diff options
Diffstat (limited to 'media-libs')
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 +} |