer ;
 			break ;
 
 		case FLAC__METADATA_TYPE_VORBIS_COMMENT :
@@ -835,7 +848,9 @@ flac_read_header (SF_PRIVATE *psf)
 
 	psf_log_printf (psf, "End\n") ;
 
-	if (psf->error == 0)
+	if (psf->error != 0)
+		FLAC__stream_decoder_delete (pflac->fsd) ;
+	else
 	{	FLAC__uint64 position ;
 
 		FLAC__stream_decoder_get_decode_position (pflac->fsd, &position) ;

src/ogg.c

@@ -193,7 +193,7 @@ ogg_stream_classify (SF_PRIVATE *psf, OGG_PRIVATE* odata)
 			break ;
 		} ;
 
-	psf_log_printf (psf, "This Ogg bitstream contains some uknown data type.\n") ;
+	psf_log_printf (psf, "This Ogg bitstream contains some unknown data type.\n") ;
 	return SFE_UNIMPLEMENTED ;
 } /* ogg_stream_classify */
 

src/rf64.c

@@ -339,6 +339,12 @@ rf64_read_header (SF_PRIVATE *psf, int *blockalign, int *framesperblock)
 					} ;
 				break ;
 
+			case JUNK_MARKER :
+			case PAD_MARKER :
+				psf_log_printf (psf, "%M : %d\n", marker, chunk_size) ;
+				psf_binheader_readf (psf, "j", chunk_size) ;
+				break ;
+
 			default :
 					if (chunk_size >= 0xffff0000)
 					{	psf_log_printf (psf, "*** Unknown chunk marker (%X) at position %D with length %u. Exiting parser.\n", marker, psf_ftell (psf) - 8, chunk_size) ;
@@ -659,7 +665,7 @@ rf64_write_header (SF_PRIVATE *psf, int calc_length)
 
 	if (wpriv->rf64_downgrade && psf->filelength < RIFF_DOWNGRADE_BYTES)
 	{	psf_binheader_writef (psf, "etm8m", RIFF_MARKER, (psf->filelength < 8) ? 8 : psf->filelength - 8, WAVE_MARKER) ;
-		psf_binheader_writef (psf, "m4884", JUNK_MARKER, 20, 0, 0, 0, 0) ;
+		psf_binheader_writef (psf, "m4z", JUNK_MARKER, 24, 24) ;
 		add_fact_chunk = 1 ;
 		}
 	else
@@ -735,9 +741,10 @@ rf64_write_header (SF_PRIVATE *psf, int calc_length)
 
 #endif
 
+	/* Padding may be needed if string data sizes change. */
 	pad_size = psf->dataoffset - 16 - psf->header.indx ;
 	if (pad_size >= 0)
-		psf_binheader_writef (psf, "m4z", PAD_MARKER, pad_size, make_size_t (pad_size)) ;
+		psf_binheader_writef (psf, "m4z", PAD_MARKER, (unsigned int) pad_size, make_size_t (pad_size)) ;
 
 	if (wpriv->rf64_downgrade && (psf->filelength < RIFF_DOWNGRADE_BYTES))
 		psf_binheader_writef (psf, "tm8", data_MARKER, psf->datalength) ;

src/sndfile.c

@@ -245,6 +245,7 @@ ErrorStruct SndfileErrors [] =
 	{	SFE_FLAC_INIT_DECODER	, "Error : problem with initialization of the flac decoder." },
 	{	SFE_FLAC_LOST_SYNC		, "Error : flac decoder lost sync." },
 	{	SFE_FLAC_BAD_SAMPLE_RATE, "Error : flac does not support this sample rate." },
+	{	SFE_FLAC_CHANNEL_COUNT_CHANGED, "Error : flac channel changed mid stream." },
 	{	SFE_FLAC_UNKOWN_ERROR	, "Error : unknown error in flac decoder." },
 
 	{	SFE_WVE_NOT_WVE			, "Error : not a WVE file." },

src/ulaw.c

@@ -19,6 +19,7 @@
 #include	"sfconfig.h"
 
 #include	<math.h>
+#include	<limits.h>
 
 #include	"sndfile.h"
 #include	"common.h"
@@ -827,7 +828,9 @@ s2ulaw_array (const short *ptr, int count, unsigned char *buffer)
 static inline void
 i2ulaw_array (const int *ptr, int count, unsigned char *buffer)
 {	while (--count >= 0)
-	{	if (ptr [count] >= 0)
+	{	if (ptr [count] == INT_MIN)
+			buffer [count] = ulaw_encode [INT_MAX >> (16 + 2)] ;
+		else if (ptr [count] >= 0)
 			buffer [count] = ulaw_encode [ptr [count] >> (16 + 2)] ;
 		else
 			buffer [count] = 0x7F & ulaw_encode [-ptr [count] >> (16 + 2)] ;
@@ -847,7 +850,9 @@ f2ulaw_array (const float *ptr, int count, unsigned char *buffer, float normfact
 static inline void
 d2ulaw_array (const double *ptr, int count, unsigned char *buffer, double normfact)
 {	while (--count >= 0)
-	{	if (ptr [count] >= 0)
+	{	if (!isfinite (ptr [count]))
+			buffer [count] = 0 ;
+		else if (ptr [count] >= 0)
 			buffer [count] = ulaw_encode [lrint (normfact * ptr [count])] ;
 		else
 			buffer [count] = 0x7F & ulaw_encode [- lrint (normfact * ptr [count])] ;

src/wav.c

@@ -1,5 +1,5 @@
 /*
-** Copyright (C) 1999-2016 Erik de Castro Lopo <erikd@mega-nerd.com>
+** Copyright (C) 1999-2019 Erik de Castro Lopo <erikd@mega-nerd.com>
 ** Copyright (C) 2004-2005 David Viens <davidv@plogue.com>
 **
 ** This program is free software; you can redistribute it and/or modify
@@ -1094,6 +1094,12 @@ wav_write_header (SF_PRIVATE *psf, int calc_length)
 		psf_binheader_writef (psf, "44", 0, 0) ; /* SMTPE format */
 		psf_binheader_writef (psf, "44", psf->instrument->loop_count, 0) ;
 
+		/* Make sure we don't read past the loops array end. */
+		if (psf->instrument->loop_count > ARRAY_LEN (psf->instrument->loops))
+			psf->instrument->loop_count = ARRAY_LEN (psf->instrument->loops) ;
+
+		/* Loop count is signed 16 bit number so we limit it range to something sensible. */
+		psf->instrument->loop_count &= 0x7fff ;
 		for (tmp = 0 ; tmp < psf->instrument->loop_count ; tmp++)
 		{	int type ;
 

src/wavlike.c

@@ -161,7 +161,7 @@ wavlike_read_fmt_chunk (SF_PRIVATE *psf, int fmtsize)
 	{	psf_log_printf (psf, "  Bit Width     : 24\n") ;
 
 		psf_log_printf (psf, "\n"
-			"  Ambiguous information in 'fmt ' chunk. Possibile file types:\n"
+			"  Ambiguous information in 'fmt ' chunk. Possible file types:\n"
 			"    0) Invalid IEEE float file generated by Syntrillium's Cooledit!\n"
 			"    1) File generated by ALSA's arecord containing 24 bit samples in 32 bit containers.\n"
 			"    2) 24 bit file with incorrect Block Align value.\n"