https://bugs.webkit.org/show_bug.cgi?id=116831
Patch by Andre Moreira Magalhaes <andre.magalhaes@collabora.co.uk> on 2013-09-04
Reviewed by Philippe Normand.
Source/WebCore:
Fix issues with rtsp streams embedded on <video> not loading.
* platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp:
Do not reset pipeline to READY state on STATE_CHANGE_ASYNC when entering PAUSED state for
live streams, otherwise we enter an endless loop of READY->PAUSED->READY->PAUSED when
starting playback.
Tools:
Fix timeout issues with rtspsrc/udpsrc gstreamer elements.
* gtk/jhbuild.modules:
* gtk/patches/rtspsrc-timeout-on-udpsrc-is-in-nanoseconds.patch: Added.
* gtk/patches/udpsrc-improve-timeouts.patch: Added.
Changed gstreamer jhbuild modules to use tarball repos (required to add patches)
and added 2 patches to gst-plugins-good to fix timeout issues with rtsp streams.
Both patches are applied upstream and can be removed once a new gstreamer release
is out (up to 1.0.10 the patches are not included) and we bump the requirements.
git-svn-id: https://svn.webkit.org/repository/webkit/trunk@155024
268f45cc-cd09-0410-ab3c-
d52691b4dbfc
+2013-09-04 Andre Moreira Magalhaes <andre.magalhaes@collabora.co.uk>
+
+ [GStreamer] cannot play live streams
+ https://bugs.webkit.org/show_bug.cgi?id=116831
+
+ Reviewed by Philippe Normand.
+
+ Fix issues with rtsp streams embedded on <video> not loading.
+
+ * platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp:
+ Do not reset pipeline to READY state on STATE_CHANGE_ASYNC when entering PAUSED state for
+ live streams, otherwise we enter an endless loop of READY->PAUSED->READY->PAUSED when
+ starting playback.
+
2013-09-03 Chris Fleizach <cfleizach@apple.com>
AX: REGRESSION: @title is exposed as AXDescription when label label from contents already exists.
case GST_STATE_CHANGE_ASYNC:
LOG_MEDIA_MESSAGE("Async: State: %s, pending: %s", gst_element_state_get_name(state), gst_element_state_get_name(pending));
// Change in progress.
-
- // A live stream was paused, reset the pipeline.
- if (state == GST_STATE_PAUSED && pending == GST_STATE_PLAYING && isLiveStream()) {
- changePipelineState(GST_STATE_READY);
- changePipelineState(GST_STATE_PLAYING);
- }
-
break;
case GST_STATE_CHANGE_FAILURE:
LOG_MEDIA_MESSAGE("Failure: State: %s, pending: %s", gst_element_state_get_name(state), gst_element_state_get_name(pending));
+2013-09-04 Andre Moreira Magalhaes <andre.magalhaes@collabora.co.uk>
+
+ [GStreamer] cannot play live streams
+ https://bugs.webkit.org/show_bug.cgi?id=116831
+
+ Reviewed by Philippe Normand.
+
+ Fix timeout issues with rtspsrc/udpsrc gstreamer elements.
+
+ * gtk/jhbuild.modules:
+ * gtk/patches/rtspsrc-timeout-on-udpsrc-is-in-nanoseconds.patch: Added.
+ * gtk/patches/udpsrc-improve-timeouts.patch: Added.
+ Changed gstreamer jhbuild modules to use tarball repos (required to add patches)
+ and added 2 patches to gst-plugins-good to fix timeout issues with rtsp streams.
+ Both patches are applied upstream and can be removed once a new gstreamer release
+ is out (up to 1.0.10 the patches are not included) and we bump the requirements.
+
2013-09-03 Alexey Proskuryakov <ap@apple.com>
[Mac] WebKitTestRunner still beeps sometimes
href="http://xorg.freedesktop.org"/>
<repository type="tarball" name="xmlsoft.org"
href="ftp://xmlsoft.org"/>
- <repository type="git" name="gstreamer"
- href="git://anongit.freedesktop.org/git/gstreamer/"/>
+ <repository type="tarball" name="gstreamer"
+ href="http://gstreamer.freedesktop.org/src/"/>
<repository type="tarball" name="savannah.gnu.org"
href="http://download.savannah.gnu.org/releases/"/>
</autotools>
<autotools id="gstreamer">
- <branch repo="gstreamer" module="gstreamer" checkoutdir="gstreamer" tag="1.0.8"/>
+ <branch module="gstreamer/gstreamer-1.0.8.tar.xz" version="1.0.8"
+ repo="gstreamer"
+ hash="sha256:ff70f45509566b88e35986971ace5e89cb6cb232e9ca249f84502abceef1762d"
+ md5sum="fb0c73470e8dd5dab01e5e9e62db9db9"/>
</autotools>
<autotools id="gst-plugins-base" autogenargs="--disable-examples">
<dependencies>
<dep package="gstreamer"/>
</dependencies>
- <branch repo="gstreamer" module="gst-plugins-base" checkoutdir="gst-plugins-base" tag="1.0.8"/>
+ <branch module="gst-plugins-base/gst-plugins-base-1.0.8.tar.xz" version="1.0.8"
+ repo="gstreamer"
+ hash="sha256:b55c9deea00acf789f82845c088b7c7c17b3ecef24a94831a819071b3dd8ef0d"
+ md5sum="c8b36678186e7d4fb1e6395ffcd1abb1"/>
</autotools>
<autotools id="gst-plugins-good" autogenargs="--disable-examples --disable-soup --disable-gst_v4l2 --without-libv4l2">
<dependencies>
<dep package="gst-plugins-base"/>
</dependencies>
- <branch repo="gstreamer" module="gst-plugins-good" checkoutdir="gst-plugins-good" tag="1.0.8"/>
+
+ <branch module="gst-plugins-good/gst-plugins-good-1.0.8.tar.xz" version="1.0.8"
+ repo="gstreamer"
+ hash="sha256:97831570ccdd8e15557f18858b965e54433d572d27fdabebb8b710cee276cfad"
+ md5sum="4969826043bbd3f76e4a0d49ea475736">
+ <patch file="udpsrc-improve-timeouts.patch" strip="1"/>
+ <patch file="rtspsrc-timeout-on-udpsrc-is-in-nanoseconds.patch" strip="1"/>
+ </branch>
</autotools>
<autotools id="gst-plugins-bad" autogenargs="--disable-examples">
<dependencies>
<dep package="gst-plugins-base"/>
</dependencies>
- <branch repo="gstreamer" module="gst-plugins-bad" checkoutdir="gst-plugins-bad" tag="1.0.8"/>
+ <branch module="gst-plugins-bad/gst-plugins-bad-1.0.8.tar.xz" version="1.0.8"
+ repo="gstreamer"
+ hash="sha256:6949b5532034fc37d5a874e4e3330107767238bc4def9f769b8193124e2420cc"
+ md5sum="a2fdf125ee2ae46047dcbcfc305949ee"/>
</autotools>
<autotools id="gst-libav" autogenargs="--with-libav-extra-configure='--disable-yasm'">
<dependencies>
<dep package="gst-plugins-base"/>
</dependencies>
- <branch repo="gstreamer" module="gst-libav" checkoutdir="gst-libav" tag="1.0.8"/>
+ <branch module="gst-libav/gst-libav-1.0.8.tar.xz" version="1.0.8"
+ repo="gstreamer"
+ hash="sha256:e6e538290e585c993609337761d894dd1b6b53ef625798b2a511d5314579995f"
+ md5sum="7c2e8db039efa1880a61a6982a6909d7"/>
</autotools>
<autotools id="xserver" autogenargs="--disable-xinerama --enable-glx --enable-composite --disable-xorg --disable-dmx --disable-xnest --disable-xquartz --disable-xwin --disable-xephyr --disable-xfake --disable-xfbdev --disable-install-setuid --disable-unit-tests --enable-unix-transport --enable-tcp-transport --enable-local-transport --with-xkb-path=/usr/share/X11/xkb --with-xkb-output=/var/lib/xkb --with-xkb-bin-directory=/usr/bin">
--- /dev/null
+From e0bb1fe30985aa0784825388500cc4fa92c1ff9c Mon Sep 17 00:00:00 2001
+From: Wim Taymans <wim.taymans@collabora.co.uk>
+Date: Wed, 12 Dec 2012 11:09:42 +0100
+Subject: [PATCH 2/2] rtspsrc: timeout on udpsrc is in nanoseconds
+
+---
+ gst/rtsp/gstrtspsrc.c | 4 ++--
+ 1 file changed, 2 insertions(+), 2 deletions(-)
+
+diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c
+index 98577b8..902d8ff 100644
+--- a/gst/rtsp/gstrtspsrc.c
++++ b/gst/rtsp/gstrtspsrc.c
+@@ -2812,8 +2812,8 @@ gst_rtspsrc_stream_configure_udp (GstRTSPSrc * src, GstRTSPStream * stream,
+ /* configure a timeout on the UDP port. When the timeout message is
+ * posted, we assume UDP transport is not possible. We reconnect using TCP
+ * if we can. */
+- g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", src->udp_timeout,
+- NULL);
++ g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout",
++ src->udp_timeout * 1000, NULL);
+
+ /* get output pad of the UDP source. */
+ *outpad = gst_element_get_static_pad (stream->udpsrc[0], "src");
+--
+1.8.1.2
+
--- /dev/null
+From dbbdf54778771535dfea5ddbdeeaba89d9bc7be6 Mon Sep 17 00:00:00 2001
+From: Wim Taymans <wim.taymans@collabora.co.uk>
+Date: Wed, 12 Dec 2012 11:08:13 +0100
+Subject: [PATCH 1/2] udpsrc: improve timeouts
+
+Make it possible to set the timeout after we went to the READY state by using
+the timeout when checking the condition. This also makes it possible to set the
+timeout with a higher granularity than seconds.
+---
+ gst/udp/gstudpsrc.c | 16 ++++++++++------
+ 1 file changed, 10 insertions(+), 6 deletions(-)
+
+diff --git a/gst/udp/gstudpsrc.c b/gst/udp/gstudpsrc.c
+index bdad5b3..5b54021 100644
+--- a/gst/udp/gstudpsrc.c
++++ b/gst/udp/gstudpsrc.c
+@@ -397,13 +397,20 @@ retry:
+ goto no_select;
+
+ do {
++ gint64 timeout;
++
+ try_again = FALSE;
+
++ if (udpsrc->timeout)
++ timeout = udpsrc->timeout / 1000;
++ else
++ timeout = -1;
++
+ GST_LOG_OBJECT (udpsrc, "doing select, timeout %" G_GUINT64_FORMAT,
+- udpsrc->timeout);
++ timeout);
+
+- if (!g_socket_condition_wait (udpsrc->used_socket, G_IO_IN | G_IO_PRI,
+- udpsrc->cancellable, &err)) {
++ if (!g_socket_condition_timed_wait (udpsrc->used_socket, G_IO_IN | G_IO_PRI,
++ timeout, udpsrc->cancellable, &err)) {
+ if (g_error_matches (err, G_IO_ERROR, G_IO_ERROR_BUSY)
+ || g_error_matches (err, G_IO_ERROR, G_IO_ERROR_CANCELLED)) {
+ goto stopped;
+@@ -823,9 +830,6 @@ gst_udpsrc_start (GstBaseSrc * bsrc)
+ goto getsockname_error;
+ }
+
+- if (src->timeout)
+- g_socket_set_timeout (src->used_socket, src->timeout / GST_SECOND);
+-
+ #if GLIB_CHECK_VERSION (2, 35, 7)
+ {
+ gint val = 0;
+--
+1.8.1.2
+
git://git.webkit.org / WebKit-https.git / commitdiff