version bump

(Portage version: 2.1.2-r12)

4 files changed, 266 insertions, 1 deletions

diff --git a/media-libs/gst-plugins-base/ChangeLog b/media-libs/gst-plugins-base/ChangeLog
index ba7d2d1b4fca..a41fcec4822b 100644
--- a/media-libs/gst-plugins-base/ChangeLog
+++ b/media-libs/gst-plugins-base/ChangeLog
@@ -1,6 +1,13 @@
 # ChangeLog for media-libs/gst-plugins-base
 # Copyright 1999-2007 Gentoo Foundation; Distributed under the GPL v2
-# $Header: /var/cvsroot/gentoo-x86/media-libs/gst-plugins-base/ChangeLog,v 1.49 2007/02/16 12:55:17 uberlord Exp $
+# $Header: /var/cvsroot/gentoo-x86/media-libs/gst-plugins-base/ChangeLog,v 1.50 2007/03/16 10:11:28 zaheerm Exp $
+
+*gst-plugins-base-0.10.12 (16 Mar 2007)
+
+  16 Mar 2007; Zaheer Abbas Merali <zaheerm@gentoo.org>
+  +files/gst-plugins-base.audioresample.patch,
+  +gst-plugins-base-0.10.12.ebuild:
+  version bump
 
   16 Feb 2007; Roy Marples <uberlord@gentoo.org>
   gst-plugins-base-0.10.11.ebuild:
diff --git a/media-libs/gst-plugins-base/files/digest-gst-plugins-base-0.10.12 b/media-libs/gst-plugins-base/files/digest-gst-plugins-base-0.10.12
new file mode 100644
index 000000000000..9f429b6590ae
--- /dev/null
+++ b/media-libs/gst-plugins-base/files/digest-gst-plugins-base-0.10.12
@@ -0,0 +1,3 @@
+MD5 0ee35455a4eb507bcfbfcd44d9e15d1e gst-plugins-base-0.10.12.tar.bz2 1460658
+RMD160 30ab89cb22b0e596749a651eea86421c72ba0425 gst-plugins-base-0.10.12.tar.bz2 1460658
+SHA256 b88a85b21499bd064a531a3b5d06ae69ee025ad5f88b6e86ed4af245509247ee gst-plugins-base-0.10.12.tar.bz2 1460658
diff --git a/media-libs/gst-plugins-base/files/gst-plugins-base.audioresample.patch b/media-libs/gst-plugins-base/files/gst-plugins-base.audioresample.patch
new file mode 100644
index 000000000000..46a28fd3df24
--- /dev/null
+++ b/media-libs/gst-plugins-base/files/gst-plugins-base.audioresample.patch
@@ -0,0 +1,200 @@
+--- gst/audioresample/gstaudioresample.c	2006/10/28 16:00:51	1.22
++++ gst/audioresample/gstaudioresample.c	2007/03/15 10:52:21	1.25
+@@ -194,6 +194,8 @@
+   gst_pad_set_bufferalloc_function (trans->sinkpad, NULL);
+ 
+   audioresample->filter_length = DEFAULT_FILTERLEN;
++
++  audioresample->need_discont = FALSE;
+ }
+ 
+ /* vmethods */
+@@ -371,7 +373,7 @@
+   gboolean use_internal = FALSE;        /* whether we use the internal state */
+   gboolean ret = TRUE;
+ 
+-  GST_DEBUG_OBJECT (base, "asked to transform size %d in direction %s",
++  GST_LOG_OBJECT (base, "asked to transform size %d in direction %s",
+       size, direction == GST_PAD_SINK ? "SINK" : "SRC");
+   if (direction == GST_PAD_SINK) {
+     sinkcaps = caps;
+@@ -406,7 +408,7 @@
+ 
+   /* we make room for one extra sample, given that the resampling filter
+    * can output an extra one for non-integral i_rate/o_rate */
+-  GST_DEBUG_OBJECT (base, "transformed size %d to %d", size, *othersize);
++  GST_LOG_OBJECT (base, "transformed size %d to %d", size, *othersize);
+ 
+   if (!use_internal) {
+     resample_free (state);
+@@ -492,8 +494,7 @@
+   r = audioresample->resample;
+ 
+   outsize = resample_get_output_size (r);
+-  GST_DEBUG_OBJECT (audioresample, "audioresample can give me %d bytes",
+-      outsize);
++  GST_LOG_OBJECT (audioresample, "audioresample can give me %d bytes", outsize);
+ 
+   /* protect against mem corruption */
+   if (outsize > GST_BUFFER_SIZE (outbuf)) {
+@@ -540,8 +541,8 @@
+   /* check for possible mem corruption */
+   if (outsize > GST_BUFFER_SIZE (outbuf)) {
+     /* this is an error that when it happens, would need fixing in the
+-     * resample library; we told
+-     * it we wanted only GST_BUFFER_SIZE (outbuf), and it gave us more ! */
++     * resample library; we told it we wanted only GST_BUFFER_SIZE (outbuf),
++     * and it gave us more ! */
+     GST_WARNING_OBJECT (audioresample,
+         "audioresample, you memory corrupting bastard. "
+         "you gave me outsize %d while my buffer was size %d",
+@@ -556,9 +557,51 @@
+   }
+   GST_BUFFER_SIZE (outbuf) = outsize;
+ 
++  if (G_UNLIKELY (audioresample->need_discont)) {
++    GST_DEBUG_OBJECT (audioresample,
++        "marking this buffer with the DISCONT flag");
++    GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
++    audioresample->need_discont = FALSE;
++  }
++
++  GST_LOG_OBJECT (audioresample, "transformed to buffer of %ld bytes, ts %"
++      GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %"
++      G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
++      outsize, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
++      GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)),
++      GST_BUFFER_OFFSET (outbuf), GST_BUFFER_OFFSET_END (outbuf));
++
++
+   return GST_FLOW_OK;
+ }
+ 
++/* llabs() is C99, so we might not have it; just use a simple macro... */
++#define LLABS(x) ((x>0)?x:-x)
++static gboolean
++audioresample_check_discont (GstAudioresample * audioresample,
++    GstClockTime timestamp)
++{
++  if (timestamp != GST_CLOCK_TIME_NONE &&
++      audioresample->prev_ts != GST_CLOCK_TIME_NONE &&
++      audioresample->prev_duration != GST_CLOCK_TIME_NONE &&
++      timestamp != audioresample->prev_ts + audioresample->prev_duration) {
++    /* Potentially a discontinuous buffer. However, it turns out that many
++     * elements generate imperfect streams due to rounding errors, so we permit
++     * a small error (up to one sample) without triggering a filter 
++     * flush/restart (if triggered incorrectly, this will be audible) */
++    GstClockTimeDiff diff = timestamp -
++        (audioresample->prev_ts + audioresample->prev_duration);
++
++    if (LLABS (diff) > GST_SECOND / audioresample->i_rate) {
++      GST_WARNING_OBJECT (audioresample,
++          "encountered timestamp discontinuity of %" G_GINT64_FORMAT, diff);
++      return TRUE;
++    }
++  }
++
++  return FALSE;
++}
++
+ static GstFlowReturn
+ audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf,
+     GstBuffer * outbuf)
+@@ -576,7 +619,22 @@
+   size = GST_BUFFER_SIZE (inbuf);
+   timestamp = GST_BUFFER_TIMESTAMP (inbuf);
+ 
+-  GST_DEBUG_OBJECT (audioresample, "got buffer of %ld bytes", size);
++  GST_LOG_OBJECT (audioresample, "transforming buffer of %ld bytes, ts %"
++      GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %"
++      G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
++      size, GST_TIME_ARGS (timestamp),
++      GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf)),
++      GST_BUFFER_OFFSET (inbuf), GST_BUFFER_OFFSET_END (inbuf));
++
++  /* check for timestamp discontinuities and flush/reset if needed */
++  if (G_UNLIKELY (audioresample_check_discont (audioresample, timestamp))) {
++    /* Flush internal samples */
++    audioresample_pushthrough (audioresample);
++    /* Inform downstream element about discontinuity */
++    audioresample->need_discont = TRUE;
++    /* We want to recalculate the offset */
++    audioresample->ts_offset = -1;
++  }
+ 
+   if (audioresample->ts_offset == -1) {
+     /* if we don't know the initial offset yet, calculate it based on the 
+@@ -584,19 +642,21 @@
+     if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
+       GstClockTime stime;
+ 
+-      /* offset used to calculate the timestamps. We use the sample offset for this
+-       * to make it more accurate. We want the first buffer to have the same timestamp
+-       * as the incomming timestamp. */
++      /* offset used to calculate the timestamps. We use the sample offset for
++       * this to make it more accurate. We want the first buffer to have the
++       * same timestamp as the incoming timestamp. */
+       audioresample->next_ts = timestamp;
+       audioresample->ts_offset =
+           gst_util_uint64_scale_int (timestamp, r->o_rate, GST_SECOND);
+-      /* offset used to set as the buffer offset, this offset is always relative
+-       * to the stream time, note that timestamp is not... */
++      /* offset used to set as the buffer offset, this offset is always
++       * relative to the stream time, note that timestamp is not... */
+       stime = (timestamp - base->segment.start) + base->segment.time;
+       audioresample->offset =
+           gst_util_uint64_scale_int (stime, r->o_rate, GST_SECOND);
+     }
+   }
++  audioresample->prev_ts = timestamp;
++  audioresample->prev_duration = GST_BUFFER_DURATION (inbuf);
+ 
+   /* need to memdup, resample takes ownership. */
+   datacopy = g_memdup (data, size);
+@@ -618,17 +678,25 @@
+   r = audioresample->resample;
+ 
+   outsize = resample_get_output_size (r);
+-  if (outsize == 0)
++  if (outsize == 0) {
++    GST_DEBUG_OBJECT (audioresample, "no internal buffers needing flush");
+     goto done;
++  }
++
++  trans = GST_BASE_TRANSFORM (audioresample);
+ 
+-  outbuf = gst_buffer_new_and_alloc (outsize);
++  res = gst_pad_alloc_buffer (trans->srcpad, GST_BUFFER_OFFSET_NONE, outsize,
++      GST_PAD_CAPS (trans->srcpad), &outbuf);
++  if (G_UNLIKELY (res != GST_FLOW_OK)) {
++    GST_WARNING_OBJECT (audioresample, "failed allocating buffer of %d bytes",
++        outsize);
++    goto done;
++  }
+ 
+   res = audioresample_do_output (audioresample, outbuf);
+-  if (res != GST_FLOW_OK)
++  if (G_UNLIKELY (res != GST_FLOW_OK))
+     goto done;
+ 
+-  trans = GST_BASE_TRANSFORM (audioresample);
+-
+   res = gst_pad_push (trans->srcpad, outbuf);
+ 
+ done:
+--- gst/audioresample/gstaudioresample.h	2006/06/01 19:19:50	1.6
++++ gst/audioresample/gstaudioresample.h	2007/03/14 17:16:30	1.7
+@@ -53,10 +53,12 @@
+   GstCaps *srccaps, *sinkcaps;
+ 
+   gboolean passthru;
++  gboolean need_discont;
+ 
+   guint64 offset;
+   guint64 ts_offset;
+   GstClockTime next_ts;
++  GstClockTime prev_ts, prev_duration;
+   int channels;
+ 
+   int i_rate;
+
diff --git a/media-libs/gst-plugins-base/gst-plugins-base-0.10.12.ebuild b/media-libs/gst-plugins-base/gst-plugins-base-0.10.12.ebuild
new file mode 100644
index 000000000000..69552f2b14ae
--- /dev/null
+++ b/media-libs/gst-plugins-base/gst-plugins-base-0.10.12.ebuild
@@ -0,0 +1,55 @@
+# Copyright 1999-2007 Gentoo Foundation
+# Distributed under the terms of the GNU General Public License v2
+# $Header: /var/cvsroot/gentoo-x86/media-libs/gst-plugins-base/gst-plugins-base-0.10.12.ebuild,v 1.1 2007/03/16 10:11:28 zaheerm Exp $
+
+# order is important, gnome2 after gst-plugins
+inherit gst-plugins-base gst-plugins10 gnome2 eutils flag-o-matic libtool
+
+DESCRIPTION="Basepack of plugins for gstreamer"
+HOMEPAGE="http://gstreamer.net/"
+SRC_URI="http://gstreamer.freedesktop.org/src/${PN}/${P}.tar.bz2"
+
+LICENSE="GPL-2"
+KEYWORDS="~alpha ~amd64 ~arm ~hppa ~ia64 ~ppc ~ppc64 ~sh ~sparc ~x86 ~x86-fbsd"
+IUSE="alsa esd oss X xv"
+
+RDEPEND=">=dev-libs/glib-2.8
+	 >=media-libs/gstreamer-0.10.11
+	 >=dev-libs/liboil-0.3.8"
+DEPEND="${RDEPEND}
+	>=sys-devel/gettext-0.11.5
+	>=dev-util/pkgconfig-0.9"
+PDEPEND="oss? ( >=media-plugins/gst-plugins-oss-0.10 )
+	alsa? ( >=media-plugins/gst-plugins-alsa-0.10 )
+	esd? ( >=media-plugins/gst-plugins-esd-0.10 )
+	X? ( >=media-plugins/gst-plugins-x-0.10 )
+	xv? ( >=media-plugins/gst-plugins-xvideo-0.10 )"
+
+DOCS="AUTHORS INSTALL README RELEASE TODO"
+
+src_unpack() {
+	unpack ${A}
+	cd ${S}
+	epatch ${FILESDIR}/${PN}.audioresample.patch
+}
+
+src_compile() {
+	elibtoolize
+
+	# gst doesnt handle optimisations well
+	strip-flags
+	replace-flags "-O3" "-O2"
+	filter-flags "-fprefetch-loop-arrays" # see bug 22249
+	if use alpha || use amd64 || use ia64 || use hppa; then
+		append-flags -fPIC
+	fi
+
+	gst-plugins-base_src_configure
+
+	emake || die
+}
+
+# override eclass
+src_install() {
+	gnome2_src_install
+}