3 files changed, 64 insertions, 13 deletions

diff --git a/audio/sox/Makefile b/audio/sox/Makefile
index 880383e723f6..95c7465620b2 100644
--- a/audio/sox/Makefile
+++ b/audio/sox/Makefile
@@ -6,7 +6,7 @@
 #
 
 PORTNAME=	sox
-PORTVERSION=	14.3.1
+PORTVERSION=	14.3.2
 CATEGORIES=	audio
 MASTER_SITES=	SF
 
@@ -23,6 +23,8 @@ GNU_CONFIGURE=	yes
 USE_LDCONFIG=	yes
 USE_AUTOTOOLS=	libltdl
 USE_GNOME=	pkgconfig
+LICENSE_COMB=	dual
+LICENSE=	LGPL21 GPLv2
 
 # Default LAME to off for packages so we don't end up RESTRICTED
 .if defined(PACKAGE_BUILDING)
@@ -51,7 +53,7 @@ OPTIONS=	ALSA "ALSA output driver" off \
 
 CPPFLAGS+=	-I${LOCALBASE}/include
 LDFLAGS+=	-L${LOCALBASE}/lib ${PTHREAD_LIBS}
-CONFIGURE_ENV+=	CPPFLAGS="${CPPFLAGS}" LDFLAGS="${LDFLAGS}"
+CONFIGURE_ENV+=	LDFLAGS="${LDFLAGS}"
 CONFIGURE_ARGS+=	--with-pkgconfigdir="${PREFIX}/libdata/pkgconfig"
 CONFIGURE_ARGS+=	--with-distro="${CONFIGURE_TARGET} ${PKGNAME}"
 
@@ -147,8 +149,9 @@ LIB_DEPENDS+=		gsm.1:${PORTSDIR}/audio/gsm
 .else
 CONFIGURE_ARGS+=	--with-gsm
 # We cannot ask for gsm to be enabled without it trying to use external gsm,
-# so workaround with the below line
-CONFIGURE_ENV+=		ac_cv_header_gsm_h=no
+# so force the autoconf checks to fail
+CONFIGURE_ENV+=	ac_cv_header_gsm_h=no ac_cv_header_gsm_gsm_h=no \
+		ac_cv_lib_gsm_gsm_create=no
 .endif
 
 .if defined(WITH_ID3TAG)
@@ -165,11 +168,4 @@ LIB_DEPENDS+=		wavpack.2:${PORTSDIR}/audio/wavpack
 CONFIGURE_ARGS+=	--without-wavpack
 .endif
 
-# Tell configure that libgsm's headers are in $LOCALDIR/include
-post-patch:
-	@${REINPLACE_CMD} -Ee 's![[:<:]]gsm/gsm.h!gsm.h!' ${WRKSRC}/configure \
-		${WRKSRC}/src/gsm.c ${WRKSRC}/src/wav.c
-	@${REINPLACE_CMD} -Ee 's!ffmpeg/avformat.h!libavformat/avformat.h!' \
-		${WRKSRC}/configure ${WRKSRC}/src/ffmpeg.c
-
 .include <bsd.port.post.mk>
diff --git a/audio/sox/distinfo b/audio/sox/distinfo
index a42ac69c28d8..cba6ab254707 100644
--- a/audio/sox/distinfo
+++ b/audio/sox/distinfo
@@ -1,2 +1,2 @@
-SHA256 (sox-14.3.1.tar.gz) = c4be9ebdb1e77a3bc9f3ee1f3fd274a1fe7b162a174fcc74874b231f3f1ace54
-SIZE (sox-14.3.1.tar.gz) = 1044595
+SHA256 (sox-14.3.2.tar.gz) = 0668cc087c346f7c4084ae294c676a11ddefb462974bc3f67be30d789c850e8f
+SIZE (sox-14.3.2.tar.gz) = 1110175
diff --git a/audio/sox/files/patch-ffmpeg b/audio/sox/files/patch-ffmpeg
new file mode 100644
index 000000000000..fd511e7348cc
--- /dev/null
+++ b/audio/sox/files/patch-ffmpeg
@@ -0,0 +1,55 @@
+*** diff
+
+Update to use the new ffmpeg library API, but still work with 
+the old one.
+
+--- src/ffmpeg.c	2011-06-07 00:29:34.000000000 -0500
++++ src/ffmpeg.c	2011-06-07 00:34:12.830996927 -0500
+@@ -50,6 +50,11 @@
+ #include <ctype.h>
+ #include "ffmpeg.h"
+ 
++#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(52, 64, 0)
++#define AVMEDIA_TYPE_AUDIO CODEC_TYPE_AUDIO
++#define AV_PKT_FLAG_KEY PKT_FLAG_KEY
++#endif
++ 		    
+ /* Private data for ffmpeg files */
+ typedef struct {
+   int audio_index;
+@@ -91,7 +96,7 @@ static int stream_component_open(priv_t 
+ 
+   if (!codec || avcodec_open(enc, codec) < 0)
+     return -1;
+-  if (enc->codec_type != CODEC_TYPE_AUDIO) {
++  if (enc->codec_type != AVMEDIA_TYPE_AUDIO) {
+     lsx_fail("ffmpeg CODEC %x is not an audio CODEC", enc->codec_type);
+     return -1;
+   }
+@@ -182,7 +187,7 @@ static int startread(sox_format_t * ft)
+   /* Find audio stream (FIXME: allow different stream to be selected) */
+   for (i = 0; (unsigned)i < ffmpeg->ctxt->nb_streams; i++) {
+     AVCodecContext *enc = ffmpeg->ctxt->streams[i]->codec;
+-    if (enc->codec_type == CODEC_TYPE_AUDIO && ffmpeg->audio_index < 0) {
++    if (enc->codec_type == AVMEDIA_TYPE_AUDIO && ffmpeg->audio_index < 0) {
+       ffmpeg->audio_index = i;
+       break;
+     }
+@@ -273,7 +278,7 @@ static AVStream *add_audio_stream(sox_fo
+ 
+   c = st->codec;
+   c->codec_id = codec_id;
+-  c->codec_type = CODEC_TYPE_AUDIO;
++  c->codec_type = AVMEDIA_TYPE_AUDIO;
+ 
+   /* put sample parameters */
+   c->bit_rate = 256000;  /* FIXME: allow specification */
+@@ -423,7 +428,7 @@ static size_t write_samples(sox_format_t
+       av_init_packet(&pkt);
+       pkt.size = avcodec_encode_audio(c, ffmpeg->audio_buf_aligned, AVCODEC_MAX_AUDIO_FRAME_SIZE, ffmpeg->samples);
+       pkt.pts = av_rescale_q(c->coded_frame->pts, c->time_base, ffmpeg->audio_st->time_base);
+-      pkt.flags |= PKT_FLAG_KEY;
++      pkt.flags |= AV_PKT_FLAG_KEY;
+       pkt.stream_index = ffmpeg->audio_st->index;
+       pkt.data = ffmpeg->audio_buf_aligned;
+