Imported Upstream version 1.5.0.1+git20150513
diff --git a/INSTALL b/INSTALL
index b42a17a..2099840 100644
--- a/INSTALL
+++ b/INSTALL
@@ -1,48 +1,80 @@
+Installation Instructions
+*************************
+
+Copyright (C) 1994-1996, 1999-2002, 2004-2013 Free Software Foundation,
+Inc.
+
+ Copying and distribution of this file, with or without modification,
+are permitted in any medium without royalty provided the copyright
+notice and this notice are preserved. This file is offered as-is,
+without warranty of any kind.
+
Basic Installation
==================
- These are generic installation instructions.
+ Briefly, the shell command `./configure && make && make install'
+should configure, build, and install this package. The following
+more-detailed instructions are generic; see the `README' file for
+instructions specific to this package. Some packages provide this
+`INSTALL' file but do not implement all of the features documented
+below. The lack of an optional feature in a given package is not
+necessarily a bug. More recommendations for GNU packages can be found
+in *note Makefile Conventions: (standards)Makefile Conventions.
The `configure' shell script attempts to guess correct values for
various system-dependent variables used during compilation. It uses
those values to create a `Makefile' in each directory of the package.
It may also create one or more `.h' files containing system-dependent
definitions. Finally, it creates a shell script `config.status' that
-you can run in the future to recreate the current configuration, a file
-`config.cache' that saves the results of its tests to speed up
-reconfiguring, and a file `config.log' containing compiler output
-(useful mainly for debugging `configure').
+you can run in the future to recreate the current configuration, and a
+file `config.log' containing compiler output (useful mainly for
+debugging `configure').
+
+ It can also use an optional file (typically called `config.cache'
+and enabled with `--cache-file=config.cache' or simply `-C') that saves
+the results of its tests to speed up reconfiguring. Caching is
+disabled by default to prevent problems with accidental use of stale
+cache files.
If you need to do unusual things to compile the package, please try
to figure out how `configure' could check whether to do them, and mail
diffs or instructions to the address given in the `README' so they can
-be considered for the next release. If at some point `config.cache'
-contains results you don't want to keep, you may remove or edit it.
+be considered for the next release. If you are using the cache, and at
+some point `config.cache' contains results you don't want to keep, you
+may remove or edit it.
- The file `configure.in' is used to create `configure' by a program
-called `autoconf'. You only need `configure.in' if you want to change
-it or regenerate `configure' using a newer version of `autoconf'.
+ The file `configure.ac' (or `configure.in') is used to create
+`configure' by a program called `autoconf'. You need `configure.ac' if
+you want to change it or regenerate `configure' using a newer version
+of `autoconf'.
-The simplest way to compile this package is:
+ The simplest way to compile this package is:
1. `cd' to the directory containing the package's source code and type
- `./configure' to configure the package for your system. If you're
- using `csh' on an old version of System V, you might need to type
- `sh ./configure' instead to prevent `csh' from trying to execute
- `configure' itself.
+ `./configure' to configure the package for your system.
- Running `configure' takes awhile. While running, it prints some
- messages telling which features it is checking for.
+ Running `configure' might take a while. While running, it prints
+ some messages telling which features it is checking for.
2. Type `make' to compile the package.
3. Optionally, type `make check' to run any self-tests that come with
- the package.
+ the package, generally using the just-built uninstalled binaries.
4. Type `make install' to install the programs and any data files and
- documentation.
+ documentation. When installing into a prefix owned by root, it is
+ recommended that the package be configured and built as a regular
+ user, and only the `make install' phase executed with root
+ privileges.
- 5. You can remove the program binaries and object files from the
+ 5. Optionally, type `make installcheck' to repeat any self-tests, but
+ this time using the binaries in their final installed location.
+ This target does not install anything. Running this target as a
+ regular user, particularly if the prior `make install' required
+ root privileges, verifies that the installation completed
+ correctly.
+
+ 6. You can remove the program binaries and object files from the
source code directory by typing `make clean'. To also remove the
files that `configure' created (so you can compile the package for
a different kind of computer), type `make distclean'. There is
@@ -51,62 +83,119 @@
all sorts of other programs in order to regenerate files that came
with the distribution.
+ 7. Often, you can also type `make uninstall' to remove the installed
+ files again. In practice, not all packages have tested that
+ uninstallation works correctly, even though it is required by the
+ GNU Coding Standards.
+
+ 8. Some packages, particularly those that use Automake, provide `make
+ distcheck', which can by used by developers to test that all other
+ targets like `make install' and `make uninstall' work correctly.
+ This target is generally not run by end users.
+
Compilers and Options
=====================
Some systems require unusual options for compilation or linking that
-the `configure' script does not know about. You can give `configure'
-initial values for variables by setting them in the environment. Using
-a Bourne-compatible shell, you can do that on the command line like
-this:
- CC=c89 CFLAGS=-O2 LIBS=-lposix ./configure
+the `configure' script does not know about. Run `./configure --help'
+for details on some of the pertinent environment variables.
-Or on systems that have the `env' program, you can do it like this:
- env CPPFLAGS=-I/usr/local/include LDFLAGS=-s ./configure
+ You can give `configure' initial values for configuration parameters
+by setting variables in the command line or in the environment. Here
+is an example:
+
+ ./configure CC=c99 CFLAGS=-g LIBS=-lposix
+
+ *Note Defining Variables::, for more details.
Compiling For Multiple Architectures
====================================
You can compile the package for more than one kind of computer at the
same time, by placing the object files for each architecture in their
-own directory. To do this, you must use a version of `make' that
-supports the `VPATH' variable, such as GNU `make'. `cd' to the
+own directory. To do this, you can use GNU `make'. `cd' to the
directory where you want the object files and executables to go and run
the `configure' script. `configure' automatically checks for the
-source code in the directory that `configure' is in and in `..'.
+source code in the directory that `configure' is in and in `..'. This
+is known as a "VPATH" build.
- If you have to use a `make' that does not supports the `VPATH'
-variable, you have to compile the package for one architecture at a time
-in the source code directory. After you have installed the package for
-one architecture, use `make distclean' before reconfiguring for another
-architecture.
+ With a non-GNU `make', it is safer to compile the package for one
+architecture at a time in the source code directory. After you have
+installed the package for one architecture, use `make distclean' before
+reconfiguring for another architecture.
+
+ On MacOS X 10.5 and later systems, you can create libraries and
+executables that work on multiple system types--known as "fat" or
+"universal" binaries--by specifying multiple `-arch' options to the
+compiler but only a single `-arch' option to the preprocessor. Like
+this:
+
+ ./configure CC="gcc -arch i386 -arch x86_64 -arch ppc -arch ppc64" \
+ CXX="g++ -arch i386 -arch x86_64 -arch ppc -arch ppc64" \
+ CPP="gcc -E" CXXCPP="g++ -E"
+
+ This is not guaranteed to produce working output in all cases, you
+may have to build one architecture at a time and combine the results
+using the `lipo' tool if you have problems.
Installation Names
==================
- By default, `make install' will install the package's files in
-`/usr/local/bin', `/usr/local/man', etc. You can specify an
-installation prefix other than `/usr/local' by giving `configure' the
-option `--prefix=PATH'.
+ By default, `make install' installs the package's commands under
+`/usr/local/bin', include files under `/usr/local/include', etc. You
+can specify an installation prefix other than `/usr/local' by giving
+`configure' the option `--prefix=PREFIX', where PREFIX must be an
+absolute file name.
You can specify separate installation prefixes for
architecture-specific files and architecture-independent files. If you
-give `configure' the option `--exec-prefix=PATH', the package will use
-PATH as the prefix for installing programs and libraries.
-Documentation and other data files will still use the regular prefix.
+pass the option `--exec-prefix=PREFIX' to `configure', the package uses
+PREFIX as the prefix for installing programs and libraries.
+Documentation and other data files still use the regular prefix.
In addition, if you use an unusual directory layout you can give
-options like `--bindir=PATH' to specify different values for particular
+options like `--bindir=DIR' to specify different values for particular
kinds of files. Run `configure --help' for a list of the directories
-you can set and what kinds of files go in them.
+you can set and what kinds of files go in them. In general, the
+default for these options is expressed in terms of `${prefix}', so that
+specifying just `--prefix' will affect all of the other directory
+specifications that were not explicitly provided.
+
+ The most portable way to affect installation locations is to pass the
+correct locations to `configure'; however, many packages provide one or
+both of the following shortcuts of passing variable assignments to the
+`make install' command line to change installation locations without
+having to reconfigure or recompile.
+
+ The first method involves providing an override variable for each
+affected directory. For example, `make install
+prefix=/alternate/directory' will choose an alternate location for all
+directory configuration variables that were expressed in terms of
+`${prefix}'. Any directories that were specified during `configure',
+but not in terms of `${prefix}', must each be overridden at install
+time for the entire installation to be relocated. The approach of
+makefile variable overrides for each directory variable is required by
+the GNU Coding Standards, and ideally causes no recompilation.
+However, some platforms have known limitations with the semantics of
+shared libraries that end up requiring recompilation when using this
+method, particularly noticeable in packages that use GNU Libtool.
+
+ The second method involves providing the `DESTDIR' variable. For
+example, `make install DESTDIR=/alternate/directory' will prepend
+`/alternate/directory' before all installation names. The approach of
+`DESTDIR' overrides is not required by the GNU Coding Standards, and
+does not work on platforms that have drive letters. On the other hand,
+it does better at avoiding recompilation issues, and works well even
+when some directory options were not specified in terms of `${prefix}'
+at `configure' time.
+
+Optional Features
+=================
If the package supports it, you can cause programs to be installed
with an extra prefix or suffix on their names by giving `configure' the
option `--program-prefix=PREFIX' or `--program-suffix=SUFFIX'.
-Optional Features
-=================
-
Some packages pay attention to `--enable-FEATURE' options to
`configure', where FEATURE indicates an optional part of the package.
They may also pay attention to `--with-PACKAGE' options, where PACKAGE
@@ -119,25 +208,80 @@
you can use the `configure' options `--x-includes=DIR' and
`--x-libraries=DIR' to specify their locations.
+ Some packages offer the ability to configure how verbose the
+execution of `make' will be. For these packages, running `./configure
+--enable-silent-rules' sets the default to minimal output, which can be
+overridden with `make V=1'; while running `./configure
+--disable-silent-rules' sets the default to verbose, which can be
+overridden with `make V=0'.
+
+Particular systems
+==================
+
+ On HP-UX, the default C compiler is not ANSI C compatible. If GNU
+CC is not installed, it is recommended to use the following options in
+order to use an ANSI C compiler:
+
+ ./configure CC="cc -Ae -D_XOPEN_SOURCE=500"
+
+and if that doesn't work, install pre-built binaries of GCC for HP-UX.
+
+ HP-UX `make' updates targets which have the same time stamps as
+their prerequisites, which makes it generally unusable when shipped
+generated files such as `configure' are involved. Use GNU `make'
+instead.
+
+ On OSF/1 a.k.a. Tru64, some versions of the default C compiler cannot
+parse its `<wchar.h>' header file. The option `-nodtk' can be used as
+a workaround. If GNU CC is not installed, it is therefore recommended
+to try
+
+ ./configure CC="cc"
+
+and if that doesn't work, try
+
+ ./configure CC="cc -nodtk"
+
+ On Solaris, don't put `/usr/ucb' early in your `PATH'. This
+directory contains several dysfunctional programs; working variants of
+these programs are available in `/usr/bin'. So, if you need `/usr/ucb'
+in your `PATH', put it _after_ `/usr/bin'.
+
+ On Haiku, software installed for all users goes in `/boot/common',
+not `/usr/local'. It is recommended to use the following options:
+
+ ./configure --prefix=/boot/common
+
Specifying the System Type
==========================
- There may be some features `configure' can not figure out
-automatically, but needs to determine by the type of host the package
-will run on. Usually `configure' can figure that out, but if it prints
-a message saying it can not guess the host type, give it the
-`--host=TYPE' option. TYPE can either be a short name for the system
-type, such as `sun4', or a canonical name with three fields:
+ There may be some features `configure' cannot figure out
+automatically, but needs to determine by the type of machine the package
+will run on. Usually, assuming the package is built to be run on the
+_same_ architectures, `configure' can figure that out, but if it prints
+a message saying it cannot guess the machine type, give it the
+`--build=TYPE' option. TYPE can either be a short name for the system
+type, such as `sun4', or a canonical name which has the form:
+
CPU-COMPANY-SYSTEM
-See the file `config.sub' for the possible values of each field. If
-`config.sub' isn't included in this package, then this package doesn't
-need to know the host type.
+where SYSTEM can have one of these forms:
- If you are building compiler tools for cross-compiling, you can also
-use the `--target=TYPE' option to select the type of system they will
-produce code for and the `--build=TYPE' option to select the type of
-system on which you are compiling the package.
+ OS
+ KERNEL-OS
+
+ See the file `config.sub' for the possible values of each field. If
+`config.sub' isn't included in this package, then this package doesn't
+need to know the machine type.
+
+ If you are _building_ compiler tools for cross-compiling, you should
+use the option `--target=TYPE' to select the type of system they will
+produce code for.
+
+ If you want to _use_ a cross compiler, that generates code for a
+platform different from the build platform, you should specify the
+"host" platform (i.e., that on which the generated programs will
+eventually be run) with `--host=TYPE'.
Sharing Defaults
================
@@ -150,19 +294,56 @@
`CONFIG_SITE' environment variable to the location of the site script.
A warning: not all `configure' scripts look for a site script.
-Operation Controls
+Defining Variables
==================
+ Variables not defined in a site shell script can be set in the
+environment passed to `configure'. However, some packages may run
+configure again during the build, and the customized values of these
+variables may be lost. In order to avoid this problem, you should set
+them in the `configure' command line, using `VAR=value'. For example:
+
+ ./configure CC=/usr/local2/bin/gcc
+
+causes the specified `gcc' to be used as the C compiler (unless it is
+overridden in the site shell script).
+
+Unfortunately, this technique does not work for `CONFIG_SHELL' due to
+an Autoconf limitation. Until the limitation is lifted, you can use
+this workaround:
+
+ CONFIG_SHELL=/bin/bash ./configure CONFIG_SHELL=/bin/bash
+
+`configure' Invocation
+======================
+
`configure' recognizes the following options to control how it
operates.
-`--cache-file=FILE'
- Use and save the results of the tests in FILE instead of
- `./config.cache'. Set FILE to `/dev/null' to disable caching, for
- debugging `configure'.
-
`--help'
- Print a summary of the options to `configure', and exit.
+`-h'
+ Print a summary of all of the options to `configure', and exit.
+
+`--help=short'
+`--help=recursive'
+ Print a summary of the options unique to this package's
+ `configure', and exit. The `short' variant lists options used
+ only in the top level, while the `recursive' variant lists options
+ also present in any nested packages.
+
+`--version'
+`-V'
+ Print the version of Autoconf used to generate the `configure'
+ script, and exit.
+
+`--cache-file=FILE'
+ Enable the cache: use and save the results of the tests in FILE,
+ traditionally `config.cache'. FILE defaults to `/dev/null' to
+ disable caching.
+
+`--config-cache'
+`-C'
+ Alias for `--cache-file=config.cache'.
`--quiet'
`--silent'
@@ -175,8 +356,15 @@
Look for the package's source code in directory DIR. Usually
`configure' can determine that directory automatically.
-`--version'
- Print the version of Autoconf used to generate the `configure'
- script, and exit.
+`--prefix=DIR'
+ Use DIR as the installation prefix. *note Installation Names::
+ for more details, including other options available for fine-tuning
+ the installation locations.
-`configure' also accepts some other, not widely useful, options.
+`--no-create'
+`-n'
+ Run the configure checks, but stop before creating any output
+ files.
+
+`configure' also accepts some other, not widely useful, options. Run
+`configure --help' for more details.
diff --git a/autogen.sh b/autogen.sh
index 632b025..01d3fda 100755
--- a/autogen.sh
+++ b/autogen.sh
@@ -57,24 +57,13 @@
printf "+ check for build tools"
if test ! -z "$NOCHECK"; then echo ": skipped version checks"; else echo; fi
-version_check "autoconf" "$AUTOCONF autoconf autoconf270 autoconf269 autoconf268 " \
+version_check "autoreconf" "autoreconf " \
"ftp://ftp.gnu.org/pub/gnu/autoconf/" 2 68 || DIE=1
-version_check "automake" "$AUTOMAKE automake automake-1.11" \
- "ftp://ftp.gnu.org/pub/gnu/automake/" 1 11 || DIE=1
-version_check "autopoint" "autopoint" \
- "ftp://ftp.gnu.org/pub/gnu/gettext/" 0 17 || DIE=1
-version_check "libtoolize" "$LIBTOOLIZE libtoolize glibtoolize" \
- "ftp://ftp.gnu.org/pub/gnu/libtool/" 2 2 6 || DIE=1
version_check "pkg-config" "" \
"http://www.freedesktop.org/software/pkgconfig" 0 8 0 || DIE=1
die_check $DIE
-aclocal_check || DIE=1
-autoheader_check || DIE=1
-
-die_check $DIE
-
# if no arguments specified then this will be printed
if test -z "$*" && test -z "$NOCONFIGURE"; then
echo "+ checking for autogen.sh options"
@@ -88,22 +77,13 @@
# autopoint
if test -d po ; then
- tool_run "$autopoint" "--force"
+ tool_run "autopoint" "--force"
fi
# aclocal
if test -f acinclude.m4; then rm acinclude.m4; fi
-tool_run "$libtoolize" "--copy --force"
-tool_run "$aclocal" "-I m4 -I common/m4 $ACLOCAL_FLAGS"
-tool_run "$autoheader"
-
-# touch the stamp-h.in build stamp so we don't re-run autoheader in maintainer mode
-echo timestamp > stamp-h.in 2> /dev/null
-
-tool_run "$autoconf"
-debug "automake: $automake"
-tool_run "$automake" "--add-missing --copy"
+autoreconf --force --install || exit 1
test -n "$NOCONFIGURE" && {
echo "+ skipping configure stage for package $package, as requested."
diff --git a/common/check.mak b/common/check.mak
index efd5458..8a90b5d 100644
--- a/common/check.mak
+++ b/common/check.mak
@@ -16,38 +16,38 @@
# run any given test by running make test.check
# if the test fails, run it again at at least debug level 2
%.check: %
- @$(TESTS_ENVIRONMENT) \
+ @$(AM_TESTS_ENVIRONMENT) \
CK_DEFAULT_TIMEOUT=20 \
$* || \
- $(TESTS_ENVIRONMENT) \
+ $(AM_TESTS_ENVIRONMENT) \
GST_DEBUG=$$GST_DEBUG,*:2 \
CK_DEFAULT_TIMEOUT=20 \
$*
# just like 'check', but don't run it again if it fails (useful for debugging)
%.check-norepeat: %
- @$(TESTS_ENVIRONMENT) \
+ @$(AM_TESTS_ENVIRONMENT) \
CK_DEFAULT_TIMEOUT=20 \
$*
# run any given test in a loop
%.torture: %
@for i in `seq 1 $(LOOPS)`; do \
- $(TESTS_ENVIRONMENT) \
+ $(AM_TESTS_ENVIRONMENT) \
CK_DEFAULT_TIMEOUT=20 \
$*; done
# run any given test in an infinite loop
%.forever: %
@while true; do \
- $(TESTS_ENVIRONMENT) \
+ $(AM_TESTS_ENVIRONMENT) \
CK_DEFAULT_TIMEOUT=20 \
$* || break; done
# valgrind any given test by running make test.valgrind
%.valgrind: %
@valgrind_log=$(subst /,-,$*-valgrind.log); \
- $(TESTS_ENVIRONMENT) \
+ $(AM_TESTS_ENVIRONMENT) \
CK_DEFAULT_TIMEOUT=360 \
G_SLICE=always-malloc \
$(LIBTOOL) --mode=execute \
@@ -65,7 +65,7 @@
# valgrind any given test and generate suppressions for it
%.valgrind.gen-suppressions: %
- @$(TESTS_ENVIRONMENT) \
+ @$(AM_TESTS_ENVIRONMENT) \
CK_DEFAULT_TIMEOUT=360 \
G_SLICE=always-malloc \
$(LIBTOOL) --mode=execute \
@@ -95,7 +95,7 @@
# gdb any given test by running make test.gdb
%.gdb: %
- @$(TESTS_ENVIRONMENT) \
+ @$(AM_TESTS_ENVIRONMENT) \
CK_FORK=no \
$(LIBTOOL) --mode=execute \
gdb $*
@@ -205,7 +205,7 @@
GST_INSPECT = $(GST_TOOLS_DIR)/gst-inspect-$(GST_API_VERSION)
inspect:
@echo "Inspecting features ..."
- @for e in `$(TESTS_ENVIRONMENT) $(GST_INSPECT) | head -n -2 \
+ @for e in `$(AM_TESTS_ENVIRONMENT) $(GST_INSPECT) | head -n -2 \
| cut -d: -f2`; \
do echo Inspecting $$e; \
$(GST_INSPECT) $$e > /dev/null 2>&1; done
diff --git a/common/gst-autogen.sh b/common/gst-autogen.sh
index 78cab66..c5798d0 100644
--- a/common/gst-autogen.sh
+++ b/common/gst-autogen.sh
@@ -169,54 +169,6 @@
return 1;
}
-aclocal_check ()
-{
- # normally aclocal is part of automake
- # so we expect it to be in the same place as automake
- # so if a different automake is supplied, we need to adapt as well
- # so how's about replacing automake with aclocal in the set var,
- # and saving that in $aclocal ?
- # note, this will fail if the actual automake isn't called automake*
- # or if part of the path before it contains it
- if [ -z "$automake" ]; then
- echo "Error: no automake variable set !"
- return 1
- else
- aclocal=`echo $automake | sed s/automake/aclocal/`
- debug "aclocal: $aclocal"
- if [ "$aclocal" != "aclocal" ];
- then
- CONFIGURE_DEF_OPT="$CONFIGURE_DEF_OPT --with-aclocal=$aclocal"
- fi
- if [ ! -x `which $aclocal` ]; then
- echo "Error: cannot execute $aclocal !"
- return 1
- fi
- fi
-}
-
-autoheader_check ()
-{
- # same here - autoheader is part of autoconf
- # use the same voodoo
- if [ -z "$autoconf" ]; then
- echo "Error: no autoconf variable set !"
- return 1
- else
- autoheader=`echo $autoconf | sed s/autoconf/autoheader/`
- debug "autoheader: $autoheader"
- if [ "$autoheader" != "autoheader" ];
- then
- CONFIGURE_DEF_OPT="$CONFIGURE_DEF_OPT --with-autoheader=$autoheader"
- fi
- if [ ! -x `which $autoheader` ]; then
- echo "Error: cannot execute $autoheader !"
- return 1
- fi
- fi
-
-}
-
die_check ()
{
# call with $DIE
diff --git a/common/gst.supp b/common/gst.supp
index c08f660..5e2aaf9 100644
--- a/common/gst.supp
+++ b/common/gst.supp
@@ -3972,3 +3972,37 @@
timer_create(evp)
fun:timer_create@@GLIBC_2.3.3
}
+
+{
+ closures aren't valgrind friendly (bgo#739850)
+ Memcheck:Leak
+ match-leak-kinds: possible
+ fun:calloc
+ ...
+ fun:g_cclosure_new
+}
+
+{
+ closures aren't valgrind friendly (bgo#739850)
+ Memcheck:Leak
+ match-leak-kinds: possible
+ fun:malloc
+ ...
+ fun:g_closure_add_invalidate_notifier
+}
+
+{
+ closures aren't valgrind friendly (bgo#739850)
+ Memcheck:Leak
+ match-leak-kinds: possible
+ fun:calloc
+ ...
+ fun:g_closure_new_simple
+}
+
+{
+ glib/giomodules2 (from libsoup.supp)
+ Memcheck:Leak
+ ...
+ fun:_g_io_module_get_default
+}
diff --git a/configure.ac b/configure.ac
index 9a286c2..9763274 100644
--- a/configure.ac
+++ b/configure.ac
@@ -1,4 +1,4 @@
-AC_PREREQ([2.68])
+AC_PREREQ([2.69])
dnl please read gstreamer/docs/random/autotools before changing this file
@@ -10,7 +10,7 @@
AG_GST_INIT
dnl initialize automake
-AM_INIT_AUTOMAKE([-Wno-portability 1.11 no-dist-gzip dist-xz tar-ustar subdir-objects])
+AM_INIT_AUTOMAKE([-Wno-portability 1.14 no-dist-gzip dist-xz tar-ustar subdir-objects])
dnl define PACKAGE_VERSION_* variables
AS_VERSION
diff --git a/docs/plugins/gst-plugins-good-plugins-sections.txt b/docs/plugins/gst-plugins-good-plugins-sections.txt
index ac6b4aa..7a9dcc8 100644
--- a/docs/plugins/gst-plugins-good-plugins-sections.txt
+++ b/docs/plugins/gst-plugins-good-plugins-sections.txt
@@ -904,6 +904,8 @@
<FILE>element-rtpbin</FILE>
<TITLE>rtpbin</TITLE>
GstRtpBin
+RTPJitterBufferMode
+GstRTCPSync
<SUBSECTION Standard>
GstRtpBinPrivate
GstRtpBinClass
diff --git a/docs/plugins/html/gst-plugins-good-plugins-1.0.devhelp2 b/docs/plugins/html/gst-plugins-good-plugins-1.0.devhelp2
index d18a636..a2adc67 100644
--- a/docs/plugins/html/gst-plugins-good-plugins-1.0.devhelp2
+++ b/docs/plugins/html/gst-plugins-good-plugins-1.0.devhelp2
@@ -775,9 +775,9 @@
<keyword type="" name="Element Information" link="gst-plugins-good-plugins-mulawenc.html#id-1.2.82.7.2.1.1"/>
<keyword type="" name="Element Pads" link="gst-plugins-good-plugins-mulawenc.html#id-1.2.82.7.2.1.2"/>
<keyword type="struct" name="struct GstMuLawEnc" link="gst-plugins-good-plugins-mulawenc.html#GstMuLawEnc-struct"/>
- <keyword type="" name="Example launch line" link="gst-plugins-good-plugins-multifilesink.html#id-1.2.83.7.9"/>
- <keyword type="" name="Element Information" link="gst-plugins-good-plugins-multifilesink.html#id-1.2.83.7.10.1"/>
- <keyword type="" name="Element Pads" link="gst-plugins-good-plugins-multifilesink.html#id-1.2.83.7.10.2"/>
+ <keyword type="" name="Example launch line" link="gst-plugins-good-plugins-multifilesink.html#id-1.2.83.7.11"/>
+ <keyword type="" name="Element Information" link="gst-plugins-good-plugins-multifilesink.html#id-1.2.83.7.12.1"/>
+ <keyword type="" name="Element Pads" link="gst-plugins-good-plugins-multifilesink.html#id-1.2.83.7.12.2"/>
<keyword type="struct" name="struct GstMultiFileSink" link="gst-plugins-good-plugins-multifilesink.html#GstMultiFileSink-struct"/>
<keyword type="enum" name="enum GstMultiFileSinkNext" link="gst-plugins-good-plugins-multifilesink.html#GstMultiFileSinkNext"/>
<keyword type="property" name="The “location” property" link="gst-plugins-good-plugins-multifilesink.html#GstMultiFileSink--location"/>
@@ -1099,6 +1099,8 @@
<keyword type="" name="Element Information" link="gst-plugins-good-plugins-rtpbin.html#id-1.2.124.9.13.1"/>
<keyword type="" name="Element Pads" link="gst-plugins-good-plugins-rtpbin.html#id-1.2.124.9.13.2"/>
<keyword type="struct" name="struct GstRtpBin" link="gst-plugins-good-plugins-rtpbin.html#GstRtpBin-struct"/>
+ <keyword type="enum" name="enum RTPJitterBufferMode" link="gst-plugins-good-plugins-rtpbin.html#RTPJitterBufferMode"/>
+ <keyword type="enum" name="enum GstRTCPSync" link="gst-plugins-good-plugins-rtpbin.html#GstRTCPSync"/>
<keyword type="property" name="The “do-lost” property" link="gst-plugins-good-plugins-rtpbin.html#GstRtpBin--do-lost"/>
<keyword type="property" name="The “latency” property" link="gst-plugins-good-plugins-rtpbin.html#GstRtpBin--latency"/>
<keyword type="property" name="The “sdes” property" link="gst-plugins-good-plugins-rtpbin.html#GstRtpBin--sdes"/>
diff --git a/docs/plugins/html/gst-plugins-good-plugins-amrparse.html b/docs/plugins/html/gst-plugins-good-plugins-amrparse.html
index 27bf175..49821dc 100644
--- a/docs/plugins/html/gst-plugins-good-plugins-amrparse.html
+++ b/docs/plugins/html/gst-plugins-good-plugins-amrparse.html
@@ -188,7 +188,7 @@
</div>
<div class="refsect1">
<a name="gst-plugins-good-plugins-amrparse.see-also"></a><h2>See Also</h2>
-<p><span class="type">GstAmrnbDec</span>, <span class="type">GstAmrnbEnc</span></p>
+<p><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-ugly/html/gst-plugins-ugly-plugins-amrnbdec.html#GstAmrnbDec"><span class="type">GstAmrnbDec</span></a>, <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-ugly/html/gst-plugins-ugly-plugins-amrnbenc.html#GstAmrnbEnc"><span class="type">GstAmrnbEnc</span></a></p>
</div>
</div>
<div class="footer">
diff --git a/docs/plugins/html/gst-plugins-good-plugins-audioamplify.html b/docs/plugins/html/gst-plugins-good-plugins-audioamplify.html
index e192734..705e4d9 100644
--- a/docs/plugins/html/gst-plugins-good-plugins-audioamplify.html
+++ b/docs/plugins/html/gst-plugins-good-plugins-audioamplify.html
@@ -93,8 +93,8 @@
2
3</pre></td>
<td class="listing_code"><pre class="programlisting">gst<span class="gtkdoc opt">-</span>launch<span class="gtkdoc opt">-</span><span class="number">1.0</span> audiotestsrc wave<span class="gtkdoc opt">=</span>saw <span class="gtkdoc opt">!</span> audioamplify amplification<span class="gtkdoc opt">=</span><span class="number">1.5</span> <span class="gtkdoc opt">!</span> alsasink
-gst<span class="gtkdoc opt">-</span>launch<span class="gtkdoc opt">-</span><span class="number">1.0</span> filesrc location<span class="gtkdoc opt">=</span><span class="string">"melo1.ogg"</span> <span class="gtkdoc opt">!</span> oggdemux <span class="gtkdoc opt">!</span> vorbisdec <span class="gtkdoc opt">!</span> audioconvert <span class="gtkdoc opt">!</span> audioamplify amplification<span class="gtkdoc opt">=</span><span class="number">1.5</span> method<span class="gtkdoc opt">=</span>wrap<span class="gtkdoc opt">-</span>negative <span class="gtkdoc opt">!</span> alsasink
-gst<span class="gtkdoc opt">-</span>launch<span class="gtkdoc opt">-</span><span class="number">1.0</span> audiotestsrc wave<span class="gtkdoc opt">=</span>saw <span class="gtkdoc opt">!</span> audioconvert <span class="gtkdoc opt">!</span> audioamplify amplification<span class="gtkdoc opt">=</span><span class="number">1.5</span> method<span class="gtkdoc opt">=</span>wrap<span class="gtkdoc opt">-</span>positive <span class="gtkdoc opt">!</span> audioconvert <span class="gtkdoc opt">!</span> alsasink</pre></td>
+gst<span class="gtkdoc opt">-</span>launch<span class="gtkdoc opt">-</span><span class="number">1.0</span> filesrc location<span class="gtkdoc opt">=</span><span class="string">"melo1.ogg"</span> <span class="gtkdoc opt">!</span> oggdemux <span class="gtkdoc opt">!</span> vorbisdec <span class="gtkdoc opt">!</span> audioconvert <span class="gtkdoc opt">!</span> audioamplify amplification<span class="gtkdoc opt">=</span><span class="number">1.5</span> clipping<span class="gtkdoc opt">-</span>method<span class="gtkdoc opt">=</span>wrap<span class="gtkdoc opt">-</span>negative <span class="gtkdoc opt">!</span> alsasink
+gst<span class="gtkdoc opt">-</span>launch<span class="gtkdoc opt">-</span><span class="number">1.0</span> audiotestsrc wave<span class="gtkdoc opt">=</span>saw <span class="gtkdoc opt">!</span> audioconvert <span class="gtkdoc opt">!</span> audioamplify amplification<span class="gtkdoc opt">=</span><span class="number">1.5</span> clipping<span class="gtkdoc opt">-</span>method<span class="gtkdoc opt">=</span>wrap<span class="gtkdoc opt">-</span>positive <span class="gtkdoc opt">!</span> audioconvert <span class="gtkdoc opt">!</span> alsasink</pre></td>
</tr>
</tbody>
</table>
diff --git a/docs/plugins/html/gst-plugins-good-plugins-audiodynamic.html b/docs/plugins/html/gst-plugins-good-plugins-audiodynamic.html
index 82de048..1f62bdb 100644
--- a/docs/plugins/html/gst-plugins-good-plugins-audiodynamic.html
+++ b/docs/plugins/html/gst-plugins-good-plugins-audiodynamic.html
@@ -104,8 +104,8 @@
<td class="listing_lines" align="right"><pre>1
2
3</pre></td>
- <td class="listing_code"><pre class="programlisting">gst<span class="gtkdoc opt">-</span>launch<span class="gtkdoc opt">-</span><span class="number">1.0</span> audiotestsrc wave<span class="gtkdoc opt">=</span>saw <span class="gtkdoc opt">!</span> audiodynamic characteristics<span class="gtkdoc opt">=</span>soft<span class="gtkdoc opt">-</span>knee mode<span class="gtkdoc opt">=</span>compressor threshold<span class="gtkdoc opt">=</span><span class="number">0.5</span> rate<span class="gtkdoc opt">=</span><span class="number">0.5</span> <span class="gtkdoc opt">!</span> alsasink
-gst<span class="gtkdoc opt">-</span>launch<span class="gtkdoc opt">-</span><span class="number">1.0</span> filesrc location<span class="gtkdoc opt">=</span><span class="string">"melo1.ogg"</span> <span class="gtkdoc opt">!</span> oggdemux <span class="gtkdoc opt">!</span> vorbisdec <span class="gtkdoc opt">!</span> audioconvert <span class="gtkdoc opt">!</span> audiodynamic characteristics<span class="gtkdoc opt">=</span>hard<span class="gtkdoc opt">-</span>knee mode<span class="gtkdoc opt">=</span>expander threshold<span class="gtkdoc opt">=</span><span class="number">0.2</span> rate<span class="gtkdoc opt">=</span><span class="number">4.0</span> <span class="gtkdoc opt">!</span> alsasink
+ <td class="listing_code"><pre class="programlisting">gst<span class="gtkdoc opt">-</span>launch<span class="gtkdoc opt">-</span><span class="number">1.0</span> audiotestsrc wave<span class="gtkdoc opt">=</span>saw <span class="gtkdoc opt">!</span> audiodynamic characteristics<span class="gtkdoc opt">=</span>soft<span class="gtkdoc opt">-</span>knee mode<span class="gtkdoc opt">=</span>compressor threshold<span class="gtkdoc opt">=</span><span class="number">0.5</span> ratio<span class="gtkdoc opt">=</span><span class="number">0.5</span> <span class="gtkdoc opt">!</span> alsasink
+gst<span class="gtkdoc opt">-</span>launch<span class="gtkdoc opt">-</span><span class="number">1.0</span> filesrc location<span class="gtkdoc opt">=</span><span class="string">"melo1.ogg"</span> <span class="gtkdoc opt">!</span> oggdemux <span class="gtkdoc opt">!</span> vorbisdec <span class="gtkdoc opt">!</span> audioconvert <span class="gtkdoc opt">!</span> audiodynamic characteristics<span class="gtkdoc opt">=</span>hard<span class="gtkdoc opt">-</span>knee mode<span class="gtkdoc opt">=</span>expander threshold<span class="gtkdoc opt">=</span><span class="number">0.2</span> ratio<span class="gtkdoc opt">=</span><span class="number">4.0</span> <span class="gtkdoc opt">!</span> alsasink
gst<span class="gtkdoc opt">-</span>launch<span class="gtkdoc opt">-</span><span class="number">1.0</span> audiotestsrc wave<span class="gtkdoc opt">=</span>saw <span class="gtkdoc opt">!</span> audioconvert <span class="gtkdoc opt">!</span> audiodynamic <span class="gtkdoc opt">!</span> audioconvert <span class="gtkdoc opt">!</span> alsasink</pre></td>
</tr>
</tbody>
diff --git a/docs/plugins/html/gst-plugins-good-plugins-multifilesink.html b/docs/plugins/html/gst-plugins-good-plugins-multifilesink.html
index 146f3d8..2a80d57 100644
--- a/docs/plugins/html/gst-plugins-good-plugins-multifilesink.html
+++ b/docs/plugins/html/gst-plugins-good-plugins-multifilesink.html
@@ -108,6 +108,12 @@
<div class="refsect1">
<a name="gst-plugins-good-plugins-multifilesink.description"></a><h2>Description</h2>
<p>Write incoming data to a series of sequentially-named files.</p>
+<p>This element is usually used with data where each buffer is an
+independent unit of data in its own right (e.g. raw video buffers or
+encoded JPEG or PNG images) or with streamable container formats such
+as MPEG-TS or MPEG-PS.</p>
+<p>It is not possible to use this element to create independently playable
+mp4 files, use the splitmuxsink element for that instead.</p>
<p>The filename property should contain a string with a %d placeholder that will
be substituted with the index for each filename.</p>
<p>If the <a class="link" href="gst-plugins-good-plugins-multifilesink.html#GstMultiFileSink--post-messages" title="The “post-messages” property"><span class="type">“post-messages”</span></a> property is <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><span class="type">TRUE</span></a>, it sends an application
@@ -158,7 +164,7 @@
</p></li>
</ul></div>
<div class="refsect2">
-<a name="id-1.2.83.7.9"></a><h3>Example launch line</h3>
+<a name="id-1.2.83.7.11"></a><h3>Example launch line</h3>
<div class="informalexample">
<table class="listing_frame" border="0" cellpadding="0" cellspacing="0">
<tbody>
@@ -176,7 +182,7 @@
<div class="refsynopsisdiv">
<h2>Synopsis</h2>
<div class="refsect2">
-<a name="id-1.2.83.7.10.1"></a><h3>Element Information</h3>
+<a name="id-1.2.83.7.12.1"></a><h3>Element Information</h3>
<div class="variablelist"><table border="0" class="variablelist">
<colgroup>
<col align="left" valign="top">
@@ -202,7 +208,7 @@
</div>
<hr>
<div class="refsect2">
-<a name="id-1.2.83.7.10.2"></a><h3>Element Pads</h3>
+<a name="id-1.2.83.7.12.2"></a><h3>Element Pads</h3>
<div class="variablelist"><table border="0" class="variablelist">
<colgroup>
<col align="left" valign="top">
diff --git a/docs/plugins/html/gst-plugins-good-plugins-qtdemux.html b/docs/plugins/html/gst-plugins-good-plugins-qtdemux.html
index 397c9ff..9005630 100644
--- a/docs/plugins/html/gst-plugins-good-plugins-qtdemux.html
+++ b/docs/plugins/html/gst-plugins-good-plugins-qtdemux.html
@@ -82,7 +82,7 @@
<tbody>
<tr>
<td class="listing_lines" align="right"><pre>1</pre></td>
- <td class="listing_code"><pre class="programlisting">gst<span class="gtkdoc opt">-</span>launch<span class="gtkdoc opt">-</span><span class="number">1.0</span> filesrc location<span class="gtkdoc opt">=</span>test<span class="gtkdoc opt">.</span>mov <span class="gtkdoc opt">!</span> qtdemux name<span class="gtkdoc opt">=</span>demux demux<span class="gtkdoc opt">.</span>audio_0 <span class="gtkdoc opt">!</span> decodebin <span class="gtkdoc opt">!</span> audioconvert <span class="gtkdoc opt">!</span> audioresample <span class="gtkdoc opt">!</span> autoaudiosink demux<span class="gtkdoc opt">.</span>video_0 <span class="gtkdoc opt">!</span> queue <span class="gtkdoc opt">!</span> decodebin <span class="gtkdoc opt">!</span> videoconvert <span class="gtkdoc opt">!</span> videoscale <span class="gtkdoc opt">!</span> autovideosink</pre></td>
+ <td class="listing_code"><pre class="programlisting">gst<span class="gtkdoc opt">-</span>launch<span class="gtkdoc opt">-</span><span class="number">1.0</span> filesrc location<span class="gtkdoc opt">=</span>test<span class="gtkdoc opt">.</span>mov <span class="gtkdoc opt">!</span> qtdemux name<span class="gtkdoc opt">=</span>demux demux<span class="gtkdoc opt">.</span>audio_0 <span class="gtkdoc opt">!</span> queue <span class="gtkdoc opt">!</span> decodebin <span class="gtkdoc opt">!</span> audioconvert <span class="gtkdoc opt">!</span> audioresample <span class="gtkdoc opt">!</span> autoaudiosink demux<span class="gtkdoc opt">.</span>video_0 <span class="gtkdoc opt">!</span> queue <span class="gtkdoc opt">!</span> decodebin <span class="gtkdoc opt">!</span> videoconvert <span class="gtkdoc opt">!</span> videoscale <span class="gtkdoc opt">!</span> autovideosink</pre></td>
</tr>
</tbody>
</table>
diff --git a/docs/plugins/html/gst-plugins-good-plugins-rtpL16depay.html b/docs/plugins/html/gst-plugins-good-plugins-rtpL16depay.html
index 045a39a..d245e97 100644
--- a/docs/plugins/html/gst-plugins-good-plugins-rtpL16depay.html
+++ b/docs/plugins/html/gst-plugins-good-plugins-rtpL16depay.html
@@ -66,7 +66,7 @@
<tbody>
<tr>
<td class="listing_lines" align="right"><pre>1</pre></td>
- <td class="listing_code"><pre class="programlisting">gst<span class="gtkdoc opt">-</span>launch udpsrc caps<span class="gtkdoc opt">=</span><span class="string">'application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1, payload=(int)96'</span> <span class="gtkdoc opt">!</span> rtpL16depay <span class="gtkdoc opt">!</span> pulsesink</pre></td>
+ <td class="listing_code"><pre class="programlisting">gst<span class="gtkdoc opt">-</span>launch<span class="gtkdoc opt">-</span><span class="number">1.0</span> udpsrc caps<span class="gtkdoc opt">=</span><span class="string">'application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1, payload=(int)96'</span> <span class="gtkdoc opt">!</span> rtpL16depay <span class="gtkdoc opt">!</span> pulsesink</pre></td>
</tr>
</tbody>
</table>
diff --git a/docs/plugins/html/gst-plugins-good-plugins-rtpL16pay.html b/docs/plugins/html/gst-plugins-good-plugins-rtpL16pay.html
index b4f1988..7e29fe0 100644
--- a/docs/plugins/html/gst-plugins-good-plugins-rtpL16pay.html
+++ b/docs/plugins/html/gst-plugins-good-plugins-rtpL16pay.html
@@ -67,7 +67,7 @@
<tbody>
<tr>
<td class="listing_lines" align="right"><pre>1</pre></td>
- <td class="listing_code"><pre class="programlisting">gst<span class="gtkdoc opt">-</span>launch <span class="gtkdoc opt">-</span>v audiotestsrc <span class="gtkdoc opt">!</span> audioconvert <span class="gtkdoc opt">!</span> rtpL16pay <span class="gtkdoc opt">!</span> udpsink</pre></td>
+ <td class="listing_code"><pre class="programlisting">gst<span class="gtkdoc opt">-</span>launch<span class="gtkdoc opt">-</span><span class="number">1.0</span> <span class="gtkdoc opt">-</span>v audiotestsrc <span class="gtkdoc opt">!</span> audioconvert <span class="gtkdoc opt">!</span> rtpL16pay <span class="gtkdoc opt">!</span> udpsink</pre></td>
</tr>
</tbody>
</table>
diff --git a/docs/plugins/html/gst-plugins-good-plugins-rtpac3pay.html b/docs/plugins/html/gst-plugins-good-plugins-rtpac3pay.html
index 6f73f90..5523da8 100644
--- a/docs/plugins/html/gst-plugins-good-plugins-rtpac3pay.html
+++ b/docs/plugins/html/gst-plugins-good-plugins-rtpac3pay.html
@@ -66,7 +66,7 @@
<tbody>
<tr>
<td class="listing_lines" align="right"><pre>1</pre></td>
- <td class="listing_code"><pre class="programlisting">gst<span class="gtkdoc opt">-</span>launch <span class="gtkdoc opt">-</span>v audiotestsrc <span class="gtkdoc opt">!</span> avenc_ac3 <span class="gtkdoc opt">!</span> rtpac3pay <span class="gtkdoc opt">!</span> udpsink</pre></td>
+ <td class="listing_code"><pre class="programlisting">gst<span class="gtkdoc opt">-</span>launch<span class="gtkdoc opt">-</span><span class="number">1.0</span> <span class="gtkdoc opt">-</span>v audiotestsrc <span class="gtkdoc opt">!</span> avenc_ac3 <span class="gtkdoc opt">!</span> rtpac3pay <span class="gtkdoc opt">!</span> udpsink</pre></td>
</tr>
</tbody>
</table>
diff --git a/docs/plugins/html/gst-plugins-good-plugins-rtpamrpay.html b/docs/plugins/html/gst-plugins-good-plugins-rtpamrpay.html
index 2a05f36..d5d4d53 100644
--- a/docs/plugins/html/gst-plugins-good-plugins-rtpamrpay.html
+++ b/docs/plugins/html/gst-plugins-good-plugins-rtpamrpay.html
@@ -66,7 +66,7 @@
<tbody>
<tr>
<td class="listing_lines" align="right"><pre>1</pre></td>
- <td class="listing_code"><pre class="programlisting">gst<span class="gtkdoc opt">-</span>launch <span class="gtkdoc opt">-</span>v audiotestsrc <span class="gtkdoc opt">!</span> amrnbenc <span class="gtkdoc opt">!</span> rtpamrpay <span class="gtkdoc opt">!</span> udpsink</pre></td>
+ <td class="listing_code"><pre class="programlisting">gst<span class="gtkdoc opt">-</span>launch<span class="gtkdoc opt">-</span><span class="number">1.0</span> <span class="gtkdoc opt">-</span>v audiotestsrc <span class="gtkdoc opt">!</span> amrnbenc <span class="gtkdoc opt">!</span> rtpamrpay <span class="gtkdoc opt">!</span> udpsink</pre></td>
</tr>
</tbody>
</table>
diff --git a/docs/plugins/html/gst-plugins-good-plugins-rtpbin.html b/docs/plugins/html/gst-plugins-good-plugins-rtpbin.html
index 95eb549..8edb983 100644
--- a/docs/plugins/html/gst-plugins-good-plugins-rtpbin.html
+++ b/docs/plugins/html/gst-plugins-good-plugins-rtpbin.html
@@ -71,7 +71,7 @@
<td class="property_flags">Read / Write</td>
</tr>
<tr>
-<td class="property_type"><span class="type">RTPJitterBufferMode</span></td>
+<td class="property_type"><a class="link" href="gst-plugins-good-plugins-rtpbin.html#RTPJitterBufferMode" title="enum RTPJitterBufferMode"><span class="type">RTPJitterBufferMode</span></a></td>
<td class="property_name"><a class="link" href="gst-plugins-good-plugins-rtpbin.html#GstRtpBin--buffer-mode" title="The “buffer-mode” property">buffer-mode</a></td>
<td class="property_flags">Read / Write</td>
</tr>
@@ -86,7 +86,7 @@
<td class="property_flags">Read / Write</td>
</tr>
<tr>
-<td class="property_type"><span class="type">GstRTCPSync</span></td>
+<td class="property_type"><a class="link" href="gst-plugins-good-plugins-rtpbin.html#GstRTCPSync" title="enum GstRTCPSync"><span class="type">GstRTCPSync</span></a></td>
<td class="property_name"><a class="link" href="gst-plugins-good-plugins-rtpbin.html#GstRtpBin--rtcp-sync" title="The “rtcp-sync” property">rtcp-sync</a></td>
<td class="property_flags">Read / Write</td>
</tr>
@@ -250,10 +250,20 @@
<col width="150px" class="name">
<col class="description">
</colgroup>
-<tbody><tr>
+<tbody>
+<tr>
<td class="datatype_keyword">struct</td>
<td class="function_name"><a class="link" href="gst-plugins-good-plugins-rtpbin.html#GstRtpBin-struct" title="struct GstRtpBin">GstRtpBin</a></td>
-</tr></tbody>
+</tr>
+<tr>
+<td class="datatype_keyword">enum</td>
+<td class="function_name"><a class="link" href="gst-plugins-good-plugins-rtpbin.html#RTPJitterBufferMode" title="enum RTPJitterBufferMode">RTPJitterBufferMode</a></td>
+</tr>
+<tr>
+<td class="datatype_keyword">enum</td>
+<td class="function_name"><a class="link" href="gst-plugins-good-plugins-rtpbin.html#GstRTCPSync" title="enum GstRTCPSync">GstRTCPSync</a></td>
+</tr>
+</tbody>
</table></div>
</div>
<div class="refsect1">
@@ -624,6 +634,94 @@
<p>
</p>
</div>
+<hr>
+<div class="refsect2">
+<a name="RTPJitterBufferMode"></a><h3>enum RTPJitterBufferMode</h3>
+<p>RTP_JITTER_BUFFER_MODE_NONE: don't do any skew correction, outgoing
+ timestamps are calculated directly from the RTP timestamps. This mode is
+ good for recording but not for real-time applications.
+RTP_JITTER_BUFFER_MODE_SLAVE: calculate the skew between sender and receiver
+ and produce smoothed adjusted outgoing timestamps. This mode is good for
+ low latency communications.
+RTP_JITTER_BUFFER_MODE_BUFFER: buffer packets between low/high watermarks.
+ This mode is good for streaming communication.
+RTP_JITTER_BUFFER_MODE_SYNCED: sender and receiver clocks are synchronized,
+ like <a class="link" href="gst-plugins-good-plugins-rtpbin.html#RTP-JITTER-BUFFER-MODE-SLAVE:CAPS"><span class="type">RTP_JITTER_BUFFER_MODE_SLAVE</span></a> but skew is assumed to be 0. Good for
+ low latency communication when sender and receiver clocks are
+ synchronized and there is thus no clock skew.
+RTP_JITTER_BUFFER_MODE_LAST: last buffer mode.</p>
+<p>The different buffer modes for a jitterbuffer.</p>
+<div class="refsect3">
+<a name="id-1.2.124.11.3.5"></a><h4>Members</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="300px" class="enum_members_name">
+<col class="enum_members_description">
+<col width="200px" class="enum_members_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="enum_member_name"><p><a name="RTP-JITTER-BUFFER-MODE-NONE:CAPS"></a>RTP_JITTER_BUFFER_MODE_NONE</p></td>
+<td class="enum_member_description"> </td>
+<td class="enum_member_annotations"> </td>
+</tr>
+<tr>
+<td class="enum_member_name"><p><a name="RTP-JITTER-BUFFER-MODE-SLAVE:CAPS"></a>RTP_JITTER_BUFFER_MODE_SLAVE</p></td>
+<td class="enum_member_description"> </td>
+<td class="enum_member_annotations"> </td>
+</tr>
+<tr>
+<td class="enum_member_name"><p><a name="RTP-JITTER-BUFFER-MODE-BUFFER:CAPS"></a>RTP_JITTER_BUFFER_MODE_BUFFER</p></td>
+<td class="enum_member_description"> </td>
+<td class="enum_member_annotations"> </td>
+</tr>
+<tr>
+<td class="enum_member_name"><p><a name="RTP-JITTER-BUFFER-MODE-SYNCED:CAPS"></a>RTP_JITTER_BUFFER_MODE_SYNCED</p></td>
+<td class="enum_member_description"> </td>
+<td class="enum_member_annotations"> </td>
+</tr>
+<tr>
+<td class="enum_member_name"><p><a name="RTP-JITTER-BUFFER-MODE-LAST:CAPS"></a>RTP_JITTER_BUFFER_MODE_LAST</p></td>
+<td class="enum_member_description"> </td>
+<td class="enum_member_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTCPSync"></a><h3>enum GstRTCPSync</h3>
+<p>
+</p>
+<div class="refsect3">
+<a name="id-1.2.124.11.4.4"></a><h4>Members</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="300px" class="enum_members_name">
+<col class="enum_members_description">
+<col width="200px" class="enum_members_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="enum_member_name"><p><a name="GST-RTP-BIN-RTCP-SYNC-ALWAYS:CAPS"></a>GST_RTP_BIN_RTCP_SYNC_ALWAYS</p></td>
+<td class="enum_member_description"> </td>
+<td class="enum_member_annotations"> </td>
+</tr>
+<tr>
+<td class="enum_member_name"><p><a name="GST-RTP-BIN-RTCP-SYNC-INITIAL:CAPS"></a>GST_RTP_BIN_RTCP_SYNC_INITIAL</p></td>
+<td class="enum_member_description"> </td>
+<td class="enum_member_annotations"> </td>
+</tr>
+<tr>
+<td class="enum_member_name"><p><a name="GST-RTP-BIN-RTCP-SYNC-RTP:CAPS"></a>GST_RTP_BIN_RTCP_SYNC_RTP</p></td>
+<td class="enum_member_description"> </td>
+<td class="enum_member_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
</div>
<div class="refsect1">
<a name="gst-plugins-good-plugins-rtpbin.property-details"></a><h2>Property Details</h2>
@@ -668,7 +766,7 @@
<hr>
<div class="refsect2">
<a name="GstRtpBin--buffer-mode"></a><h3>The <code class="literal">“buffer-mode”</code> property</h3>
-<pre class="programlisting"> “buffer-mode” <span class="type">RTPJitterBufferMode</span></pre>
+<pre class="programlisting"> “buffer-mode” <a class="link" href="gst-plugins-good-plugins-rtpbin.html#RTPJitterBufferMode" title="enum RTPJitterBufferMode"><span class="type">RTPJitterBufferMode</span></a></pre>
<p>Control the buffering and timestamping mode used by the jitterbuffer.</p>
<p>Flags: Read / Write</p>
<p>Default value: Slave receiver to sender clock</p>
@@ -696,7 +794,7 @@
<hr>
<div class="refsect2">
<a name="GstRtpBin--rtcp-sync"></a><h3>The <code class="literal">“rtcp-sync”</code> property</h3>
-<pre class="programlisting"> “rtcp-sync” <span class="type">GstRTCPSync</span></pre>
+<pre class="programlisting"> “rtcp-sync” <a class="link" href="gst-plugins-good-plugins-rtpbin.html#GstRTCPSync" title="enum GstRTCPSync"><span class="type">GstRTCPSync</span></a></pre>
<p>If not synchronizing (directly) to the NTP clock, determines how to sync
the various streams.</p>
<p>Flags: Read / Write</p>
diff --git a/docs/plugins/html/gst-plugins-good-plugins-rtpjitterbuffer.html b/docs/plugins/html/gst-plugins-good-plugins-rtpjitterbuffer.html
index b7ea686..6f336ca 100644
--- a/docs/plugins/html/gst-plugins-good-plugins-rtpjitterbuffer.html
+++ b/docs/plugins/html/gst-plugins-good-plugins-rtpjitterbuffer.html
@@ -64,7 +64,7 @@
<td class="property_flags">Read / Write</td>
</tr>
<tr>
-<td class="property_type"><span class="type">RTPJitterBufferMode</span></td>
+<td class="property_type"><a class="link" href="gst-plugins-good-plugins-rtpbin.html#RTPJitterBufferMode" title="enum RTPJitterBufferMode"><span class="type">RTPJitterBufferMode</span></a></td>
<td class="property_name"><a class="link" href="gst-plugins-good-plugins-rtpjitterbuffer.html#GstRtpJitterBuffer--mode" title="The “mode” property">mode</a></td>
<td class="property_flags">Read / Write</td>
</tr>
@@ -400,7 +400,7 @@
<hr>
<div class="refsect2">
<a name="GstRtpJitterBuffer--mode"></a><h3>The <code class="literal">“mode”</code> property</h3>
-<pre class="programlisting"> “mode” <span class="type">RTPJitterBufferMode</span></pre>
+<pre class="programlisting"> “mode” <a class="link" href="gst-plugins-good-plugins-rtpbin.html#RTPJitterBufferMode" title="enum RTPJitterBufferMode"><span class="type">RTPJitterBufferMode</span></a></pre>
<p>Control the buffering and timestamping mode used by the jitterbuffer.</p>
<p>Flags: Read / Write</p>
<p>Default value: Slave receiver to sender clock</p>
@@ -476,10 +476,28 @@
<pre class="programlisting"> “stats” <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html"><span class="type">GstStructure</span></a> *</pre>
<p>Various jitterbuffer statistics. This property returns a GstStructure
with name application/x-rtp-jitterbuffer-stats with the following fields:</p>
-<p> "rtx-count" G_TYPE_UINT64 The number of retransmissions requested
- "rtx-success-count" G_TYPE_UINT64 The number of successful retransmissions
- "rtx-per-packet" G_TYPE_DOUBLE Average number of RTX per packet
- "rtx-rtt" G_TYPE_UINT64 Average round trip time per RTX</p>
+<div class="itemizedlist"><ul class="itemizedlist" style="list-style-type: disc; ">
+<li class="listitem"><p>
+ <span class="type">guint64</span>
+ <code class="classname">"rtx-count"</code>:
+ the number of retransmissions requested.
+ </p></li>
+<li class="listitem"><p>
+ <span class="type">guint64</span>
+ <code class="classname">"rtx-success-count"</code>:
+ the number of successful retransmissions.
+ </p></li>
+<li class="listitem"><p>
+ <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gdouble"><span class="type">gdouble</span></a>
+ <code class="classname">"rtx-per-packet"</code>:
+ average number of RTX per packet.
+ </p></li>
+<li class="listitem"><p>
+ <span class="type">guint64</span>
+ <code class="classname">"rtx-rtt"</code>:
+ average round trip time per RTX.
+ </p></li>
+</ul></div>
<p>Flags: Read</p>
<p class="since">Since 1.4</p>
</div>
diff --git a/docs/plugins/html/gst-plugins-good-plugins-rtpmux.html b/docs/plugins/html/gst-plugins-good-plugins-rtpmux.html
index 0ad6964..3842034 100644
--- a/docs/plugins/html/gst-plugins-good-plugins-rtpmux.html
+++ b/docs/plugins/html/gst-plugins-good-plugins-rtpmux.html
@@ -104,7 +104,7 @@
4
5
6</pre></td>
- <td class="listing_code"><pre class="programlisting">gst<span class="gtkdoc opt">-</span>launch rtpmux name<span class="gtkdoc opt">=</span>mux <span class="gtkdoc opt">!</span> udpsink host<span class="gtkdoc opt">=</span><span class="number">127.0.0.1</span> port<span class="gtkdoc opt">=</span><span class="number">8888</span> \
+ <td class="listing_code"><pre class="programlisting">gst<span class="gtkdoc opt">-</span>launch<span class="gtkdoc opt">-</span><span class="number">1.0</span> rtpmux name<span class="gtkdoc opt">=</span>mux <span class="gtkdoc opt">!</span> udpsink host<span class="gtkdoc opt">=</span><span class="number">127.0.0.1</span> port<span class="gtkdoc opt">=</span><span class="number">8888</span> \
alsasrc <span class="gtkdoc opt">!</span> alawenc <span class="gtkdoc opt">!</span> rtppcmapay <span class="gtkdoc opt">!</span> \
application<span class="gtkdoc opt">/</span>x<span class="gtkdoc opt">-</span>rtp<span class="gtkdoc opt">,</span> payload<span class="gtkdoc opt">=</span><span class="number">8</span><span class="gtkdoc opt">,</span> rate<span class="gtkdoc opt">=</span><span class="number">8000</span> <span class="gtkdoc opt">!</span> mux<span class="gtkdoc opt">.</span>sink_0 \
audiotestsrc is<span class="gtkdoc opt">-</span>live<span class="gtkdoc opt">=</span><span class="number">1</span> <span class="gtkdoc opt">!</span> \
diff --git a/docs/plugins/html/index.sgml b/docs/plugins/html/index.sgml
index 6f09c4a..153f822 100644
--- a/docs/plugins/html/index.sgml
+++ b/docs/plugins/html/index.sgml
@@ -1626,6 +1626,8 @@
<ANCHOR id="gst-plugins-good-plugins-rtpbin.functions_details" href="gst-plugins-good-plugins-1.0/gst-plugins-good-plugins-rtpbin.html#gst-plugins-good-plugins-rtpbin.functions_details">
<ANCHOR id="gst-plugins-good-plugins-rtpbin.other_details" href="gst-plugins-good-plugins-1.0/gst-plugins-good-plugins-rtpbin.html#gst-plugins-good-plugins-rtpbin.other_details">
<ANCHOR id="GstRtpBin-struct" href="gst-plugins-good-plugins-1.0/gst-plugins-good-plugins-rtpbin.html#GstRtpBin-struct">
+<ANCHOR id="RTPJitterBufferMode" href="gst-plugins-good-plugins-1.0/gst-plugins-good-plugins-rtpbin.html#RTPJitterBufferMode">
+<ANCHOR id="GstRTCPSync" href="gst-plugins-good-plugins-1.0/gst-plugins-good-plugins-rtpbin.html#GstRTCPSync">
<ANCHOR id="gst-plugins-good-plugins-rtpbin.property-details" href="gst-plugins-good-plugins-1.0/gst-plugins-good-plugins-rtpbin.html#gst-plugins-good-plugins-rtpbin.property-details">
<ANCHOR id="GstRtpBin--do-lost" href="gst-plugins-good-plugins-1.0/gst-plugins-good-plugins-rtpbin.html#GstRtpBin--do-lost">
<ANCHOR id="GstRtpBin--latency" href="gst-plugins-good-plugins-1.0/gst-plugins-good-plugins-rtpbin.html#GstRtpBin--latency">
diff --git a/ext/dv/gstdvdemux.c b/ext/dv/gstdvdemux.c
index 352283a..be99647 100644
--- a/ext/dv/gstdvdemux.c
+++ b/ext/dv/gstdvdemux.c
@@ -303,6 +303,8 @@
GstPad *pad;
GstEvent *event;
gchar *stream_id;
+ gchar rec_datetime[40];
+ GstDateTime *rec_dt;
pad = gst_pad_new_from_static_template (template, template->name_template);
@@ -341,6 +343,16 @@
tags = gst_tag_list_new (GST_TAG_CONTAINER_FORMAT, "DV", NULL);
gst_tag_list_set_scope (tags, GST_TAG_SCOPE_GLOBAL);
+
+ if (dv_get_recording_datetime (dvdemux->decoder, rec_datetime)) {
+ rec_dt = gst_date_time_new_from_iso8601_string (rec_datetime);
+ if (rec_dt) {
+ gst_tag_list_add (tags, GST_TAG_MERGE_REPLACE, GST_TAG_DATE_TIME,
+ rec_dt, NULL);
+ gst_date_time_unref (rec_dt);
+ }
+ }
+
if (dvdemux->videosrcpad)
gst_pad_push_event (dvdemux->videosrcpad,
gst_event_new_tag (gst_tag_list_ref (tags)));
diff --git a/ext/jpeg/gstjpegdec.c b/ext/jpeg/gstjpegdec.c
index 25dfb00..a1ba7c0 100644
--- a/ext/jpeg/gstjpegdec.c
+++ b/ext/jpeg/gstjpegdec.c
@@ -994,6 +994,7 @@
guint code, hdr_ok;
gboolean need_unmap = TRUE;
GstVideoCodecState *state = NULL;
+ gboolean release_frame = TRUE;
dec->current_frame = frame;
gst_buffer_map (frame->input_buffer, &dec->current_frame_map, GST_MAP_READ);
@@ -1141,6 +1142,7 @@
gst_buffer_unmap (frame->input_buffer, &dec->current_frame_map);
ret = gst_video_decoder_finish_frame (bdec, frame);
+ release_frame = FALSE;
need_unmap = FALSE;
done:
@@ -1150,6 +1152,9 @@
if (need_unmap)
gst_buffer_unmap (frame->input_buffer, &dec->current_frame_map);
+ if (release_frame)
+ gst_video_decoder_release_frame (bdec, frame);
+
if (state)
gst_video_codec_state_unref (state);
@@ -1183,6 +1188,7 @@
gst_buffer_unmap (frame->input_buffer, &dec->current_frame_map);
gst_video_decoder_drop_frame (bdec, frame);
+ release_frame = FALSE;
need_unmap = FALSE;
jpeg_abort_decompress (&dec->cinfo);
diff --git a/ext/shout2/gstshout2.c b/ext/shout2/gstshout2.c
index 613bc7a..2384ebc 100644
--- a/ext/shout2/gstshout2.c
+++ b/ext/shout2/gstshout2.c
@@ -27,7 +27,7 @@
* <refsect2>
* <title>Example launch line</title>
* |[
- * gst-launch uridecodebin uri=file:///path/to/audiofile ! audioconvert ! vorbisenc ! oggmux ! shout2send mount=/stream.ogg port=8000 username=source password=somepassword ip=server_IP_address_or_hostname
+ * gst-launch-1.0 uridecodebin uri=file:///path/to/audiofile ! audioconvert ! vorbisenc ! oggmux ! shout2send mount=/stream.ogg port=8000 username=source password=somepassword ip=server_IP_address_or_hostname
* ]| This pipeline demuxes, decodes, re-encodes and re-muxes an audio
* media file into oggvorbis and sends the resulting stream to an Icecast
* server. Properties mount, port, username and password are all server-config
diff --git a/ext/vpx/gstvp8dec.c b/ext/vpx/gstvp8dec.c
index 2951118..4b05f99 100644
--- a/ext/vpx/gstvp8dec.c
+++ b/ext/vpx/gstvp8dec.c
@@ -32,7 +32,7 @@
* <refsect2>
* <title>Example pipeline</title>
* |[
- * gst-launch -v filesrc location=videotestsrc.webm ! matroskademux ! vp8dec ! xvimagesink
+ * gst-launch-1.0 -v filesrc location=videotestsrc.webm ! matroskademux ! vp8dec ! videoconvert ! videoscale ! autovideosink
* ]| This example pipeline will decode a WebM stream and decodes the VP8 video.
* </refsect2>
*/
@@ -390,12 +390,18 @@
deststride = GST_VIDEO_FRAME_COMP_STRIDE (&frame, comp);
srcstride = img->stride[comp];
- /* FIXME (Edward) : Do a plane memcpy is srcstride == deststride instead
- * of copying line by line */
- for (line = 0; line < height; line++) {
- memcpy (dest, src, width);
- dest += deststride;
- src += srcstride;
+ if (srcstride == deststride) {
+ GST_TRACE_OBJECT (dec, "Stride matches. Comp %d: %d, copying full plane",
+ comp, srcstride);
+ memcpy (dest, src, srcstride * height);
+ } else {
+ GST_TRACE_OBJECT (dec, "Stride mismatch. Comp %d: %d != %d, copying "
+ "line by line.", comp, srcstride, deststride);
+ for (line = 0; line < height; line++) {
+ memcpy (dest, src, width);
+ dest += deststride;
+ src += srcstride;
+ }
}
}
diff --git a/ext/vpx/gstvp8enc.c b/ext/vpx/gstvp8enc.c
index 9d38de3..364bfa3 100644
--- a/ext/vpx/gstvp8enc.c
+++ b/ext/vpx/gstvp8enc.c
@@ -40,7 +40,7 @@
* <refsect2>
* <title>Example pipeline</title>
* |[
- * gst-launch -v videotestsrc num-buffers=1000 ! vp8enc ! webmmux ! filesink location=videotestsrc.webm
+ * gst-launch-1.0 -v videotestsrc num-buffers=1000 ! vp8enc ! webmmux ! filesink location=videotestsrc.webm
* ]| This example pipeline will encode a test video source to VP8 muxed in an
* WebM container.
* </refsect2>
@@ -385,7 +385,10 @@
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
- GST_STATIC_CAPS (GST_VIDEO_CAPS_MAKE ("I420"))
+ GST_STATIC_CAPS ("video/x-raw, "
+ "format = (string) \"I420\", "
+ "width = (int) [1, 16383], "
+ "height = (int) [1, 16383], framerate = (fraction) [ 0/1, MAX ]")
);
static GstStaticPadTemplate gst_vp8_enc_src_template =
@@ -1537,22 +1540,12 @@
GST_DEBUG_OBJECT (video_encoder, "Using timebase configuration");
encoder->cfg.g_timebase.num = encoder->timebase_n;
encoder->cfg.g_timebase.den = encoder->timebase_d;
- } else if (GST_VIDEO_INFO_FPS_D (info) != 0
- && GST_VIDEO_INFO_FPS_N (info) != 0) {
- /* GstVideoInfo holds either the framerate or max-framerate (if framerate
- * is 0) in FPS so this will be used if max-framerate or framerate
- * is set */
- GST_DEBUG_OBJECT (video_encoder, "Setting timebase from framerate");
- encoder->cfg.g_timebase.num = GST_VIDEO_INFO_FPS_D (info);
- encoder->cfg.g_timebase.den = GST_VIDEO_INFO_FPS_N (info);
} else {
/* Zero framerate and max-framerate but still need to setup the timebase to avoid
* a divide by zero error. Presuming the lowest common denominator will be RTP -
* VP8 payload draft states clock rate of 90000 which should work for anyone where
* FPS < 90000 (shouldn't be too many cases where it's higher) though wouldn't be optimal. RTP specification
* http://tools.ietf.org/html/draft-ietf-payload-vp8-01 section 6.3.1 */
- GST_WARNING_OBJECT (encoder,
- "No timebase and zero framerate setting timebase to 1/90000");
encoder->cfg.g_timebase.num = 1;
encoder->cfg.g_timebase.den = 90000;
}
@@ -1893,8 +1886,8 @@
pts =
gst_util_uint64_scale (encoder->last_pts,
- encoder->cfg.g_timebase.num * (GstClockTime) GST_SECOND,
- encoder->cfg.g_timebase.den);
+ encoder->cfg.g_timebase.den,
+ encoder->cfg.g_timebase.num * (GstClockTime) GST_SECOND);
status = vpx_codec_encode (&encoder->encoder, NULL, pts, 0, flags, deadline);
g_mutex_unlock (&encoder->encoder_lock);
@@ -2018,15 +2011,14 @@
g_mutex_lock (&encoder->encoder_lock);
pts =
gst_util_uint64_scale (frame->pts,
- encoder->cfg.g_timebase.num * (GstClockTime) GST_SECOND,
- encoder->cfg.g_timebase.den);
+ encoder->cfg.g_timebase.den,
+ encoder->cfg.g_timebase.num * (GstClockTime) GST_SECOND);
encoder->last_pts = frame->pts;
if (frame->duration != GST_CLOCK_TIME_NONE) {
duration =
- gst_util_uint64_scale (frame->duration,
- encoder->cfg.g_timebase.num * (GstClockTime) GST_SECOND,
- encoder->cfg.g_timebase.den);
+ gst_util_uint64_scale (frame->duration, encoder->cfg.g_timebase.den,
+ encoder->cfg.g_timebase.num * (GstClockTime) GST_SECOND);
encoder->last_pts += frame->duration;
} else {
duration = 1;
diff --git a/ext/vpx/gstvp9dec.c b/ext/vpx/gstvp9dec.c
index 5a18f3a..a11c848 100644
--- a/ext/vpx/gstvp9dec.c
+++ b/ext/vpx/gstvp9dec.c
@@ -32,7 +32,7 @@
* <refsect2>
* <title>Example pipeline</title>
* |[
- * gst-launch -v filesrc location=videotestsrc.webm ! matroskademux ! vp9dec ! xvimagesink
+ * gst-launch-1.0 -v filesrc location=videotestsrc.webm ! matroskademux ! vp9dec ! videoconvert ! videoscale ! autovideosink
* ]| This example pipeline will decode a WebM stream and decodes the VP9 video.
* </refsect2>
*/
@@ -387,12 +387,18 @@
deststride = GST_VIDEO_FRAME_COMP_STRIDE (&frame, comp);
srcstride = img->stride[comp];
- /* FIXME (Edward) : Do a plane memcpy is srcstride == deststride instead
- * of copying line by line */
- for (line = 0; line < height; line++) {
- memcpy (dest, src, width);
- dest += deststride;
- src += srcstride;
+ if (srcstride == deststride) {
+ GST_TRACE_OBJECT (dec, "Stride matches. Comp %d: %d, copying full plane",
+ comp, srcstride);
+ memcpy (dest, src, srcstride * height);
+ } else {
+ GST_TRACE_OBJECT (dec, "Stride mismatch. Comp %d: %d != %d, copying "
+ "line by line.", comp, srcstride, deststride);
+ for (line = 0; line < height; line++) {
+ memcpy (dest, src, width);
+ dest += deststride;
+ src += srcstride;
+ }
}
}
diff --git a/ext/vpx/gstvp9enc.c b/ext/vpx/gstvp9enc.c
index e64f5af..4f272b4 100644
--- a/ext/vpx/gstvp9enc.c
+++ b/ext/vpx/gstvp9enc.c
@@ -40,7 +40,7 @@
* <refsect2>
* <title>Example pipeline</title>
* |[
- * gst-launch -v videotestsrc num-buffers=1000 ! vp9enc ! webmmux ! filesink location=videotestsrc.webm
+ * gst-launch-1.0 -v videotestsrc num-buffers=1000 ! vp9enc ! webmmux ! filesink location=videotestsrc.webm
* ]| This example pipeline will encode a test video source to VP9 muxed in an
* WebM container.
* </refsect2>
@@ -1513,22 +1513,12 @@
GST_DEBUG_OBJECT (video_encoder, "Using timebase configuration");
encoder->cfg.g_timebase.num = encoder->timebase_n;
encoder->cfg.g_timebase.den = encoder->timebase_d;
- } else if (GST_VIDEO_INFO_FPS_D (info) != 0
- && GST_VIDEO_INFO_FPS_N (info) != 0) {
- /* GstVideoInfo holds either the framerate or max-framerate (if framerate
- * is 0) in FPS so this will be used if max-framerate or framerate
- * is set */
- GST_DEBUG_OBJECT (video_encoder, "Setting timebase from framerate");
- encoder->cfg.g_timebase.num = GST_VIDEO_INFO_FPS_D (info);
- encoder->cfg.g_timebase.den = GST_VIDEO_INFO_FPS_N (info);
} else {
/* Zero framerate and max-framerate but still need to setup the timebase to avoid
* a divide by zero error. Presuming the lowest common denominator will be RTP -
* VP9 payload draft states clock rate of 90000 which should work for anyone where
* FPS < 90000 (shouldn't be too many cases where it's higher) though wouldn't be optimal. RTP specification
* http://tools.ietf.org/html/draft-ietf-payload-vp9-01 section 6.3.1 */
- GST_WARNING_OBJECT (encoder,
- "No timebase and zero framerate setting timebase to 1/90000");
encoder->cfg.g_timebase.num = 1;
encoder->cfg.g_timebase.den = 90000;
}
@@ -1832,8 +1822,8 @@
pts =
gst_util_uint64_scale (encoder->last_pts,
- encoder->cfg.g_timebase.num * (GstClockTime) GST_SECOND,
- encoder->cfg.g_timebase.den);
+ encoder->cfg.g_timebase.den,
+ encoder->cfg.g_timebase.num * (GstClockTime) GST_SECOND);
status = vpx_codec_encode (&encoder->encoder, NULL, pts, 0, flags, deadline);
g_mutex_unlock (&encoder->encoder_lock);
@@ -1951,15 +1941,14 @@
g_mutex_lock (&encoder->encoder_lock);
pts =
gst_util_uint64_scale (frame->pts,
- encoder->cfg.g_timebase.num * (GstClockTime) GST_SECOND,
- encoder->cfg.g_timebase.den);
+ encoder->cfg.g_timebase.den,
+ encoder->cfg.g_timebase.num * (GstClockTime) GST_SECOND);
encoder->last_pts = frame->pts;
if (frame->duration != GST_CLOCK_TIME_NONE) {
duration =
- gst_util_uint64_scale (frame->duration,
- encoder->cfg.g_timebase.num * (GstClockTime) GST_SECOND,
- encoder->cfg.g_timebase.den);
+ gst_util_uint64_scale (frame->duration, encoder->cfg.g_timebase.den,
+ encoder->cfg.g_timebase.num * (GstClockTime) GST_SECOND);
encoder->last_pts += frame->duration;
} else {
duration = 1;
diff --git a/gst-plugins-good.spec b/gst-plugins-good.spec
index 632fbb0..acb204e 100644
--- a/gst-plugins-good.spec
+++ b/gst-plugins-good.spec
@@ -5,7 +5,7 @@
Name: %{gstreamer}-plugins-good
Version: 1.5.0.1
-Release: 0.20150316.185508.gst
+Release: 0.20150513.125648.gst
Summary: GStreamer plug-ins with good code and licensing
Group: Applications/Multimedia
diff --git a/gst/alpha/Makefile.am b/gst/alpha/Makefile.am
index ee6495e..abadffe 100644
--- a/gst/alpha/Makefile.am
+++ b/gst/alpha/Makefile.am
@@ -17,17 +17,3 @@
libgstalphacolor_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
noinst_HEADERS = gstalpha.h gstalphacolor.h
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstalpha -:SHARED libgstalpha \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstalpha_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstalpha_la_CFLAGS) \
- -:LDFLAGS $(libgstalpha_la_LDFLAGS) \
- $(libgstalpha_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/alpha/Makefile.in b/gst/alpha/Makefile.in
index 06af5b0..247ea97 100644
--- a/gst/alpha/Makefile.in
+++ b/gst/alpha/Makefile.in
@@ -916,20 +916,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstalpha -:SHARED libgstalpha \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstalpha_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstalpha_la_CFLAGS) \
- -:LDFLAGS $(libgstalpha_la_LDFLAGS) \
- $(libgstalpha_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/alpha/gstalpha.c b/gst/alpha/gstalpha.c
index 9566f30..736f377 100644
--- a/gst/alpha/gstalpha.c
+++ b/gst/alpha/gstalpha.c
@@ -120,8 +120,7 @@
PROP_NOISE_LEVEL,
PROP_BLACK_SENSITIVITY,
PROP_WHITE_SENSITIVITY,
- PROP_PREFER_PASSTHROUGH,
- PROP_LAST
+ PROP_PREFER_PASSTHROUGH
};
static GstStaticPadTemplate gst_alpha_src_template =
diff --git a/gst/apetag/Makefile.am b/gst/apetag/Makefile.am
index f0984af..bb06771 100644
--- a/gst/apetag/Makefile.am
+++ b/gst/apetag/Makefile.am
@@ -14,17 +14,3 @@
libgstapetag_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
noinst_HEADERS = gstapedemux.h
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstapetag -:SHARED libgstapetag \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstapetag_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstapetag_la_CFLAGS) \
- -:LDFLAGS $(libgstapetag_la_LDFLAGS) \
- $(libgstapetag_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/apetag/Makefile.in b/gst/apetag/Makefile.in
index 3e2b294..3039913 100644
--- a/gst/apetag/Makefile.in
+++ b/gst/apetag/Makefile.in
@@ -891,20 +891,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstapetag -:SHARED libgstapetag \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstapetag_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstapetag_la_CFLAGS) \
- -:LDFLAGS $(libgstapetag_la_LDFLAGS) \
- $(libgstapetag_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/audiofx/Makefile.am b/gst/audiofx/Makefile.am
index 3ca249c..a3dd174 100644
--- a/gst/audiofx/Makefile.am
+++ b/gst/audiofx/Makefile.am
@@ -59,17 +59,3 @@
audioecho.h \
gstscaletempo.h \
math_compat.h
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstaudiofx -:SHARED libgstaudiofx \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstaudiofx_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstaudiofx_la_CFLAGS) \
- -:LDFLAGS $(libgstaudiofx_la_LDFLAGS) \
- $(libgstaudiofx_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/audiofx/Makefile.in b/gst/audiofx/Makefile.in
index fb45652..56b6a2e 100644
--- a/gst/audiofx/Makefile.in
+++ b/gst/audiofx/Makefile.in
@@ -1161,20 +1161,6 @@
cp -p tmp-orc.c $(distdir)/$(ORC_SOURCE)-dist.c
cp -p $(ORC_SOURCE).h $(distdir)/$(ORC_SOURCE)-dist.h
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstaudiofx -:SHARED libgstaudiofx \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstaudiofx_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstaudiofx_la_CFLAGS) \
- -:LDFLAGS $(libgstaudiofx_la_LDFLAGS) \
- $(libgstaudiofx_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/audiofx/audioamplify.c b/gst/audiofx/audioamplify.c
index ad5f3c5..726ce0e 100644
--- a/gst/audiofx/audioamplify.c
+++ b/gst/audiofx/audioamplify.c
@@ -29,8 +29,8 @@
* <title>Example launch line</title>
* |[
* gst-launch-1.0 audiotestsrc wave=saw ! audioamplify amplification=1.5 ! alsasink
- * gst-launch-1.0 filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audioamplify amplification=1.5 method=wrap-negative ! alsasink
- * gst-launch-1.0 audiotestsrc wave=saw ! audioconvert ! audioamplify amplification=1.5 method=wrap-positive ! audioconvert ! alsasink
+ * gst-launch-1.0 filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audioamplify amplification=1.5 clipping-method=wrap-negative ! alsasink
+ * gst-launch-1.0 audiotestsrc wave=saw ! audioconvert ! audioamplify amplification=1.5 clipping-method=wrap-positive ! audioconvert ! alsasink
* ]|
* </refsect2>
*/
diff --git a/gst/audiofx/audiodynamic.c b/gst/audiofx/audiodynamic.c
index a695191..9b3a62c 100644
--- a/gst/audiofx/audiodynamic.c
+++ b/gst/audiofx/audiodynamic.c
@@ -29,8 +29,8 @@
* <refsect2>
* <title>Example launch line</title>
* |[
- * gst-launch-1.0 audiotestsrc wave=saw ! audiodynamic characteristics=soft-knee mode=compressor threshold=0.5 rate=0.5 ! alsasink
- * gst-launch-1.0 filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiodynamic characteristics=hard-knee mode=expander threshold=0.2 rate=4.0 ! alsasink
+ * gst-launch-1.0 audiotestsrc wave=saw ! audiodynamic characteristics=soft-knee mode=compressor threshold=0.5 ratio=0.5 ! alsasink
+ * gst-launch-1.0 filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiodynamic characteristics=hard-knee mode=expander threshold=0.2 ratio=4.0 ! alsasink
* gst-launch-1.0 audiotestsrc wave=saw ! audioconvert ! audiodynamic ! audioconvert ! alsasink
* ]|
* </refsect2>
diff --git a/gst/audiofx/audiofxbasefirfilter.c b/gst/audiofx/audiofxbasefirfilter.c
index 6af55f7..ccafb0a 100644
--- a/gst/audiofx/audiofxbasefirfilter.c
+++ b/gst/audiofx/audiofxbasefirfilter.c
@@ -116,6 +116,7 @@
self->buffer = buffer = g_new0 (gdouble, self->buffer_length); \
} \
\
+ input_samples *= channels; \
/* convolution */ \
for (i = 0; i < input_samples; i++) { \
dst[i] = 0.0; \
@@ -155,7 +156,7 @@
if (self->buffer_fill > kernel_length) \
self->buffer_fill = kernel_length; \
\
- return input_samples; \
+ return input_samples / channels; \
} G_STMT_END
DEFINE_PROCESS_FUNC (32, float);
diff --git a/gst/audiofx/audiokaraoke.c b/gst/audiofx/audiokaraoke.c
index 9d5d575..76697ca 100644
--- a/gst/audiofx/audiokaraoke.c
+++ b/gst/audiofx/audiokaraoke.c
@@ -66,8 +66,7 @@
PROP_LEVEL,
PROP_MONO_LEVEL,
PROP_FILTER_BAND,
- PROP_FILTER_WIDTH,
- PROP_LAST
+ PROP_FILTER_WIDTH
};
#define ALLOWED_CAPS \
diff --git a/gst/audiofx/audiopanoramaorc-dist.c b/gst/audiofx/audiopanoramaorc-dist.c
index 12d6aa0..3d5999a 100644
--- a/gst/audiofx/audiopanoramaorc-dist.c
+++ b/gst/audiofx/audiopanoramaorc-dist.c
@@ -264,7 +264,7 @@
static const orc_uint8 bc[] = {
1, 9, 39, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 115,
- 49,
+ 49,
54, 95, 99, 104, 49, 95, 110, 111, 110, 101, 11, 4, 4, 12, 2, 2,
195, 0, 4, 4, 2, 0,
};
@@ -390,7 +390,7 @@
static const orc_uint8 bc[] = {
1, 9, 39, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 102,
- 51,
+ 51,
50, 95, 99, 104, 49, 95, 110, 111, 110, 101, 11, 8, 8, 12, 4, 4,
194, 0, 4, 4, 2, 0,
};
@@ -502,7 +502,7 @@
static const orc_uint8 bc[] = {
1, 9, 39, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 115,
- 49,
+ 49,
54, 95, 99, 104, 50, 95, 110, 111, 110, 101, 11, 4, 4, 12, 4, 4,
21, 1, 79, 0, 4, 2, 0,
};
@@ -614,7 +614,7 @@
static const orc_uint8 bc[] = {
1, 9, 39, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 102,
- 51,
+ 51,
50, 95, 99, 104, 50, 95, 110, 111, 110, 101, 11, 8, 8, 12, 8, 8,
21, 1, 112, 0, 4, 2, 0,
};
@@ -842,7 +842,7 @@
static const orc_uint8 bc[] = {
1, 9, 38, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 115,
- 49,
+ 49,
54, 95, 99, 104, 49, 95, 112, 115, 121, 11, 4, 4, 12, 2, 2, 17,
4, 17, 4, 20, 8, 20, 4, 20, 4, 153, 33, 4, 211, 33, 33, 202,
34, 33, 25, 202, 33, 33, 24, 194, 32, 33, 34, 21, 1, 210, 32, 32,
@@ -1053,7 +1053,7 @@
static const orc_uint8 bc[] = {
1, 9, 38, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 102,
- 51,
+ 51,
50, 95, 99, 104, 49, 95, 112, 115, 121, 11, 8, 8, 12, 4, 4, 17,
4, 17, 4, 20, 4, 20, 4, 202, 33, 4, 25, 202, 32, 4, 24, 194,
0, 32, 33, 2, 0,
@@ -1355,7 +1355,7 @@
static const orc_uint8 bc[] = {
1, 9, 44, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 115,
- 49,
+ 49,
54, 95, 99, 104, 50, 95, 112, 115, 121, 95, 114, 105, 103, 104, 116, 11,
4, 4, 12, 4, 4, 17, 4, 17, 4, 20, 8, 20, 4, 20, 4, 20,
4, 21, 1, 153, 32, 4, 21, 1, 211, 32, 32, 192, 33, 32, 193, 34,
@@ -1675,7 +1675,7 @@
static const orc_uint8 bc[] = {
1, 9, 43, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 115,
- 49,
+ 49,
54, 95, 99, 104, 50, 95, 112, 115, 121, 95, 108, 101, 102, 116, 11, 4,
4, 12, 4, 4, 17, 4, 17, 4, 20, 8, 20, 4, 20, 4, 20, 4,
21, 1, 153, 32, 4, 21, 1, 211, 32, 32, 192, 33, 32, 193, 35, 32,
@@ -1945,7 +1945,7 @@
static const orc_uint8 bc[] = {
1, 9, 44, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 102,
- 51,
+ 51,
50, 95, 99, 104, 50, 95, 112, 115, 121, 95, 114, 105, 103, 104, 116, 11,
8, 8, 12, 8, 8, 17, 4, 17, 4, 20, 4, 20, 4, 20, 4, 192,
32, 4, 193, 33, 4, 202, 34, 32, 25, 202, 32, 32, 24, 200, 33, 34,
@@ -2205,7 +2205,7 @@
static const orc_uint8 bc[] = {
1, 9, 43, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 102,
- 51,
+ 51,
50, 95, 99, 104, 50, 95, 112, 115, 121, 95, 108, 101, 102, 116, 11, 8,
8, 12, 8, 8, 17, 4, 17, 4, 20, 4, 20, 4, 20, 4, 192, 32,
4, 193, 34, 4, 202, 33, 34, 24, 202, 34, 34, 25, 200, 32, 33, 32,
@@ -2433,7 +2433,7 @@
static const orc_uint8 bc[] = {
1, 9, 44, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 115,
- 49,
+ 49,
54, 95, 99, 104, 49, 95, 115, 105, 109, 95, 114, 105, 103, 104, 116, 11,
4, 4, 12, 2, 2, 17, 4, 20, 8, 20, 4, 20, 4, 153, 33, 4,
211, 33, 33, 202, 34, 33, 24, 194, 32, 33, 34, 21, 1, 210, 32, 32,
@@ -2655,7 +2655,7 @@
static const orc_uint8 bc[] = {
1, 9, 43, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 115,
- 49,
+ 49,
54, 95, 99, 104, 49, 95, 115, 105, 109, 95, 108, 101, 102, 116, 11, 4,
4, 12, 2, 2, 17, 4, 20, 8, 20, 4, 20, 4, 153, 34, 4, 211,
34, 34, 202, 33, 34, 24, 194, 32, 33, 34, 21, 1, 210, 32, 32, 21,
@@ -2909,7 +2909,7 @@
static const orc_uint8 bc[] = {
1, 9, 44, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 115,
- 49,
+ 49,
54, 95, 99, 104, 50, 95, 115, 105, 109, 95, 114, 105, 103, 104, 116, 11,
4, 4, 12, 4, 4, 17, 4, 20, 8, 20, 4, 20, 4, 21, 1, 153,
32, 4, 21, 1, 211, 32, 32, 192, 33, 32, 193, 34, 32, 202, 34, 34,
@@ -3168,7 +3168,7 @@
static const orc_uint8 bc[] = {
1, 9, 43, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 115,
- 49,
+ 49,
54, 95, 99, 104, 50, 95, 115, 105, 109, 95, 108, 101, 102, 116, 11, 4,
4, 12, 4, 4, 17, 4, 20, 8, 20, 4, 20, 4, 21, 1, 153, 32,
4, 21, 1, 211, 32, 32, 192, 33, 32, 193, 34, 32, 202, 33, 33, 24,
@@ -3355,7 +3355,7 @@
static const orc_uint8 bc[] = {
1, 9, 44, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 102,
- 51,
+ 51,
50, 95, 99, 104, 49, 95, 115, 105, 109, 95, 114, 105, 103, 104, 116, 11,
8, 8, 12, 4, 4, 17, 4, 20, 4, 20, 4, 112, 32, 4, 202, 33,
4, 24, 194, 0, 32, 33, 2, 0,
@@ -3529,7 +3529,7 @@
static const orc_uint8 bc[] = {
1, 9, 43, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 102,
- 51,
+ 51,
50, 95, 99, 104, 49, 95, 115, 105, 109, 95, 108, 101, 102, 116, 11, 8,
8, 12, 4, 4, 17, 4, 20, 4, 20, 4, 202, 32, 4, 24, 112, 33,
4, 194, 0, 32, 33, 2, 0,
@@ -3725,7 +3725,7 @@
static const orc_uint8 bc[] = {
1, 9, 44, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 102,
- 51,
+ 51,
50, 95, 99, 104, 50, 95, 115, 105, 109, 95, 114, 105, 103, 104, 116, 11,
8, 8, 12, 8, 8, 17, 4, 20, 4, 20, 4, 192, 32, 4, 193, 33,
4, 202, 33, 33, 24, 194, 0, 32, 33, 2, 0,
@@ -3923,7 +3923,7 @@
static const orc_uint8 bc[] = {
1, 9, 43, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 102,
- 51,
+ 51,
50, 95, 99, 104, 50, 95, 115, 105, 109, 95, 108, 101, 102, 116, 11, 8,
8, 12, 8, 8, 17, 4, 20, 4, 20, 4, 192, 32, 4, 193, 33, 4,
202, 32, 32, 24, 194, 0, 32, 33, 2, 0,
diff --git a/gst/audioparsers/gstac3parse.c b/gst/audioparsers/gstac3parse.c
index c166dc8..4a6cd6d 100644
--- a/gst/audioparsers/gstac3parse.c
+++ b/gst/audioparsers/gstac3parse.c
@@ -478,7 +478,8 @@
goto cleanup;
} else {
GST_DEBUG_OBJECT (parse, "unexpected bsid %d", bsid);
- return FALSE;
+ ret = FALSE;
+ goto cleanup;
}
GST_DEBUG_OBJECT (parse, "unexpected bsid %d", bsid);
diff --git a/gst/audioparsers/gstflacparse.c b/gst/audioparsers/gstflacparse.c
index 51e1bfc..93ff7bd 100644
--- a/gst/audioparsers/gstflacparse.c
+++ b/gst/audioparsers/gstflacparse.c
@@ -305,6 +305,10 @@
gst_toc_unref (flacparse->toc);
flacparse->toc = NULL;
}
+ if (flacparse->seektable) {
+ gst_buffer_unref (flacparse->seektable);
+ flacparse->seektable = NULL;
+ }
g_list_foreach (flacparse->headers, (GFunc) gst_mini_object_unref, NULL);
g_list_free (flacparse->headers);
@@ -1163,6 +1167,8 @@
GST_DEBUG_OBJECT (flacparse, "storing seektable");
/* only store for now;
* offset of the first frame is needed to get real info */
+ if (flacparse->seektable)
+ gst_buffer_unref (flacparse->seektable);
flacparse->seektable = gst_buffer_ref (buffer);
return TRUE;
diff --git a/gst/auparse/Makefile.am b/gst/auparse/Makefile.am
index 4e8bfd8..643aa8a 100644
--- a/gst/auparse/Makefile.am
+++ b/gst/auparse/Makefile.am
@@ -7,17 +7,3 @@
libgstauparse_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
noinst_HEADERS = gstauparse.h
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstauparse -:SHARED libgstauparse \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstauparse_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstauparse_la_CFLAGS) \
- -:LDFLAGS $(libgstauparse_la_LDFLAGS) \
- $(libgstauparse_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/auparse/Makefile.in b/gst/auparse/Makefile.in
index 3922dc6..e882474 100644
--- a/gst/auparse/Makefile.in
+++ b/gst/auparse/Makefile.in
@@ -882,20 +882,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstauparse -:SHARED libgstauparse \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstauparse_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstauparse_la_CFLAGS) \
- -:LDFLAGS $(libgstauparse_la_LDFLAGS) \
- $(libgstauparse_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/autodetect/Makefile.am b/gst/autodetect/Makefile.am
index f670147..6815c4b 100644
--- a/gst/autodetect/Makefile.am
+++ b/gst/autodetect/Makefile.am
@@ -17,17 +17,3 @@
gstautodetect.h \
gstautovideosink.h \
gstautovideosrc.h
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstautodetect -:SHARED libgstautodetect \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstautodetect_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstautodetect_la_CFLAGS) \
- -:LDFLAGS $(libgstautodetect_la_LDFLAGS) \
- $(libgstautodetect_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/autodetect/Makefile.in b/gst/autodetect/Makefile.in
index d22b4ce..23118cc 100644
--- a/gst/autodetect/Makefile.in
+++ b/gst/autodetect/Makefile.in
@@ -930,20 +930,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstautodetect -:SHARED libgstautodetect \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstautodetect_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstautodetect_la_CFLAGS) \
- -:LDFLAGS $(libgstautodetect_la_LDFLAGS) \
- $(libgstautodetect_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/avi/Makefile.am b/gst/avi/Makefile.am
index 7170e1f..926df27 100644
--- a/gst/avi/Makefile.am
+++ b/gst/avi/Makefile.am
@@ -26,17 +26,3 @@
libgstavi_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
EXTRA_DIST = README
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstavi -:SHARED libgstavi \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstavi_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstavi_la_CFLAGS) \
- -:LDFLAGS $(libgstavi_la_LDFLAGS) \
- $(libgstavi_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/avi/Makefile.in b/gst/avi/Makefile.in
index ed187b0..22a14a2 100644
--- a/gst/avi/Makefile.in
+++ b/gst/avi/Makefile.in
@@ -927,20 +927,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstavi -:SHARED libgstavi \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstavi_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstavi_la_CFLAGS) \
- -:LDFLAGS $(libgstavi_la_LDFLAGS) \
- $(libgstavi_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/avi/gstavimux.c b/gst/avi/gstavimux.c
index 225e824..ae3d74d 100644
--- a/gst/avi/gstavimux.c
+++ b/gst/avi/gstavimux.c
@@ -76,8 +76,8 @@
enum
{
- ARG_0,
- ARG_BIGFILE
+ PROP_0,
+ PROP_BIGFILE
};
#define DEFAULT_BIGFILE TRUE
@@ -242,7 +242,7 @@
gobject_class->set_property = gst_avi_mux_set_property;
gobject_class->finalize = gst_avi_mux_finalize;
- g_object_class_install_property (gobject_class, ARG_BIGFILE,
+ g_object_class_install_property (gobject_class, PROP_BIGFILE,
g_param_spec_boolean ("bigfile", "Bigfile Support (>2GB)",
"Support for openDML-2.0 (big) AVI files", DEFAULT_BIGFILE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
@@ -2217,7 +2217,7 @@
avimux = GST_AVI_MUX (object);
switch (prop_id) {
- case ARG_BIGFILE:
+ case PROP_BIGFILE:
g_value_set_boolean (value, avimux->enable_large_avi);
break;
default:
@@ -2235,7 +2235,7 @@
avimux = GST_AVI_MUX (object);
switch (prop_id) {
- case ARG_BIGFILE:
+ case PROP_BIGFILE:
avimux->enable_large_avi = g_value_get_boolean (value);
break;
default:
diff --git a/gst/cutter/Makefile.am b/gst/cutter/Makefile.am
index f72f946..a285fab 100644
--- a/gst/cutter/Makefile.am
+++ b/gst/cutter/Makefile.am
@@ -9,17 +9,3 @@
noinst_HEADERS = gstcutter.h filter.func
EXTRA_DIST = README
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstcutter -:SHARED libgstcutter \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstcutter_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstcutter_la_CFLAGS) \
- -:LDFLAGS $(libgstcutter_la_LDFLAGS) \
- $(libgstcutter_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/cutter/Makefile.in b/gst/cutter/Makefile.in
index 54c0649..99cca3e 100644
--- a/gst/cutter/Makefile.in
+++ b/gst/cutter/Makefile.in
@@ -883,20 +883,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstcutter -:SHARED libgstcutter \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstcutter_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstcutter_la_CFLAGS) \
- -:LDFLAGS $(libgstcutter_la_LDFLAGS) \
- $(libgstcutter_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/debugutils/Makefile.am b/gst/debugutils/Makefile.am
index 1a953d4..105be36 100644
--- a/gst/debugutils/Makefile.am
+++ b/gst/debugutils/Makefile.am
@@ -39,17 +39,3 @@
libgstdebug_la_LIBADD = $(GST_LIBS) $(GST_BASE_LIBS)
libgstdebug_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstdebug_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstdebug -:SHARED libgstdebug \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstdebug_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstdebug_la_CFLAGS) \
- -:LDFLAGS $(libgstdebug_la_LDFLAGS) \
- $(libgstdebug_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/debugutils/Makefile.in b/gst/debugutils/Makefile.in
index 1b55cae..adffb89 100644
--- a/gst/debugutils/Makefile.in
+++ b/gst/debugutils/Makefile.in
@@ -1023,20 +1023,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstdebug -:SHARED libgstdebug \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstdebug_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstdebug_la_CFLAGS) \
- -:LDFLAGS $(libgstdebug_la_LDFLAGS) \
- $(libgstdebug_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/debugutils/breakmydata.c b/gst/debugutils/breakmydata.c
index c7306b1..7fe01b6 100644
--- a/gst/debugutils/breakmydata.c
+++ b/gst/debugutils/breakmydata.c
@@ -49,11 +49,11 @@
enum
{
- ARG_0,
- ARG_SEED,
- ARG_SET_TO,
- ARG_SKIP,
- ARG_PROBABILITY
+ PROP_0,
+ PROP_SEED,
+ PROP_SET_TO,
+ PROP_SKIP,
+ PROP_PROBABILITY
};
typedef struct _GstBreakMyData GstBreakMyData;
@@ -118,22 +118,22 @@
gobject_class->set_property = gst_break_my_data_set_property;
gobject_class->get_property = gst_break_my_data_get_property;
- g_object_class_install_property (gobject_class, ARG_SEED,
+ g_object_class_install_property (gobject_class, PROP_SEED,
g_param_spec_uint ("seed", "seed",
"seed for randomness (initialized when going from READY to PAUSED)",
0, G_MAXUINT32, 0,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
- g_object_class_install_property (gobject_class, ARG_SET_TO,
+ g_object_class_install_property (gobject_class, PROP_SET_TO,
g_param_spec_int ("set-to", "set-to",
"set changed bytes to this value (-1 means random value",
-1, G_MAXUINT8, -1,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
- g_object_class_install_property (gobject_class, ARG_SKIP,
+ g_object_class_install_property (gobject_class, PROP_SKIP,
g_param_spec_uint ("skip", "skip",
"amount of bytes skipped at the beginning of stream",
0, G_MAXUINT, 0,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
- g_object_class_install_property (gobject_class, ARG_PROBABILITY,
+ g_object_class_install_property (gobject_class, PROP_PROBABILITY,
g_param_spec_double ("probability", "probability",
"probability for each byte in the buffer to be changed", 0.0, 1.0,
0.0, G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
@@ -168,16 +168,16 @@
GST_OBJECT_LOCK (bmd);
switch (prop_id) {
- case ARG_SEED:
+ case PROP_SEED:
bmd->seed = g_value_get_uint (value);
break;
- case ARG_SET_TO:
+ case PROP_SET_TO:
bmd->set = g_value_get_int (value);
break;
- case ARG_SKIP:
+ case PROP_SKIP:
bmd->skip = g_value_get_uint (value);
break;
- case ARG_PROBABILITY:
+ case PROP_PROBABILITY:
bmd->probability = g_value_get_double (value);
break;
default:
@@ -197,16 +197,16 @@
GST_OBJECT_LOCK (bmd);
switch (prop_id) {
- case ARG_SEED:
+ case PROP_SEED:
g_value_set_uint (value, bmd->seed);
break;
- case ARG_SET_TO:
+ case PROP_SET_TO:
g_value_set_int (value, bmd->set);
break;
- case ARG_SKIP:
+ case PROP_SKIP:
g_value_set_uint (value, bmd->skip);
break;
- case ARG_PROBABILITY:
+ case PROP_PROBABILITY:
g_value_set_double (value, bmd->probability);
break;
default:
diff --git a/gst/debugutils/cpureport.c b/gst/debugutils/cpureport.c
index 85cc294..670794d 100644
--- a/gst/debugutils/cpureport.c
+++ b/gst/debugutils/cpureport.c
@@ -31,7 +31,7 @@
enum
{
- ARG_0,
+ PROP_0,
};
GstStaticPadTemplate cpu_report_src_template = GST_STATIC_PAD_TEMPLATE ("src",
diff --git a/gst/debugutils/gstnavseek.c b/gst/debugutils/gstnavseek.c
index b6fae2e..a38b608 100644
--- a/gst/debugutils/gstnavseek.c
+++ b/gst/debugutils/gstnavseek.c
@@ -33,8 +33,8 @@
enum
{
- ARG_0,
- ARG_SEEKOFFSET
+ PROP_0,
+ PROP_SEEKOFFSET
};
GstStaticPadTemplate navseek_src_template = GST_STATIC_PAD_TEMPLATE ("src",
@@ -80,7 +80,7 @@
gobject_class->get_property = gst_navseek_get_property;
g_object_class_install_property (gobject_class,
- ARG_SEEKOFFSET, g_param_spec_double ("seek-offset", "Seek Offset",
+ PROP_SEEKOFFSET, g_param_spec_double ("seek-offset", "Seek Offset",
"Time in seconds to seek by", 0.0, G_MAXDOUBLE, 5.0,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
@@ -302,7 +302,7 @@
GstNavSeek *navseek = GST_NAVSEEK (object);
switch (prop_id) {
- case ARG_SEEKOFFSET:
+ case PROP_SEEKOFFSET:
GST_OBJECT_LOCK (navseek);
navseek->seek_offset = g_value_get_double (value);
GST_OBJECT_UNLOCK (navseek);
@@ -320,7 +320,7 @@
GstNavSeek *navseek = GST_NAVSEEK (object);
switch (prop_id) {
- case ARG_SEEKOFFSET:
+ case PROP_SEEKOFFSET:
GST_OBJECT_LOCK (navseek);
g_value_set_double (value, navseek->seek_offset);
GST_OBJECT_UNLOCK (navseek);
diff --git a/gst/debugutils/progressreport.c b/gst/debugutils/progressreport.c
index 859e4b9..7dd05dc 100644
--- a/gst/debugutils/progressreport.c
+++ b/gst/debugutils/progressreport.c
@@ -78,11 +78,11 @@
enum
{
- ARG_0,
- ARG_UPDATE_FREQ,
- ARG_SILENT,
- ARG_DO_QUERY,
- ARG_FORMAT
+ PROP_0,
+ PROP_UPDATE_FREQ,
+ PROP_SILENT,
+ PROP_DO_QUERY,
+ PROP_FORMAT
};
GstStaticPadTemplate progress_report_src_template =
@@ -145,23 +145,23 @@
gobject_class->get_property = gst_progress_report_get_property;
g_object_class_install_property (gobject_class,
- ARG_UPDATE_FREQ, g_param_spec_int ("update-freq", "Update Frequency",
+ PROP_UPDATE_FREQ, g_param_spec_int ("update-freq", "Update Frequency",
"Number of seconds between reports when data is flowing", 1, G_MAXINT,
DEFAULT_UPDATE_FREQ, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
- ARG_SILENT, g_param_spec_boolean ("silent",
+ PROP_SILENT, g_param_spec_boolean ("silent",
"Do not print output to stdout", "Do not print output to stdout",
DEFAULT_SILENT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
- ARG_DO_QUERY, g_param_spec_boolean ("do-query",
+ PROP_DO_QUERY, g_param_spec_boolean ("do-query",
"Use a query instead of buffer metadata to determine stream position",
"Use a query instead of buffer metadata to determine stream position",
DEFAULT_DO_QUERY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
- ARG_FORMAT, g_param_spec_string ("format", "format",
+ PROP_FORMAT, g_param_spec_string ("format", "format",
"Format to use for the querying", DEFAULT_FORMAT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
@@ -452,22 +452,22 @@
filter = GST_PROGRESS_REPORT (object);
switch (prop_id) {
- case ARG_UPDATE_FREQ:
+ case PROP_UPDATE_FREQ:
GST_OBJECT_LOCK (filter);
filter->update_freq = g_value_get_int (value);
GST_OBJECT_UNLOCK (filter);
break;
- case ARG_SILENT:
+ case PROP_SILENT:
GST_OBJECT_LOCK (filter);
filter->silent = g_value_get_boolean (value);
GST_OBJECT_UNLOCK (filter);
break;
- case ARG_DO_QUERY:
+ case PROP_DO_QUERY:
GST_OBJECT_LOCK (filter);
filter->do_query = g_value_get_boolean (value);
GST_OBJECT_UNLOCK (filter);
break;
- case ARG_FORMAT:
+ case PROP_FORMAT:
GST_OBJECT_LOCK (filter);
g_free (filter->format);
filter->format = g_value_dup_string (value);
@@ -489,22 +489,22 @@
filter = GST_PROGRESS_REPORT (object);
switch (prop_id) {
- case ARG_UPDATE_FREQ:
+ case PROP_UPDATE_FREQ:
GST_OBJECT_LOCK (filter);
g_value_set_int (value, filter->update_freq);
GST_OBJECT_UNLOCK (filter);
break;
- case ARG_SILENT:
+ case PROP_SILENT:
GST_OBJECT_LOCK (filter);
g_value_set_boolean (value, filter->silent);
GST_OBJECT_UNLOCK (filter);
break;
- case ARG_DO_QUERY:
+ case PROP_DO_QUERY:
GST_OBJECT_LOCK (filter);
g_value_set_boolean (value, filter->do_query);
GST_OBJECT_UNLOCK (filter);
break;
- case ARG_FORMAT:
+ case PROP_FORMAT:
GST_OBJECT_LOCK (filter);
g_value_set_string (value, filter->format);
GST_OBJECT_UNLOCK (filter);
diff --git a/gst/debugutils/rndbuffersize.c b/gst/debugutils/rndbuffersize.c
index 009f833..9ebd12c 100644
--- a/gst/debugutils/rndbuffersize.c
+++ b/gst/debugutils/rndbuffersize.c
@@ -65,9 +65,9 @@
enum
{
- ARG_SEED = 1,
- ARG_MINIMUM,
- ARG_MAXIMUM
+ PROP_SEED = 1,
+ PROP_MINIMUM,
+ PROP_MAXIMUM
};
#define DEFAULT_SEED 0
@@ -132,16 +132,16 @@
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_rnd_buffer_size_change_state);
- g_object_class_install_property (gobject_class, ARG_SEED,
+ g_object_class_install_property (gobject_class, PROP_SEED,
g_param_spec_uint ("seed", "random number seed",
"seed for randomness (initialized when going from READY to PAUSED)",
0, G_MAXUINT32, DEFAULT_SEED,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
- g_object_class_install_property (gobject_class, ARG_MINIMUM,
+ g_object_class_install_property (gobject_class, PROP_MINIMUM,
g_param_spec_int ("min", "mininum", "mininum buffer size",
0, G_MAXINT32, DEFAULT_MIN,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
- g_object_class_install_property (gobject_class, ARG_MAXIMUM,
+ g_object_class_install_property (gobject_class, PROP_MAXIMUM,
g_param_spec_int ("max", "maximum", "maximum buffer size",
1, G_MAXINT32, DEFAULT_MAX,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
@@ -195,13 +195,13 @@
GstRndBufferSize *self = GST_RND_BUFFER_SIZE (object);
switch (prop_id) {
- case ARG_SEED:
+ case PROP_SEED:
self->seed = g_value_get_uint (value);
break;
- case ARG_MINIMUM:
+ case PROP_MINIMUM:
self->min = g_value_get_int (value);
break;
- case ARG_MAXIMUM:
+ case PROP_MAXIMUM:
self->max = g_value_get_int (value);
break;
default:
@@ -218,13 +218,13 @@
GstRndBufferSize *self = GST_RND_BUFFER_SIZE (object);
switch (prop_id) {
- case ARG_SEED:
+ case PROP_SEED:
g_value_set_uint (value, self->seed);
break;
- case ARG_MINIMUM:
+ case PROP_MINIMUM:
g_value_set_int (value, self->min);
break;
- case ARG_MAXIMUM:
+ case PROP_MAXIMUM:
g_value_set_int (value, self->max);
break;
default:
diff --git a/gst/deinterlace/Makefile.am b/gst/deinterlace/Makefile.am
index 7d5dbb0..c0f4fa9 100644
--- a/gst/deinterlace/Makefile.am
+++ b/gst/deinterlace/Makefile.am
@@ -51,18 +51,3 @@
tvtime/tomsmocomp/TomsMoCompAll.inc \
tvtime/tomsmocomp/tomsmocompmacros.h \
tvtime/tomsmocomp/WierdBob.inc
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstdeinterlace -:SHARED libgstdeinterlace \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstdeinterlace_la_SOURCES) \
- $(nodist_libgstdeinterlace_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstdeinterlace_la_CFLAGS) \
- -:LDFLAGS $(libgstdeinterlace_la_LDFLAGS) \
- $(libgstdeinterlace_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/deinterlace/Makefile.in b/gst/deinterlace/Makefile.in
index b6b3a85..926ab5d 100644
--- a/gst/deinterlace/Makefile.in
+++ b/gst/deinterlace/Makefile.in
@@ -1148,21 +1148,6 @@
cp -p tmp-orc.c $(distdir)/$(ORC_SOURCE)-dist.c
cp -p $(ORC_SOURCE).h $(distdir)/$(ORC_SOURCE)-dist.h
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstdeinterlace -:SHARED libgstdeinterlace \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstdeinterlace_la_SOURCES) \
- $(nodist_libgstdeinterlace_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstdeinterlace_la_CFLAGS) \
- -:LDFLAGS $(libgstdeinterlace_la_LDFLAGS) \
- $(libgstdeinterlace_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/deinterlace/gstdeinterlace.c b/gst/deinterlace/gstdeinterlace.c
index 9a89f5c..f0e0f31 100644
--- a/gst/deinterlace/gstdeinterlace.c
+++ b/gst/deinterlace/gstdeinterlace.c
@@ -75,8 +75,7 @@
PROP_FIELD_LAYOUT,
PROP_LOCKING,
PROP_IGNORE_OBSCURE,
- PROP_DROP_ORPHANS,
- PROP_LAST
+ PROP_DROP_ORPHANS
};
#define GST_DEINTERLACE_BUFFER_STATE_P (1<<0)
diff --git a/gst/deinterlace/tvtime-dist.c b/gst/deinterlace/tvtime-dist.c
index d92ce30..9ac3346 100644
--- a/gst/deinterlace/tvtime-dist.c
+++ b/gst/deinterlace/tvtime-dist.c
@@ -713,7 +713,7 @@
static const orc_uint8 bc[] = {
1, 9, 29, 100, 101, 105, 110, 116, 101, 114, 108, 97, 99, 101, 95, 108,
105, 110, 101, 95, 108, 105, 110, 101, 97, 114, 95, 98, 108, 101, 110,
- 100,
+ 100,
11, 1, 1, 12, 1, 1, 12, 1, 1, 12, 1, 1, 14, 2, 2, 0,
0, 0, 20, 2, 20, 2, 20, 2, 150, 32, 4, 150, 33, 5, 150, 34,
6, 70, 32, 32, 33, 70, 34, 34, 34, 70, 32, 32, 34, 70, 32, 32,
diff --git a/gst/dtmf/Makefile.am b/gst/dtmf/Makefile.am
index f5d36ed..464b90c 100644
--- a/gst/dtmf/Makefile.am
+++ b/gst/dtmf/Makefile.am
@@ -15,17 +15,3 @@
$(GST_BASE_LIBS) $(GST_LIBS) $(LIBM)
libgstdtmf_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstdtmf_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstdtmf -:SHARED libgstdtmf \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstdtmf_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstdtmf_la_CFLAGS) \
- -:LDFLAGS $(libgstdtmf_la_LDFLAGS) \
- $(libgstdtmf_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/dtmf/Makefile.in b/gst/dtmf/Makefile.in
index 2debf60..7af3f46 100644
--- a/gst/dtmf/Makefile.in
+++ b/gst/dtmf/Makefile.in
@@ -919,20 +919,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstdtmf -:SHARED libgstdtmf \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstdtmf_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstdtmf_la_CFLAGS) \
- -:LDFLAGS $(libgstdtmf_la_LDFLAGS) \
- $(libgstdtmf_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/dtmf/gstrtpdtmfdepay.c b/gst/dtmf/gstrtpdtmfdepay.c
index 01c039c..f625ac5 100644
--- a/gst/dtmf/gstrtpdtmfdepay.c
+++ b/gst/dtmf/gstrtpdtmfdepay.c
@@ -158,8 +158,6 @@
enum
{
-
-
/* FILL ME */
LAST_SIGNAL
};
@@ -171,11 +169,6 @@
PROP_MAX_DURATION
};
-enum
-{
- ARG_0
-};
-
static GstStaticPadTemplate gst_rtp_dtmf_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
diff --git a/gst/effectv/Makefile.am b/gst/effectv/Makefile.am
index d09111d..14dff1d 100644
--- a/gst/effectv/Makefile.am
+++ b/gst/effectv/Makefile.am
@@ -20,17 +20,3 @@
noinst_HEADERS = gsteffectv.h gstaging.h gstdice.h gstedge.h \
gstquark.h gstrev.h gstshagadelic.h gstvertigo.h gstwarp.h gstop.h \
gstradioac.h gststreak.h gstripple.h
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgsteffectv -:SHARED libgsteffectv \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgsteffectv_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgsteffectv_la_CFLAGS) \
- -:LDFLAGS $(libgsteffectv_la_LDFLAGS) \
- $(libgsteffectv_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/effectv/Makefile.in b/gst/effectv/Makefile.in
index a879f37..2b26f90 100644
--- a/gst/effectv/Makefile.in
+++ b/gst/effectv/Makefile.in
@@ -1003,20 +1003,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgsteffectv -:SHARED libgsteffectv \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgsteffectv_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgsteffectv_la_CFLAGS) \
- -:LDFLAGS $(libgsteffectv_la_LDFLAGS) \
- $(libgsteffectv_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/effectv/gstquark.c b/gst/effectv/gstquark.c
index cb02643..098e20c 100644
--- a/gst/effectv/gstquark.c
+++ b/gst/effectv/gstquark.c
@@ -272,7 +272,7 @@
g_object_class_install_property (gobject_class, PROP_PLANES,
g_param_spec_int ("planes", "Planes",
- "Number of planes", 0, 64, PLANES,
+ "Number of planes", 1, 64, PLANES,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | GST_PARAM_CONTROLLABLE));
gst_element_class_set_static_metadata (gstelement_class, "QuarkTV effect",
diff --git a/gst/equalizer/Makefile.am b/gst/equalizer/Makefile.am
index 9f92a9e..8ce4b78 100644
--- a/gst/equalizer/Makefile.am
+++ b/gst/equalizer/Makefile.am
@@ -20,17 +20,3 @@
preset_DATA = GstIirEqualizer3Bands.prs GstIirEqualizer10Bands.prs
EXTRA_DIST = $(preset_DATA)
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstequalizer -:SHARED libgstequalizer \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstequalizer_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstequalizer_la_CFLAGS) \
- -:LDFLAGS $(libgstequalizer_la_LDFLAGS) \
- $(libgstequalizer_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/equalizer/Makefile.in b/gst/equalizer/Makefile.in
index f69929f..c8a10f0 100644
--- a/gst/equalizer/Makefile.in
+++ b/gst/equalizer/Makefile.in
@@ -946,20 +946,6 @@
uninstall-am uninstall-pluginLTLIBRARIES uninstall-presetDATA
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstequalizer -:SHARED libgstequalizer \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstequalizer_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstequalizer_la_CFLAGS) \
- -:LDFLAGS $(libgstequalizer_la_LDFLAGS) \
- $(libgstequalizer_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/equalizer/gstiirequalizer.c b/gst/equalizer/gstiirequalizer.c
index e912ca5..eb23d21 100644
--- a/gst/equalizer/gstiirequalizer.c
+++ b/gst/equalizer/gstiirequalizer.c
@@ -49,6 +49,7 @@
const GstAudioInfo * info);
static GstFlowReturn gst_iir_equalizer_transform_ip (GstBaseTransform * btrans,
GstBuffer * buf);
+static void set_passthrough (GstIirEqualizer * equ);
#define ALLOWED_CAPS \
"audio/x-raw," \
@@ -159,6 +160,7 @@
BANDS_LOCK (equ);
equ->need_new_coefficients = TRUE;
band->gain = gain;
+ set_passthrough (equ);
BANDS_UNLOCK (equ);
GST_DEBUG_OBJECT (band, "changed gain = %lf ", band->gain);
}
@@ -373,7 +375,8 @@
gst_iir_equalizer_init (GstIirEqualizer * eq)
{
g_mutex_init (&eq->bands_lock);
- eq->need_new_coefficients = TRUE;
+ /* Band gains are 0 by default, passthrough until they are changed */
+ gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (eq), TRUE);
}
static void
@@ -852,7 +855,6 @@
BANDS_LOCK (equ);
if (need_new_coefficients) {
update_coefficients (equ);
- set_passthrough (equ);
}
BANDS_UNLOCK (equ);
diff --git a/gst/flv/Makefile.am b/gst/flv/Makefile.am
index e6e6578..b396743 100644
--- a/gst/flv/Makefile.am
+++ b/gst/flv/Makefile.am
@@ -11,17 +11,3 @@
noinst_HEADERS = gstflvdemux.h gstflvmux.h amfdefs.h gstindex.h
EXTRA_DIST = gstindex.c gstmemindex.c
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstflv -:SHARED libgstflv \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstflv_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstflv_la_CFLAGS) \
- -:LDFLAGS $(libgstflv_la_LDFLAGS) \
- $(libgstflv_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/flv/Makefile.in b/gst/flv/Makefile.in
index ff636bb..84b4254 100644
--- a/gst/flv/Makefile.in
+++ b/gst/flv/Makefile.in
@@ -895,20 +895,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstflv -:SHARED libgstflv \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstflv_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstflv_la_CFLAGS) \
- -:LDFLAGS $(libgstflv_la_LDFLAGS) \
- $(libgstflv_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/flv/gstflvdemux.c b/gst/flv/gstflvdemux.c
index ec8f78b..afa1490 100644
--- a/gst/flv/gstflvdemux.c
+++ b/gst/flv/gstflvdemux.c
@@ -663,7 +663,7 @@
gst_flv_demux_audio_negotiate (GstFlvDemux * demux, guint32 codec_tag,
guint32 rate, guint32 channels, guint32 width)
{
- GstCaps *caps = NULL;
+ GstCaps *caps = NULL, *old_caps;
gchar *codec_name = NULL;
gboolean ret = FALSE;
guint adjusted_rate = rate;
@@ -704,30 +704,33 @@
break;
case 10:
{
- if (demux->audio_codec_data) {
- GstMapInfo map;
-
- gst_buffer_map (demux->audio_codec_data, &map, GST_MAP_READ);
-
- /* use codec-data to extract and verify samplerate */
- if (map.size >= 2) {
- gint freq_index;
-
- freq_index = GST_READ_UINT16_BE (map.data);
- freq_index = (freq_index & 0x0780) >> 7;
- adjusted_rate =
- gst_codec_utils_aac_get_sample_rate_from_index (freq_index);
-
- if (adjusted_rate && (rate != adjusted_rate)) {
- GST_LOG_OBJECT (demux, "Ajusting AAC sample rate %d -> %d", rate,
- adjusted_rate);
- } else {
- adjusted_rate = rate;
- }
- }
- gst_buffer_unmap (demux->audio_codec_data, &map);
+ GstMapInfo map;
+ if (!demux->audio_codec_data) {
+ GST_DEBUG_OBJECT (demux, "don't have AAC codec data yet");
+ ret = TRUE;
+ goto done;
}
+ gst_buffer_map (demux->audio_codec_data, &map, GST_MAP_READ);
+
+ /* use codec-data to extract and verify samplerate */
+ if (map.size >= 2) {
+ gint freq_index;
+
+ freq_index = GST_READ_UINT16_BE (map.data);
+ freq_index = (freq_index & 0x0780) >> 7;
+ adjusted_rate =
+ gst_codec_utils_aac_get_sample_rate_from_index (freq_index);
+
+ if (adjusted_rate && (rate != adjusted_rate)) {
+ GST_LOG_OBJECT (demux, "Ajusting AAC sample rate %d -> %d", rate,
+ adjusted_rate);
+ } else {
+ adjusted_rate = rate;
+ }
+ }
+ gst_buffer_unmap (demux->audio_codec_data, &map);
+
caps = gst_caps_new_simple ("audio/mpeg",
"mpegversion", G_TYPE_INT, 4, "framed", G_TYPE_BOOLEAN, TRUE,
"stream-format", G_TYPE_STRING, "raw", NULL);
@@ -811,18 +814,27 @@
demux->audio_codec_data, NULL);
}
- stream_id =
- gst_pad_create_stream_id (demux->audio_pad, GST_ELEMENT_CAST (demux),
- "audio");
+ old_caps = gst_pad_get_current_caps (demux->audio_pad);
+ if (!old_caps) {
+ stream_id =
+ gst_pad_create_stream_id (demux->audio_pad, GST_ELEMENT_CAST (demux),
+ "audio");
- event = gst_event_new_stream_start (stream_id);
- if (have_group_id (demux))
- gst_event_set_group_id (event, demux->group_id);
- gst_pad_push_event (demux->audio_pad, event);
- g_free (stream_id);
+ event = gst_event_new_stream_start (stream_id);
+ if (have_group_id (demux))
+ gst_event_set_group_id (event, demux->group_id);
+ gst_pad_push_event (demux->audio_pad, event);
+ g_free (stream_id);
+ }
+ if (!old_caps || !gst_caps_is_equal (old_caps, caps))
+ ret = gst_pad_set_caps (demux->audio_pad, caps);
+ else
+ ret = TRUE;
- ret = gst_pad_set_caps (demux->audio_pad, caps);
+ if (old_caps)
+ gst_caps_unref (old_caps);
+done:
if (G_LIKELY (ret)) {
/* Store the caps we got from tags */
demux->audio_codec_tag = codec_tag;
@@ -830,24 +842,29 @@
demux->channels = channels;
demux->width = width;
- codec_name = gst_pb_utils_get_codec_description (caps);
+ if (caps) {
+ codec_name = gst_pb_utils_get_codec_description (caps);
- if (codec_name) {
- if (demux->taglist == NULL)
- demux->taglist = gst_tag_list_new_empty ();
- gst_tag_list_add (demux->taglist, GST_TAG_MERGE_REPLACE,
- GST_TAG_AUDIO_CODEC, codec_name, NULL);
- g_free (codec_name);
+ if (codec_name) {
+ if (demux->taglist == NULL)
+ demux->taglist = gst_tag_list_new_empty ();
+ gst_tag_list_add (demux->taglist, GST_TAG_MERGE_REPLACE,
+ GST_TAG_AUDIO_CODEC, codec_name, NULL);
+ g_free (codec_name);
+ }
+
+ GST_DEBUG_OBJECT (demux->audio_pad, "successfully negotiated caps %"
+ GST_PTR_FORMAT, caps);
+ } else {
+ GST_DEBUG_OBJECT (demux->audio_pad, "delayed setting caps");
}
-
- GST_DEBUG_OBJECT (demux->audio_pad, "successfully negotiated caps %"
- GST_PTR_FORMAT, caps);
} else {
GST_WARNING_OBJECT (demux->audio_pad, "failed negotiating caps %"
GST_PTR_FORMAT, caps);
}
- gst_caps_unref (caps);
+ if (caps)
+ gst_caps_unref (caps);
beach:
return ret;
@@ -927,16 +944,18 @@
GST_LOG_OBJECT (demux, "parsing an audio tag");
- if G_UNLIKELY (!demux->audio_pad && demux->no_more_pads) {
+ if G_UNLIKELY
+ (!demux->audio_pad && demux->no_more_pads) {
#ifndef GST_DISABLE_DEBUG
- if G_UNLIKELY (!demux->no_audio_warned) {
+ if G_UNLIKELY
+ (!demux->no_audio_warned) {
GST_WARNING_OBJECT (demux,
"Signaled no-more-pads already but had no audio pad -- ignoring");
demux->no_audio_warned = TRUE;
- }
+ }
#endif
return GST_FLOW_OK;
- }
+ }
g_return_val_if_fail (gst_buffer_get_size (buffer) == demux->tag_size,
GST_FLOW_ERROR);
@@ -1066,6 +1085,8 @@
codec_tag != demux->audio_codec_tag || width != demux->width)) {
GST_DEBUG_OBJECT (demux, "audio settings have changed, changing caps");
+ gst_buffer_replace (&demux->audio_codec_data, NULL);
+
/* Negotiate caps */
if (!gst_flv_demux_audio_negotiate (demux, codec_tag, rate, channels,
width)) {
@@ -1105,6 +1126,12 @@
goto beach;
}
case 1:
+ if (!demux->audio_codec_data) {
+ GST_ERROR_OBJECT (demux, "got AAC audio packet before codec data");
+ ret = GST_FLOW_OK;
+ gst_buffer_unref (outbuf);
+ goto beach;
+ }
/* AAC raw packet */
GST_LOG_OBJECT (demux, "got a raw AAC audio packet");
break;
@@ -1203,7 +1230,8 @@
goto beach;
}
- ret = gst_flow_combiner_update_flow (demux->flowcombiner, ret);
+ ret = gst_flow_combiner_update_pad_flow (demux->flowcombiner,
+ demux->audio_pad, ret);
beach:
gst_buffer_unmap (buffer, &map);
@@ -1215,7 +1243,7 @@
gst_flv_demux_video_negotiate (GstFlvDemux * demux, guint32 codec_tag)
{
gboolean ret = FALSE;
- GstCaps *caps = NULL;
+ GstCaps *caps = NULL, *old_caps;
gchar *codec_name = NULL;
GstEvent *event;
gchar *stream_id;
@@ -1237,6 +1265,11 @@
caps = gst_caps_new_empty_simple ("video/x-vp6-alpha");
break;
case 7:
+ if (!demux->video_codec_data) {
+ GST_DEBUG_OBJECT (demux, "don't have h264 codec data yet");
+ ret = TRUE;
+ goto done;
+ }
caps =
gst_caps_new_simple ("video/x-h264", "stream-format", G_TYPE_STRING,
"avc", NULL);
@@ -1277,39 +1310,55 @@
demux->video_codec_data, NULL);
}
- stream_id =
- gst_pad_create_stream_id (demux->video_pad, GST_ELEMENT_CAST (demux),
- "video");
- event = gst_event_new_stream_start (stream_id);
- g_free (stream_id);
+ old_caps = gst_pad_get_current_caps (demux->video_pad);
+ if (!old_caps) {
+ stream_id =
+ gst_pad_create_stream_id (demux->video_pad, GST_ELEMENT_CAST (demux),
+ "video");
+ event = gst_event_new_stream_start (stream_id);
+ g_free (stream_id);
- if (have_group_id (demux))
- gst_event_set_group_id (event, demux->group_id);
- gst_pad_push_event (demux->video_pad, event);
- ret = gst_pad_set_caps (demux->video_pad, caps);
+ if (have_group_id (demux))
+ gst_event_set_group_id (event, demux->group_id);
+ gst_pad_push_event (demux->video_pad, event);
+ }
+ if (!old_caps || !gst_caps_is_equal (old_caps, caps))
+ ret = gst_pad_set_caps (demux->video_pad, caps);
+ else
+ ret = TRUE;
+
+ if (old_caps)
+ gst_caps_unref (old_caps);
+
+done:
if (G_LIKELY (ret)) {
/* Store the caps we have set */
demux->video_codec_tag = codec_tag;
- codec_name = gst_pb_utils_get_codec_description (caps);
+ if (caps) {
+ codec_name = gst_pb_utils_get_codec_description (caps);
- if (codec_name) {
- if (demux->taglist == NULL)
- demux->taglist = gst_tag_list_new_empty ();
- gst_tag_list_add (demux->taglist, GST_TAG_MERGE_REPLACE,
- GST_TAG_VIDEO_CODEC, codec_name, NULL);
- g_free (codec_name);
+ if (codec_name) {
+ if (demux->taglist == NULL)
+ demux->taglist = gst_tag_list_new_empty ();
+ gst_tag_list_add (demux->taglist, GST_TAG_MERGE_REPLACE,
+ GST_TAG_VIDEO_CODEC, codec_name, NULL);
+ g_free (codec_name);
+ }
+
+ GST_DEBUG_OBJECT (demux->video_pad, "successfully negotiated caps %"
+ GST_PTR_FORMAT, caps);
+ } else {
+ GST_DEBUG_OBJECT (demux->video_pad, "delayed setting caps");
}
-
- GST_DEBUG_OBJECT (demux->video_pad, "successfully negotiated caps %"
- GST_PTR_FORMAT, caps);
} else {
GST_WARNING_OBJECT (demux->video_pad, "failed negotiating caps %"
GST_PTR_FORMAT, caps);
}
- gst_caps_unref (caps);
+ if (caps)
+ gst_caps_unref (caps);
beach:
return ret;
@@ -1332,16 +1381,18 @@
GST_LOG_OBJECT (demux, "parsing a video tag");
- if G_UNLIKELY (!demux->video_pad && demux->no_more_pads) {
+ if G_UNLIKELY
+ (!demux->video_pad && demux->no_more_pads) {
#ifndef GST_DISABLE_DEBUG
- if G_UNLIKELY (!demux->no_video_warned) {
+ if G_UNLIKELY
+ (!demux->no_video_warned) {
GST_WARNING_OBJECT (demux,
"Signaled no-more-pads already but had no video pad -- ignoring");
demux->no_video_warned = TRUE;
- }
+ }
#endif
return GST_FLOW_OK;
- }
+ }
if (gst_buffer_get_size (buffer) < 12) {
GST_ERROR_OBJECT (demux, "Too small tag size");
@@ -1448,8 +1499,8 @@
/* Check if caps have changed */
if (G_UNLIKELY (codec_tag != demux->video_codec_tag || demux->got_par)) {
-
GST_DEBUG_OBJECT (demux, "video settings have changed, changing caps");
+ gst_buffer_replace (&demux->video_codec_data, NULL);
if (!gst_flv_demux_video_negotiate (demux, codec_tag)) {
ret = GST_FLOW_ERROR;
@@ -1493,6 +1544,12 @@
}
case 1:
/* H.264 NALU packet */
+ if (!demux->video_codec_data) {
+ GST_ERROR_OBJECT (demux, "got H.264 video packet before codec data");
+ ret = GST_FLOW_OK;
+ gst_buffer_unref (outbuf);
+ goto beach;
+ }
GST_LOG_OBJECT (demux, "got a H.264 NALU video packet");
break;
default:
@@ -1595,7 +1652,8 @@
goto beach;
}
- ret = gst_flow_combiner_update_flow (demux->flowcombiner, ret);
+ ret = gst_flow_combiner_update_pad_flow (demux->flowcombiner,
+ demux->video_pad, ret);
beach:
gst_buffer_unmap (buffer, &map);
@@ -2353,7 +2411,6 @@
gst_object_unref (index);
}
-
done:
return ret;
}
@@ -2706,6 +2763,7 @@
GST_WARNING_OBJECT (demux, "upstream seek failed");
}
+ gst_flow_combiner_reset (demux->flowcombiner);
/* Tell all the stream we moved to a different position (discont) */
demux->audio_need_discont = TRUE;
demux->video_need_discont = TRUE;
@@ -2917,6 +2975,7 @@
demux->segment.format, demux->segment.position));
}
+ gst_flow_combiner_reset (demux->flowcombiner);
/* Tell all the stream a new segment is needed */
demux->audio_need_segment = TRUE;
demux->video_need_segment = TRUE;
@@ -3107,6 +3166,7 @@
demux->new_seg_event = NULL;
}
}
+ gst_flow_combiner_reset (demux->flowcombiner);
break;
}
default:
@@ -3130,9 +3190,12 @@
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:
/* Try to push upstream first */
+ gst_event_ref (event);
ret = gst_pad_push_event (demux->sinkpad, event);
- if (ret)
+ if (ret) {
+ gst_event_unref (event);
break;
+ }
if (demux->random_access) {
ret = gst_flv_demux_handle_seek_pull (demux, event, TRUE);
} else {
diff --git a/gst/flv/gstindex.c b/gst/flv/gstindex.c
index d56f3cc..26d7fa3 100644
--- a/gst/flv/gstindex.c
+++ b/gst/flv/gstindex.c
@@ -70,8 +70,8 @@
enum
{
- ARG_0,
- ARG_RESOLVER
+ PROP_0,
+ PROP_RESOLVER
/* FILL ME */
};
@@ -178,7 +178,7 @@
gobject_class->get_property = gst_index_get_property;
gobject_class->finalize = gst_index_finalize;
- g_object_class_install_property (gobject_class, ARG_RESOLVER,
+ g_object_class_install_property (gobject_class, PROP_RESOLVER,
g_param_spec_enum ("resolver", "Resolver",
"Select a predefined object to string mapper",
GST_TYPE_INDEX_RESOLVER, GST_INDEX_RESOLVER_PATH,
@@ -250,7 +250,7 @@
index = GST_INDEX (object);
switch (prop_id) {
- case ARG_RESOLVER:
+ case PROP_RESOLVER:
index->method = (GstIndexResolverMethod) g_value_get_enum (value);
index->resolver = resolvers[index->method].resolver;
index->resolver_user_data = resolvers[index->method].user_data;
@@ -270,7 +270,7 @@
index = GST_INDEX (object);
switch (prop_id) {
- case ARG_RESOLVER:
+ case PROP_RESOLVER:
g_value_set_enum (value, index->method);
break;
default:
diff --git a/gst/flx/Makefile.am b/gst/flx/Makefile.am
index 3aa4eb0..aaecaa0 100644
--- a/gst/flx/Makefile.am
+++ b/gst/flx/Makefile.am
@@ -7,18 +7,3 @@
libgstflxdec_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
noinst_HEADERS = flx_fmt.h flx_color.h gstflxdec.h
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstflxdec -:SHARED libgstflxdec \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstflxdec_la_SOURCES) \
- $(nodist_libgstflxdec_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstflxdec_la_CFLAGS) \
- -:LDFLAGS $(libgstflxdec_la_LDFLAGS) \
- $(libgstflxdec_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/flx/Makefile.in b/gst/flx/Makefile.in
index 2b40f7d..5137bfa 100644
--- a/gst/flx/Makefile.in
+++ b/gst/flx/Makefile.in
@@ -891,21 +891,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstflxdec -:SHARED libgstflxdec \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstflxdec_la_SOURCES) \
- $(nodist_libgstflxdec_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstflxdec_la_CFLAGS) \
- -:LDFLAGS $(libgstflxdec_la_LDFLAGS) \
- $(libgstflxdec_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/goom/Makefile.am b/gst/goom/Makefile.am
index 7baa1ed..a0e6424 100644
--- a/gst/goom/Makefile.am
+++ b/gst/goom/Makefile.am
@@ -45,17 +45,3 @@
EXTRA_DIST = $(PPC_FILES) $(MMX_FILES)
.NOTPARALLEL:
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstgoom -:SHARED libgstgoom \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstgoom_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstgoom_la_CFLAGS) \
- -:LDFLAGS $(libgstgoom_la_LDFLAGS) \
- $(libgstgoom_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/goom/Makefile.in b/gst/goom/Makefile.in
index 3bc56c4..80d1275 100644
--- a/gst/goom/Makefile.in
+++ b/gst/goom/Makefile.in
@@ -1076,20 +1076,6 @@
.NOTPARALLEL:
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstgoom -:SHARED libgstgoom \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstgoom_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstgoom_la_CFLAGS) \
- -:LDFLAGS $(libgstgoom_la_LDFLAGS) \
- $(libgstgoom_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/goom/gstgoom.c b/gst/goom/gstgoom.c
index b22531a..02715f9 100644
--- a/gst/goom/gstgoom.c
+++ b/gst/goom/gstgoom.c
@@ -66,7 +66,7 @@
enum
{
- ARG_0
+ PROP_0
/* FILL ME */
};
diff --git a/gst/goom2k1/Makefile.am b/gst/goom2k1/Makefile.am
index b32753f..dc92766 100644
--- a/gst/goom2k1/Makefile.am
+++ b/gst/goom2k1/Makefile.am
@@ -27,17 +27,3 @@
libgstgoom2k1_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
EXTRA_DIST = filters.c
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstgoom2k1 -:SHARED libgstgoom2k1 \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstgoom2k1_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstgoom2k1_la_CFLAGS) \
- -:LDFLAGS $(libgstgoom2k1_la_LDFLAGS) \
- $(libgstgoom2k1_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/goom2k1/Makefile.in b/gst/goom2k1/Makefile.in
index 9e96360..f8d72c4 100644
--- a/gst/goom2k1/Makefile.in
+++ b/gst/goom2k1/Makefile.in
@@ -934,20 +934,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstgoom2k1 -:SHARED libgstgoom2k1 \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstgoom2k1_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstgoom2k1_la_CFLAGS) \
- -:LDFLAGS $(libgstgoom2k1_la_LDFLAGS) \
- $(libgstgoom2k1_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/goom2k1/gstgoom.c b/gst/goom2k1/gstgoom.c
index 447352c..7f37b32 100644
--- a/gst/goom2k1/gstgoom.c
+++ b/gst/goom2k1/gstgoom.c
@@ -62,7 +62,7 @@
enum
{
- ARG_0
+ PROP_0
/* FILL ME */
};
diff --git a/gst/icydemux/Makefile.am b/gst/icydemux/Makefile.am
index 98af4ed..c2ed649 100644
--- a/gst/icydemux/Makefile.am
+++ b/gst/icydemux/Makefile.am
@@ -7,17 +7,3 @@
libgsticydemux_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
noinst_HEADERS = gsticydemux.h
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgsticydemux -:SHARED libgsticydemux \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgsticydemux_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgsticydemux_la_CFLAGS) \
- -:LDFLAGS $(libgsticydemux_la_LDFLAGS) \
- $(libgsticydemux_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/icydemux/Makefile.in b/gst/icydemux/Makefile.in
index d851cbd..9cf7501 100644
--- a/gst/icydemux/Makefile.in
+++ b/gst/icydemux/Makefile.in
@@ -882,20 +882,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgsticydemux -:SHARED libgsticydemux \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgsticydemux_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgsticydemux_la_CFLAGS) \
- -:LDFLAGS $(libgsticydemux_la_LDFLAGS) \
- $(libgsticydemux_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/icydemux/gsticydemux.c b/gst/icydemux/gsticydemux.c
index cdcb5ed..dade21b 100644
--- a/gst/icydemux/gsticydemux.c
+++ b/gst/icydemux/gsticydemux.c
@@ -551,6 +551,7 @@
if (icydemux->meta_interval == 0) {
ret = gst_icydemux_typefind_or_forward (icydemux, buf);
+ buf = NULL;
goto done;
}
diff --git a/gst/id3demux/Makefile.am b/gst/id3demux/Makefile.am
index 0c67df4..e4794e5 100644
--- a/gst/id3demux/Makefile.am
+++ b/gst/id3demux/Makefile.am
@@ -8,17 +8,3 @@
libgstid3demux_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
noinst_HEADERS = gstid3demux.h
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstid3demux -:SHARED libgstid3demux \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstid3demux_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(CPPFLAGS) $(libgstid3demux_la_CFLAGS) \
- -:LDFLAGS $(libgstid3demux_la_LDFLAGS) \
- $(libgstid3demux_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/id3demux/Makefile.in b/gst/id3demux/Makefile.in
index 1452d90..63abc19 100644
--- a/gst/id3demux/Makefile.in
+++ b/gst/id3demux/Makefile.in
@@ -884,20 +884,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstid3demux -:SHARED libgstid3demux \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstid3demux_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(CPPFLAGS) $(libgstid3demux_la_CFLAGS) \
- -:LDFLAGS $(libgstid3demux_la_LDFLAGS) \
- $(libgstid3demux_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/id3demux/gstid3demux.c b/gst/id3demux/gstid3demux.c
index a55832a..c67a6ff 100644
--- a/gst/id3demux/gstid3demux.c
+++ b/gst/id3demux/gstid3demux.c
@@ -58,8 +58,8 @@
enum
{
- ARG_0,
- ARG_PREFER_V1
+ PROP_0,
+ PROP_PREFER_V1
};
#define DEFAULT_PREFER_V1 FALSE
@@ -102,7 +102,7 @@
gobject_class->set_property = gst_id3demux_set_property;
gobject_class->get_property = gst_id3demux_get_property;
- g_object_class_install_property (gobject_class, ARG_PREFER_V1,
+ g_object_class_install_property (gobject_class, PROP_PREFER_V1,
g_param_spec_boolean ("prefer-v1", "Prefer version 1 tag",
"Prefer tags from ID3v1 tag at end of file when both ID3v1 "
"and ID3v2 tags are present", DEFAULT_PREFER_V1,
@@ -239,7 +239,7 @@
id3demux = GST_ID3DEMUX (object);
switch (prop_id) {
- case ARG_PREFER_V1:{
+ case PROP_PREFER_V1:{
GST_OBJECT_LOCK (id3demux);
id3demux->prefer_v1 = g_value_get_boolean (value);
GST_OBJECT_UNLOCK (id3demux);
@@ -260,7 +260,7 @@
id3demux = GST_ID3DEMUX (object);
switch (prop_id) {
- case ARG_PREFER_V1:
+ case PROP_PREFER_V1:
GST_OBJECT_LOCK (id3demux);
g_value_set_boolean (value, id3demux->prefer_v1);
GST_OBJECT_UNLOCK (id3demux);
diff --git a/gst/imagefreeze/Makefile.am b/gst/imagefreeze/Makefile.am
index e31288e..72615a5 100644
--- a/gst/imagefreeze/Makefile.am
+++ b/gst/imagefreeze/Makefile.am
@@ -8,17 +8,3 @@
libgstimagefreeze_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
noinst_HEADERS = gstimagefreeze.h
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstimagefreeze -:SHARED libgstimagefreeze \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstimagefreeze_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstimagefreeze_la_CFLAGS) \
- -:LDFLAGS $(libgstimagefreeze_la_LDFLAGS) \
- $(libgstimagefreeze_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/imagefreeze/Makefile.in b/gst/imagefreeze/Makefile.in
index d80c7ca..f4e92ba 100644
--- a/gst/imagefreeze/Makefile.in
+++ b/gst/imagefreeze/Makefile.in
@@ -882,20 +882,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstimagefreeze -:SHARED libgstimagefreeze \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstimagefreeze_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstimagefreeze_la_CFLAGS) \
- -:LDFLAGS $(libgstimagefreeze_la_LDFLAGS) \
- $(libgstimagefreeze_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/interleave/Makefile.am b/gst/interleave/Makefile.am
index f27de70..dddaefa 100644
--- a/gst/interleave/Makefile.am
+++ b/gst/interleave/Makefile.am
@@ -8,17 +8,3 @@
libgstinterleave_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
noinst_HEADERS = plugin.h interleave.h deinterleave.h
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstinterleave -:SHARED libgstinterleave \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstinterleave_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstinterleave_la_CFLAGS) \
- -:LDFLAGS $(libgstinterleave_la_LDFLAGS) \
- $(libgstinterleave_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/interleave/Makefile.in b/gst/interleave/Makefile.in
index b821aef..fcb8719 100644
--- a/gst/interleave/Makefile.in
+++ b/gst/interleave/Makefile.in
@@ -900,20 +900,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstinterleave -:SHARED libgstinterleave \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstinterleave_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstinterleave_la_CFLAGS) \
- -:LDFLAGS $(libgstinterleave_la_LDFLAGS) \
- $(libgstinterleave_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/isomp4/Makefile.am b/gst/isomp4/Makefile.am
index 30ef999..8cfdd15 100644
--- a/gst/isomp4/Makefile.am
+++ b/gst/isomp4/Makefile.am
@@ -39,17 +39,3 @@
EXTRA_DIST = \
gstqtmux-doc.c \
gstqtmux-doc.h
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstisomp4 -:SHARED libgstisomp4 \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstisomp4_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(CPPFLAGS) $(libgstisomp4_la_CFLAGS) \
- -:LDFLAGS $(libgstisomp4_la_LDFLAGS) \
- $(libgstisomp4_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/isomp4/Makefile.in b/gst/isomp4/Makefile.in
index 1ae0dd7..5dbfcdf 100644
--- a/gst/isomp4/Makefile.in
+++ b/gst/isomp4/Makefile.in
@@ -1029,20 +1029,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstisomp4 -:SHARED libgstisomp4 \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstisomp4_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(CPPFLAGS) $(libgstisomp4_la_CFLAGS) \
- -:LDFLAGS $(libgstisomp4_la_LDFLAGS) \
- $(libgstisomp4_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/isomp4/gstqtmux.c b/gst/isomp4/gstqtmux.c
index e61ec6c..14e699b 100644
--- a/gst/isomp4/gstqtmux.c
+++ b/gst/isomp4/gstqtmux.c
@@ -127,6 +127,58 @@
GST_DEBUG_CATEGORY_STATIC (gst_qt_mux_debug);
#define GST_CAT_DEFAULT gst_qt_mux_debug
+/* Hacker notes.
+ *
+ * The basic building blocks of MP4 files are:
+ * - an 'ftyp' box at the very start
+ * - an 'mdat' box which contains the raw audio/video/subtitle data;
+ * this is just a bunch of bytes, completely unframed and possibly
+ * unordered with no additional meta-information
+ * - a 'moov' box that contains information about the different streams
+ * and what they contain, as well as sample tables for each stream
+ * that tell the demuxer where in the mdat box each buffer/sample is
+ * and what its duration/timestamp etc. is, and whether it's a
+ * keyframe etc.
+ * Additionally, fragmented MP4 works by writing chunks of data in
+ * pairs of 'moof' and 'mdat' boxes:
+ * - 'moof' boxes, header preceding each mdat fragment describing the
+ * contents, like a moov but only for that fragment.
+ * - a 'mfra' box for Fragmented MP4, which is written at the end and
+ * contains a summary of all fragments and seek tables.
+ *
+ * Currently mp4mux can work in 3 different modes / generate 3 types
+ * of output files/streams:
+ *
+ * - Normal mp4: mp4mux will write a little ftyp identifier at the
+ * beginning, then start an mdat box into which it will write all the
+ * sample data. At EOS it will then write the moov header with track
+ * headers and sample tables at the end of the file, and rewrite the
+ * start of the file to fix up the mdat box size at the beginning.
+ * It has to wait for EOS to write the moov (which includes the
+ * sample tables) because it doesn't know how much space those
+ * tables will be. The output downstream must be seekable to rewrite
+ * the mdat box at EOS.
+ *
+ * - Fragmented mp4: moov header with track headers at start
+ * but no sample table), followed by N fragments, each containing
+ * track headers with sample tables followed by some data. Downstream
+ * does not need to be seekable if the 'streamable' flag is TRUE,
+ * as the final mfra and total duration will be omitted.
+ *
+ * - Fast-start mp4: the goal here is to create a file where the moov
+ * headers are at the beginning; what mp4mux will do is write all
+ * sample data into a temp file and build moov header plus sample
+ * tables in memory and then when EOS comes, it will push out the
+ * moov header plus sample tables at the beginning, followed by the
+ * mdat sample data at the end which is read in from the temp file
+ * Files created in this mode are better for streaming over the
+ * network, since the client doesn't have to seek to the end of the
+ * file to get the headers, but it requires copying all sample data
+ * out of the temp file at EOS, which can be expensive. Downstream does
+ * not need to be seekable, because of the use of the temp file.
+ *
+ */
+
#ifndef GST_REMOVE_DEPRECATED
enum
{
@@ -1691,6 +1743,75 @@
}
}
+static gboolean
+gst_qt_mux_downstream_is_seekable (GstQTMux * qtmux)
+{
+ gboolean seekable = FALSE;
+ GstQuery *query = gst_query_new_seeking (GST_FORMAT_BYTES);
+
+ if (gst_pad_peer_query (qtmux->srcpad, query)) {
+ gst_query_parse_seeking (query, NULL, &seekable, NULL, NULL);
+ GST_INFO_OBJECT (qtmux, "downstream is %sseekable", seekable ? "" : "not ");
+ } else {
+ /* have to assume seeking is not supported if query not handled downstream */
+ GST_WARNING_OBJECT (qtmux, "downstream did not handle seeking query");
+ seekable = FALSE;
+ }
+ gst_query_unref (query);
+
+ return seekable;
+}
+
+static void
+gst_qt_mux_prepare_moov_recovery (GstQTMux * qtmux)
+{
+ GSList *walk;
+ gboolean fail = FALSE;
+ AtomFTYP *ftyp = NULL;
+ GstBuffer *prefix = NULL;
+
+ GST_DEBUG_OBJECT (qtmux, "Openning moov recovery file: %s",
+ qtmux->moov_recov_file_path);
+
+ qtmux->moov_recov_file = g_fopen (qtmux->moov_recov_file_path, "wb+");
+ if (qtmux->moov_recov_file == NULL) {
+ GST_WARNING_OBJECT (qtmux, "Failed to open moov recovery file in %s",
+ qtmux->moov_recov_file_path);
+ return;
+ }
+
+ gst_qt_mux_prepare_ftyp (qtmux, &ftyp, &prefix);
+
+ if (!atoms_recov_write_headers (qtmux->moov_recov_file, ftyp, prefix,
+ qtmux->moov, qtmux->timescale, g_slist_length (qtmux->sinkpads))) {
+ GST_WARNING_OBJECT (qtmux, "Failed to write moov recovery file " "headers");
+ goto fail;
+ }
+
+ atom_ftyp_free (ftyp);
+ if (prefix)
+ gst_buffer_unref (prefix);
+
+ for (walk = qtmux->sinkpads; walk && !fail; walk = g_slist_next (walk)) {
+ GstCollectData *cdata = (GstCollectData *) walk->data;
+ GstQTPad *qpad = (GstQTPad *) cdata;
+ /* write info for each stream */
+ fail = atoms_recov_write_trak_info (qtmux->moov_recov_file, qpad->trak);
+ if (fail) {
+ GST_WARNING_OBJECT (qtmux, "Failed to write trak info to recovery "
+ "file");
+ break;
+ }
+ }
+
+fail:
+ /* cleanup */
+ fclose (qtmux->moov_recov_file);
+ qtmux->moov_recov_file = NULL;
+ GST_WARNING_OBJECT (qtmux, "An error was detected while writing to "
+ "recover file, moov recovery won't work");
+}
+
static GstFlowReturn
gst_qt_mux_start_file (GstQTMux * qtmux)
{
@@ -1713,39 +1834,49 @@
gst_pad_set_caps (qtmux->srcpad, caps);
gst_caps_unref (caps);
- /* if not streaming, check if downstream is seekable */
- if (!qtmux->streamable) {
- gboolean seekable;
- GstQuery *query;
+ /* Default is 'normal' mode */
+ qtmux->mux_mode = GST_QT_MUX_MODE_MOOV_AT_END;
- query = gst_query_new_seeking (GST_FORMAT_BYTES);
- if (gst_pad_peer_query (qtmux->srcpad, query)) {
- gst_query_parse_seeking (query, NULL, &seekable, NULL, NULL);
- GST_INFO_OBJECT (qtmux, "downstream is %sseekable",
- seekable ? "" : "not ");
- } else {
- /* have to assume seeking is supported if query not handled downstream */
- GST_WARNING_OBJECT (qtmux, "downstream did not handle seeking query");
- seekable = FALSE;
- }
- gst_query_unref (query);
- if (!seekable) {
- if (qtmux_klass->format != GST_QT_MUX_FORMAT_ISML) {
- if (!qtmux->fast_start) {
- GST_ELEMENT_WARNING (qtmux, STREAM, FAILED,
- ("Downstream is not seekable and headers can't be rewritten"),
- (NULL));
- /* FIXME: Is there something better we can do? */
- qtmux->streamable = TRUE;
- }
- } else {
+ /* Require a sensible fragment duration when muxing
+ * using the ISML muxer */
+ if (qtmux_klass->format == GST_QT_MUX_FORMAT_ISML &&
+ qtmux->fragment_duration == 0)
+ goto invalid_isml;
+
+ if (qtmux->fragment_duration > 0) {
+ if (qtmux->streamable)
+ qtmux->mux_mode = GST_QT_MUX_MODE_FRAGMENTED_STREAMABLE;
+ else
+ qtmux->mux_mode = GST_QT_MUX_MODE_FRAGMENTED;
+ } else if (qtmux->fast_start) {
+ qtmux->mux_mode = GST_QT_MUX_MODE_FAST_START;
+ }
+
+ switch (qtmux->mux_mode) {
+ case GST_QT_MUX_MODE_MOOV_AT_END:
+ /* We have to be able to seek to rewrite the mdat header, or any
+ * moov atom we write will not be visible in the file, because an
+ * MDAT with 0 as the size covers the rest of the file. A file
+ * with no moov is not playable, so error out now. */
+ if (!gst_qt_mux_downstream_is_seekable (qtmux)) {
+ GST_ELEMENT_ERROR (qtmux, STREAM, MUX,
+ ("Downstream is not seekable - will not be able to create a playable file"),
+ (NULL));
+ return GST_FLOW_ERROR;
+ }
+ break;
+ case GST_QT_MUX_MODE_FAST_START:
+ case GST_QT_MUX_MODE_FRAGMENTED_STREAMABLE:
+ break; /* Don't need seekability, ignore */
+ case GST_QT_MUX_MODE_FRAGMENTED:
+ if (!gst_qt_mux_downstream_is_seekable (qtmux)) {
GST_WARNING_OBJECT (qtmux, "downstream is not seekable, but "
"streamable=false. Will ignore that and create streamable output "
"instead");
qtmux->streamable = TRUE;
g_object_notify (G_OBJECT (qtmux), "streamable");
}
- }
+ break;
}
/* let downstream know we think in BYTES and expect to do seeking later on */
@@ -1755,81 +1886,44 @@
/* initialize our moov recovery file */
GST_OBJECT_LOCK (qtmux);
if (qtmux->moov_recov_file_path) {
- GST_DEBUG_OBJECT (qtmux, "Openning moov recovery file: %s",
- qtmux->moov_recov_file_path);
- qtmux->moov_recov_file = g_fopen (qtmux->moov_recov_file_path, "wb+");
- if (qtmux->moov_recov_file == NULL) {
- GST_WARNING_OBJECT (qtmux, "Failed to open moov recovery file in %s",
- qtmux->moov_recov_file_path);
- } else {
- GSList *walk;
- gboolean fail = FALSE;
- AtomFTYP *ftyp = NULL;
- GstBuffer *prefix = NULL;
-
- gst_qt_mux_prepare_ftyp (qtmux, &ftyp, &prefix);
-
- if (!atoms_recov_write_headers (qtmux->moov_recov_file, ftyp, prefix,
- qtmux->moov, qtmux->timescale,
- g_slist_length (qtmux->sinkpads))) {
- GST_WARNING_OBJECT (qtmux, "Failed to write moov recovery file "
- "headers");
- fail = TRUE;
- }
-
- atom_ftyp_free (ftyp);
- if (prefix)
- gst_buffer_unref (prefix);
-
- for (walk = qtmux->sinkpads; walk && !fail; walk = g_slist_next (walk)) {
- GstCollectData *cdata = (GstCollectData *) walk->data;
- GstQTPad *qpad = (GstQTPad *) cdata;
- /* write info for each stream */
- fail = atoms_recov_write_trak_info (qtmux->moov_recov_file, qpad->trak);
- if (fail) {
- GST_WARNING_OBJECT (qtmux, "Failed to write trak info to recovery "
- "file");
- }
- }
- if (fail) {
- /* cleanup */
- fclose (qtmux->moov_recov_file);
- qtmux->moov_recov_file = NULL;
- GST_WARNING_OBJECT (qtmux, "An error was detected while writing to "
- "recover file, moov recovery won't work");
- }
- }
+ gst_qt_mux_prepare_moov_recovery (qtmux);
}
GST_OBJECT_UNLOCK (qtmux);
- /*
+ /*
* send mdat header if already needed, and mark position for later update.
* We don't send ftyp now if we are on fast start mode, because we can
* better fine tune using the information we gather to create the whole moov
* atom.
*/
- if (qtmux->fast_start) {
- GST_OBJECT_LOCK (qtmux);
- qtmux->fast_start_file = g_fopen (qtmux->fast_start_file_path, "wb+");
- if (!qtmux->fast_start_file)
- goto open_failed;
- GST_OBJECT_UNLOCK (qtmux);
+ switch (qtmux->mux_mode) {
+ case GST_QT_MUX_MODE_MOOV_AT_END:
+ ret = gst_qt_mux_prepare_and_send_ftyp (qtmux);
+ if (ret != GST_FLOW_OK)
+ break;
- /* send a dummy buffer for preroll */
- ret = gst_qt_mux_send_buffer (qtmux, gst_buffer_new (), NULL, FALSE);
- if (ret != GST_FLOW_OK)
- goto exit;
-
- } else {
- ret = gst_qt_mux_prepare_and_send_ftyp (qtmux);
- if (ret != GST_FLOW_OK) {
- goto exit;
- }
-
- /* well, it's moov pos if fragmented ... */
- qtmux->mdat_pos = qtmux->header_size;
-
- if (qtmux->fragment_duration) {
+ /* store the mdat position for rewriting later ... */
+ qtmux->mdat_pos = qtmux->header_size;
+ /* extended atom in case we go over 4GB while writing and need
+ * the full 64-bit atom */
+ ret = gst_qt_mux_send_mdat_header (qtmux, &qtmux->header_size, 0, TRUE);
+ break;
+ case GST_QT_MUX_MODE_FAST_START:
+ GST_OBJECT_LOCK (qtmux);
+ qtmux->fast_start_file = g_fopen (qtmux->fast_start_file_path, "wb+");
+ if (!qtmux->fast_start_file)
+ goto open_failed;
+ GST_OBJECT_UNLOCK (qtmux);
+ /* send a dummy buffer for preroll */
+ ret = gst_qt_mux_send_buffer (qtmux, gst_buffer_new (), NULL, FALSE);
+ break;
+ case GST_QT_MUX_MODE_FRAGMENTED:
+ case GST_QT_MUX_MODE_FRAGMENTED_STREAMABLE:
+ ret = gst_qt_mux_prepare_and_send_ftyp (qtmux);
+ if (ret != GST_FLOW_OK)
+ break;
+ /* well, it's moov pos if fragmented ... */
+ qtmux->mdat_pos = qtmux->header_size;
GST_DEBUG_OBJECT (qtmux, "fragment duration %d ms, writing headers",
qtmux->fragment_duration);
/* also used as snapshot marker to indicate fragmented file */
@@ -1844,25 +1938,27 @@
ret =
gst_qt_mux_send_extra_atoms (qtmux, TRUE, &qtmux->header_size, FALSE);
if (ret != GST_FLOW_OK)
- return ret;
- /* prepare index */
- if (!qtmux->streamable)
+ break;
+ /* prepare index if not streamable */
+ if (qtmux->mux_mode == GST_QT_MUX_MODE_FRAGMENTED)
qtmux->mfra = atom_mfra_new (qtmux->context);
- } else {
- /* extended to ensure some spare space */
- ret = gst_qt_mux_send_mdat_header (qtmux, &qtmux->header_size, 0, TRUE);
- }
+ break;
}
-exit:
return ret;
-
/* ERRORS */
+invalid_isml:
+ {
+ GST_ELEMENT_ERROR (qtmux, STREAM, MUX,
+ ("Cannot create an ISML file with 0 fragment duration"), (NULL));
+ return GST_FLOW_ERROR;
+ }
+
open_failed:
{
GST_ELEMENT_ERROR (qtmux, RESOURCE, OPEN_READ_WRITE,
- (("Could not open temporary file \"%s\""), qtmux->fast_start_file_path),
- GST_ERROR_SYSTEM);
+ (("Could not open temporary file \"%s\""),
+ qtmux->fast_start_file_path), GST_ERROR_SYSTEM);
GST_OBJECT_UNLOCK (qtmux);
return GST_FLOW_ERROR;
}
@@ -1959,10 +2055,16 @@
}
}
- if (qtmux->fragment_sequence) {
- GstSegment segment;
-
- if (qtmux->mfra) {
+ switch (qtmux->mux_mode) {
+ case GST_QT_MUX_MODE_FRAGMENTED_STREAMABLE:
+ {
+ /* Streamable mode; no need to write duration or MFRA */
+ GST_DEBUG_OBJECT (qtmux, "streamable file; nothing to stop");
+ return GST_FLOW_OK;
+ }
+ case GST_QT_MUX_MODE_FRAGMENTED:
+ {
+ GstSegment segment;
guint8 *data = NULL;
GstBuffer *buf;
@@ -1974,31 +2076,29 @@
ret = gst_qt_mux_send_buffer (qtmux, buf, NULL, FALSE);
if (ret != GST_FLOW_OK)
return ret;
- } else {
- /* must have been streamable; no need to write duration */
- GST_DEBUG_OBJECT (qtmux, "streamable file; nothing to stop");
- return GST_FLOW_OK;
- }
- timescale = qtmux->timescale;
- /* only mvex duration is updated,
- * mvhd should be consistent with empty moov
- * (but TODO maybe some clients do not handle that well ?) */
- qtmux->moov->mvex.mehd.fragment_duration =
- gst_util_uint64_scale (first_ts, timescale, GST_SECOND);
- GST_DEBUG_OBJECT (qtmux, "rewriting moov with mvex duration %"
- GST_TIME_FORMAT, GST_TIME_ARGS (first_ts));
- /* seek and rewrite the header */
- gst_segment_init (&segment, GST_FORMAT_BYTES);
- segment.start = qtmux->mdat_pos;
- gst_pad_push_event (qtmux->srcpad, gst_event_new_segment (&segment));
- /* no need to seek back */
- return gst_qt_mux_send_moov (qtmux, NULL, FALSE);
+ timescale = qtmux->timescale;
+ /* only mvex duration is updated,
+ * mvhd should be consistent with empty moov
+ * (but TODO maybe some clients do not handle that well ?) */
+ qtmux->moov->mvex.mehd.fragment_duration =
+ gst_util_uint64_scale (first_ts, timescale, GST_SECOND);
+ GST_DEBUG_OBJECT (qtmux, "rewriting moov with mvex duration %"
+ GST_TIME_FORMAT, GST_TIME_ARGS (first_ts));
+ /* seek and rewrite the header */
+ gst_segment_init (&segment, GST_FORMAT_BYTES);
+ segment.start = qtmux->mdat_pos;
+ gst_pad_push_event (qtmux->srcpad, gst_event_new_segment (&segment));
+ /* no need to seek back */
+ return gst_qt_mux_send_moov (qtmux, NULL, FALSE);
+ }
+ default:
+ break;
}
+ /* Moov-at-end or fast-start mode from here down */
gst_qt_mux_configure_moov (qtmux, ×cale);
-
- /* check for late streams */
+ /* check for late streams. First, find the earliest start time */
first_ts = GST_CLOCK_TIME_NONE;
for (walk = qtmux->collect->data; walk; walk = g_slist_next (walk)) {
GstCollectData *cdata = (GstCollectData *) walk->data;
@@ -2046,12 +2146,12 @@
/* tags into file metadata */
gst_qt_mux_setup_metadata (qtmux);
-
large_file = (qtmux->mdat_size > MDAT_LARGE_FILE_LIMIT);
+
/* if faststart, update the offset of the atoms in the movie with the offset
* that the movie headers before mdat will cause.
* Also, send the ftyp */
- if (qtmux->fast_start_file) {
+ if (qtmux->mux_mode == GST_QT_MUX_MODE_FAST_START) {
GstFlowReturn flow_ret;
offset = size = 0;
@@ -2073,9 +2173,12 @@
} else {
offset = qtmux->header_size;
}
+
+ /* Now that we know the size of moov + extra atoms, we can adjust
+ * the chunk offsets stored into the moov */
atom_moov_chunks_add_offset (qtmux->moov, offset);
- /* moov */
+ /* write out moov and extra atoms */
/* note: as of this point, we no longer care about tracking written data size,
* since there is no more use for it anyway */
ret = gst_qt_mux_send_moov (qtmux, NULL, FALSE);
@@ -2087,23 +2190,32 @@
if (ret != GST_FLOW_OK)
return ret;
- /* if needed, send mdat atom and move buffered data into it */
- if (qtmux->fast_start_file) {
- /* mdat_size = accumulated (buffered data) */
- ret = gst_qt_mux_send_mdat_header (qtmux, NULL, qtmux->mdat_size,
- large_file);
- if (ret != GST_FLOW_OK)
- return ret;
- ret = gst_qt_mux_send_buffered_data (qtmux, NULL);
- if (ret != GST_FLOW_OK)
- return ret;
- } else if (!qtmux->streamable) {
- /* mdat needs update iff not using faststart */
- GST_DEBUG_OBJECT (qtmux, "updating mdat size");
- ret = gst_qt_mux_update_mdat_size (qtmux, qtmux->mdat_pos,
- qtmux->mdat_size, NULL);
- /* note; no seeking back to the end of file is done,
- * since we no longer write anything anyway */
+ switch (qtmux->mux_mode) {
+ case GST_QT_MUX_MODE_MOOV_AT_END:
+ {
+ /* mdat needs update iff not using faststart */
+ GST_DEBUG_OBJECT (qtmux, "updating mdat size");
+ ret = gst_qt_mux_update_mdat_size (qtmux, qtmux->mdat_pos,
+ qtmux->mdat_size, NULL);
+ /* note; no seeking back to the end of file is done,
+ * since we no longer write anything anyway */
+ break;
+ }
+ case GST_QT_MUX_MODE_FAST_START:
+ {
+ /* send mdat atom and move buffered data into it */
+ /* mdat_size = accumulated (buffered data) */
+ ret = gst_qt_mux_send_mdat_header (qtmux, NULL, qtmux->mdat_size,
+ large_file);
+ if (ret != GST_FLOW_OK)
+ return ret;
+ ret = gst_qt_mux_send_buffered_data (qtmux, NULL);
+ if (ret != GST_FLOW_OK)
+ return ret;
+ break;
+ }
+ default:
+ g_assert_not_reached ();
}
return ret;
@@ -2218,21 +2330,6 @@
return ret;
}
-static void
-check_and_subtract_ts (GstQTMux * qtmux, GstClockTime * ts_a, GstClockTime ts_b)
-{
- if (G_LIKELY (GST_CLOCK_TIME_IS_VALID (*ts_a))) {
- if (G_LIKELY (*ts_a >= ts_b)) {
- *ts_a -= ts_b;
- } else {
- *ts_a = 0;
- GST_WARNING_OBJECT (qtmux, "Subtraction would result in negative value, "
- "using 0 as result");
- }
- }
-}
-
-
static GstFlowReturn
gst_qt_mux_register_and_push_sample (GstQTMux * qtmux, GstQTPad * pad,
GstBuffer * buffer, gboolean is_last_buffer, guint nsamples,
@@ -2253,15 +2350,21 @@
}
}
- if (qtmux->fragment_sequence) {
- /* ensure that always sync samples are marked as such */
- ret = gst_qt_mux_pad_fragment_add_buffer (qtmux, pad, buffer,
- is_last_buffer, nsamples, last_dts, (gint32) scaled_duration,
- sample_size, !pad->sync || sync, pts_offset);
- } else {
- atom_trak_add_samples (pad->trak, nsamples, (gint32) scaled_duration,
- sample_size, chunk_offset, sync, pts_offset);
- ret = gst_qt_mux_send_buffer (qtmux, buffer, &qtmux->mdat_size, TRUE);
+ switch (qtmux->mux_mode) {
+ case GST_QT_MUX_MODE_MOOV_AT_END:
+ case GST_QT_MUX_MODE_FAST_START:
+
+ atom_trak_add_samples (pad->trak, nsamples, (gint32) scaled_duration,
+ sample_size, chunk_offset, sync, pts_offset);
+ ret = gst_qt_mux_send_buffer (qtmux, buffer, &qtmux->mdat_size, TRUE);
+ break;
+ case GST_QT_MUX_MODE_FRAGMENTED:
+ case GST_QT_MUX_MODE_FRAGMENTED_STREAMABLE:
+ /* ensure that always sync samples are marked as such */
+ ret = gst_qt_mux_pad_fragment_add_buffer (qtmux, pad, buffer,
+ is_last_buffer, nsamples, last_dts, (gint32) scaled_duration,
+ sample_size, !pad->sync || sync, pts_offset);
+ break;
}
return ret;
@@ -2279,7 +2382,7 @@
guint64 chunk_offset;
gint64 last_dts, scaled_duration;
gint64 pts_offset = 0;
- gboolean sync = FALSE, do_pts = FALSE;
+ gboolean sync = FALSE;
GstFlowReturn ret = GST_FLOW_OK;
if (!pad->fourcc)
@@ -2290,13 +2393,6 @@
buf = pad->prepare_buf_func (pad, buf, qtmux);
}
- if (G_LIKELY (buf != NULL && GST_CLOCK_TIME_IS_VALID (pad->first_ts) &&
- pad->first_ts != 0)) {
- buf = gst_buffer_make_writable (buf);
- check_and_subtract_ts (qtmux, &GST_BUFFER_DTS (buf), pad->first_ts);
- check_and_subtract_ts (qtmux, &GST_BUFFER_PTS (buf), pad->first_ts);
- }
-
last_buf = pad->last_buf;
/* DTS delta is used to calculate sample duration.
@@ -2311,11 +2407,7 @@
GST_BUFFER_DURATION (last_buf) : 0;
buf = gst_buffer_make_writable (buf);
- GST_BUFFER_DTS (buf) =
- gst_segment_to_running_time (&pad->collect.segment, GST_FORMAT_TIME,
- pad->collect.segment.start);
- if (GST_CLOCK_TIME_IS_VALID (pad->first_ts))
- check_and_subtract_ts (qtmux, &GST_BUFFER_DTS (buf), pad->first_ts);
+ GST_BUFFER_DTS (buf) = 0; /* running-time 0 */
if (last_buf
&& (GST_BUFFER_DTS (last_buf) + last_buf_duration) >
@@ -2353,7 +2445,6 @@
/* if this is the first buffer, store the timestamp */
if (G_UNLIKELY (pad->first_ts == GST_CLOCK_TIME_NONE) && last_buf) {
if (GST_BUFFER_DTS_IS_VALID (last_buf)) {
- /* first pad always has DTS. If it was not provided by upstream it was set to segment start */
pad->first_ts = GST_BUFFER_DTS (last_buf);
} else if (GST_BUFFER_PTS_IS_VALID (last_buf)) {
pad->first_ts = GST_BUFFER_PTS (last_buf);
@@ -2361,23 +2452,14 @@
if (GST_CLOCK_TIME_IS_VALID (pad->first_ts)) {
GST_DEBUG ("setting first_ts to %" G_GUINT64_FORMAT, pad->first_ts);
- last_buf = gst_buffer_make_writable (last_buf);
- check_and_subtract_ts (qtmux, &GST_BUFFER_DTS (last_buf), pad->first_ts);
- check_and_subtract_ts (qtmux, &GST_BUFFER_PTS (last_buf), pad->first_ts);
- if (buf) {
- buf = gst_buffer_make_writable (buf);
- check_and_subtract_ts (qtmux, &GST_BUFFER_DTS (buf), pad->first_ts);
- check_and_subtract_ts (qtmux, &GST_BUFFER_PTS (buf), pad->first_ts);
- }
} else {
- GST_ERROR_OBJECT (qtmux, "First buffer for pad %s has no timestamp, "
+ GST_WARNING_OBJECT (qtmux, "First buffer for pad %s has no timestamp, "
"using 0 as first timestamp", GST_PAD_NAME (pad->collect.pad));
pad->first_ts = 0;
}
GST_DEBUG_OBJECT (qtmux, "Stored first timestamp for pad %s %"
GST_TIME_FORMAT, GST_PAD_NAME (pad->collect.pad),
GST_TIME_ARGS (pad->first_ts));
-
}
if (last_buf && buf && GST_CLOCK_TIME_IS_VALID (GST_BUFFER_DTS (buf)) &&
@@ -2488,7 +2570,6 @@
}
if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_DTS (last_buf))) {
- do_pts = TRUE;
last_dts = gst_util_uint64_scale_round (GST_BUFFER_DTS (last_buf),
atom_trak_get_timescale (pad->trak), GST_SECOND);
pts_offset =
@@ -2497,7 +2578,6 @@
} else {
pts_offset = 0;
- do_pts = TRUE;
last_dts = gst_util_uint64_scale_round (GST_BUFFER_PTS (last_buf),
atom_trak_get_timescale (pad->trak), GST_SECOND);
}
@@ -2521,7 +2601,7 @@
/* now we go and register this buffer/sample all over */
ret = gst_qt_mux_register_and_push_sample (qtmux, pad, last_buf,
buf == NULL, nsamples, last_dts, scaled_duration, sample_size,
- chunk_offset, sync, do_pts, pts_offset);
+ chunk_offset, sync, TRUE, pts_offset);
/* if this is sparse and we have a next buffer, check if there is any gap
* between them to insert an empty sample */
@@ -2544,7 +2624,7 @@
ret =
gst_qt_mux_register_and_push_sample (qtmux, pad, empty_buf, FALSE, 1,
last_dts + scaled_duration, empty_duration_scaled,
- gst_buffer_get_size (empty_buf), qtmux->mdat_size, sync, do_pts, 0);
+ gst_buffer_get_size (empty_buf), qtmux->mdat_size, sync, TRUE, 0);
} else {
/* our only case currently is tx3g subtitles, so there is no reason to fill this yet */
g_assert_not_reached ();
diff --git a/gst/isomp4/gstqtmux.h b/gst/isomp4/gstqtmux.h
index a9ea362..0e30eb4 100644
--- a/gst/isomp4/gstqtmux.h
+++ b/gst/isomp4/gstqtmux.h
@@ -144,6 +144,13 @@
GST_QT_MUX_STATE_EOS
} GstQTMuxState;
+typedef enum _GstQtMuxMode {
+ GST_QT_MUX_MODE_MOOV_AT_END,
+ GST_QT_MUX_MODE_FRAGMENTED,
+ GST_QT_MUX_MODE_FRAGMENTED_STREAMABLE,
+ GST_QT_MUX_MODE_FAST_START
+} GstQtMuxMode;
+
struct _GstQTMux
{
GstElement element;
@@ -155,7 +162,11 @@
/* state */
GstQTMuxState state;
- /* size of header (prefix, atoms (ftyp, mdat)) */
+ /* Mux mode, inferred from property
+ * set in gst_qt_mux_start_file() */
+ GstQtMuxMode mux_mode;
+
+ /* size of header (prefix, atoms (ftyp, possibly moov, mdat header)) */
guint64 header_size;
/* accumulated size of raw media data (a priori not including mdat header) */
guint64 mdat_size;
@@ -196,6 +207,9 @@
gchar *fast_start_file_path;
gchar *moov_recov_file_path;
guint32 fragment_duration;
+ /* Whether or not to work in 'streamable' mode and not
+ * seek to rewrite headers - only valid for fragmented
+ * mode. */
gboolean streamable;
/* for request pad naming */
diff --git a/gst/isomp4/gstrtpxqtdepay.c b/gst/isomp4/gstrtpxqtdepay.c
index 3bce461..9116ccc 100644
--- a/gst/isomp4/gstrtpxqtdepay.c
+++ b/gst/isomp4/gstrtpxqtdepay.c
@@ -73,7 +73,7 @@
enum
{
- ARG_0,
+ PROP_0,
};
static GstStaticPadTemplate gst_rtp_xqt_depay_src_template =
diff --git a/gst/isomp4/qtdemux.c b/gst/isomp4/qtdemux.c
index 153d031..f50ff20 100644
--- a/gst/isomp4/qtdemux.c
+++ b/gst/isomp4/qtdemux.c
@@ -36,7 +36,7 @@
* <refsect2>
* <title>Example launch line</title>
* |[
- * gst-launch-1.0 filesrc location=test.mov ! qtdemux name=demux demux.audio_0 ! decodebin ! audioconvert ! audioresample ! autoaudiosink demux.video_0 ! queue ! decodebin ! videoconvert ! videoscale ! autovideosink
+ * gst-launch-1.0 filesrc location=test.mov ! qtdemux name=demux demux.audio_0 ! queue ! decodebin ! audioconvert ! audioresample ! autoaudiosink demux.video_0 ! queue ! decodebin ! videoconvert ! videoscale ! autovideosink
* ]| Play (parse and decode) a .mov file and try to output it to
* an automatically detected soundcard and videosink. If the MOV file contains
* compressed audio or video data, this will only work if you have the
@@ -1416,6 +1416,7 @@
}
/* and set all streams to the final position */
+ gst_flow_combiner_reset (qtdemux->flowcombiner);
for (n = 0; n < qtdemux->n_streams; n++) {
QtDemuxStream *stream = qtdemux->streams[n];
@@ -1431,6 +1432,7 @@
}
segment->position = desired_offset;
segment->time = desired_offset;
+ segment->start = desired_offset;
/* we stop at the end */
if (segment->stop == -1)
@@ -1530,6 +1532,7 @@
gst_element_post_message (GST_ELEMENT_CAST (qtdemux), msg);
}
+ /* restart streaming, NEWSEGMENT will be sent from the streaming thread. */
gst_pad_start_task (qtdemux->sinkpad, (GstTaskFunction) gst_qtdemux_loop,
qtdemux->sinkpad, NULL);
@@ -1896,9 +1899,11 @@
qtdemux->timescale = 0;
qtdemux->got_moov = FALSE;
} else if (qtdemux->mss_mode) {
+ gst_flow_combiner_reset (qtdemux->flowcombiner);
for (n = 0; n < qtdemux->n_streams; n++)
gst_qtdemux_stream_clear (qtdemux, qtdemux->streams[n]);
} else {
+ gst_flow_combiner_reset (qtdemux->flowcombiner);
for (n = 0; n < qtdemux->n_streams; n++) {
qtdemux->streams[n]->sent_eos = FALSE;
qtdemux->streams[n]->segment_seqnum = 0;
@@ -2365,7 +2370,7 @@
GST_DEBUG ("atom type %" GST_FOURCC_FORMAT, GST_FOURCC_ARGS (fourcc));
if (length == 0) {
- length = G_MAXUINT32;
+ length = G_MAXUINT64;
} else if (length == 1 && size >= 16) {
/* this means we have an extended size, which is the 64 bit value of
* the next 8 bytes */
@@ -3236,6 +3241,15 @@
}
}
+static guint64
+add_offset (guint64 offset, guint64 advance)
+{
+ /* Avoid 64-bit overflow by clamping */
+ if (offset > G_MAXUINT64 - advance)
+ return G_MAXUINT64;
+ return offset + advance;
+}
+
static GstFlowReturn
gst_qtdemux_loop_state_header (GstQTDemux * qtdemux)
{
@@ -3289,7 +3303,7 @@
GST_LOG_OBJECT (qtdemux,
"skipping atom '%" GST_FOURCC_FORMAT "' at %" G_GUINT64_FORMAT,
GST_FOURCC_ARGS (fourcc), cur_offset);
- qtdemux->offset += length;
+ qtdemux->offset = add_offset (qtdemux->offset, length);
break;
}
case FOURCC_moov:
@@ -3298,7 +3312,7 @@
if (qtdemux->got_moov) {
GST_DEBUG_OBJECT (qtdemux, "Skipping moov atom as we have one already");
- qtdemux->offset += length;
+ qtdemux->offset = add_offset (qtdemux->offset, length);
goto beach;
}
@@ -3712,9 +3726,9 @@
/* update the segment values used for clipping */
/* accumulate previous segments */
- if (GST_CLOCK_TIME_IS_VALID (stream->segment.stop))
- stream->segment.base += (stream->segment.stop - stream->segment.start) /
- ABS (stream->segment.rate);
+ stream->segment.offset = qtdemux->segment.offset;
+ stream->segment.base = qtdemux->segment.base;
+ stream->segment.applied_rate = qtdemux->segment.applied_rate;
stream->segment.rate = rate;
stream->segment.start = start;
stream->segment.stop = stop;
@@ -3728,6 +3742,8 @@
gst_event_set_seqnum (event, stream->segment_seqnum);
}
gst_pad_push_event (stream->pad, event);
+ /* assume we can send more data now */
+ GST_PAD_LAST_FLOW_RETURN (stream->pad) = GST_FLOW_OK;
/* clear to send tags on this pad now */
gst_qtdemux_push_tags (qtdemux, stream);
}
@@ -4052,11 +4068,16 @@
* GST_FLOW_EOS: when all pads EOS or NOT_LINKED.
*/
static GstFlowReturn
-gst_qtdemux_combine_flows (GstQTDemux * demux, GstFlowReturn ret)
+gst_qtdemux_combine_flows (GstQTDemux * demux, QtDemuxStream * stream,
+ GstFlowReturn ret)
{
GST_LOG_OBJECT (demux, "flow return: %s", gst_flow_get_name (ret));
- ret = gst_flow_combiner_update_flow (demux->flowcombiner, ret);
+ if (stream->pad)
+ ret = gst_flow_combiner_update_pad_flow (demux->flowcombiner, stream->pad,
+ ret);
+ else
+ ret = gst_flow_combiner_update_flow (demux->flowcombiner, ret);
GST_LOG_OBJECT (demux, "combined flow return: %s", gst_flow_get_name (ret));
return ret;
@@ -4657,7 +4678,7 @@
}
/* combine flows */
- ret = gst_qtdemux_combine_flows (qtdemux, ret);
+ ret = gst_qtdemux_combine_flows (qtdemux, stream, ret);
/* ignore unlinked, we will not push on the pad anymore and we will EOS when
* we have no more data for the pad to push */
if (ret == GST_FLOW_EOS)
@@ -5324,7 +5345,6 @@
break;
}
case QTDEMUX_STATE_MOVIE:{
- GstBuffer *outbuf;
QtDemuxStream *stream = NULL;
QtDemuxSample *sample;
int i = -1;
@@ -5419,12 +5439,9 @@
sample = &stream->samples[stream->sample_index];
if (G_LIKELY (!(STREAM_IS_EOS (stream)))) {
- outbuf = gst_adapter_take_buffer (demux->adapter, demux->neededbytes);
GST_DEBUG_OBJECT (demux, "stream : %" GST_FOURCC_FORMAT,
GST_FOURCC_ARGS (stream->fourcc));
- g_return_val_if_fail (outbuf != NULL, GST_FLOW_ERROR);
-
dts = QTSAMPLE_DTS (stream, sample);
pts = QTSAMPLE_PTS (stream, sample);
duration = QTSAMPLE_DUR_DTS (stream, sample, dts);
@@ -5436,6 +5453,9 @@
GST_DEBUG_OBJECT (demux, "we reached the end of our segment.");
stream->time_position = GST_CLOCK_TIME_NONE; /* this means EOS */
+ /* skip this data, stream is EOS */
+ gst_adapter_flush (demux->adapter, demux->neededbytes);
+
/* check if all streams are eos */
ret = GST_FLOW_EOS;
for (i = 0; i < demux->n_streams; i++) {
@@ -5450,12 +5470,20 @@
goto eos;
}
} else {
+ GstBuffer *outbuf;
+
+ outbuf =
+ gst_adapter_take_buffer (demux->adapter, demux->neededbytes);
+
+ /* FIXME: should either be an assert or a plain check */
+ g_return_val_if_fail (outbuf != NULL, GST_FLOW_ERROR);
+
ret = gst_qtdemux_decorate_and_push_buffer (demux, stream, outbuf,
dts, pts, duration, keyframe, dts, demux->offset);
}
/* combine flows */
- ret = gst_qtdemux_combine_flows (demux, ret);
+ ret = gst_qtdemux_combine_flows (demux, stream, ret);
} else {
/* skip this data, stream is EOS */
gst_adapter_flush (demux->adapter, demux->neededbytes);
@@ -6370,10 +6398,13 @@
if (stream->pending_tags)
gst_tag_list_unref (stream->pending_tags);
stream->pending_tags = list;
+ list = NULL;
/* global tags go on each pad anyway */
stream->send_global_tags = TRUE;
}
done:
+ if (list)
+ gst_tag_list_unref (list);
return TRUE;
}
@@ -9247,7 +9278,7 @@
{
GST_INFO_OBJECT (qtdemux, "skip disabled track");
if (new_stream)
- g_free (stream);
+ gst_qtdemux_stream_free (qtdemux, stream);
return TRUE;
}
corrupt_file:
@@ -9255,14 +9286,14 @@
GST_ELEMENT_ERROR (qtdemux, STREAM, DEMUX,
(_("This file is corrupt and cannot be played.")), (NULL));
if (new_stream)
- g_free (stream);
+ gst_qtdemux_stream_free (qtdemux, stream);
return FALSE;
}
error_encrypted:
{
GST_ELEMENT_ERROR (qtdemux, STREAM, DECRYPT, (NULL), (NULL));
if (new_stream)
- g_free (stream);
+ gst_qtdemux_stream_free (qtdemux, stream);
return FALSE;
}
samples_failed:
@@ -9272,7 +9303,7 @@
/* free stbl sub-atoms */
gst_qtdemux_stbl_free (stream);
if (new_stream)
- g_free (stream);
+ gst_qtdemux_stream_free (qtdemux, stream);
return FALSE;
}
existing_stream:
@@ -9280,7 +9311,7 @@
GST_INFO_OBJECT (qtdemux, "stream with track id %i already exists",
track_id);
if (new_stream)
- g_free (stream);
+ gst_qtdemux_stream_free (qtdemux, stream);
return TRUE;
}
unknown_stream:
@@ -9288,7 +9319,7 @@
GST_INFO_OBJECT (qtdemux, "unknown subtype %" GST_FOURCC_FORMAT,
GST_FOURCC_ARGS (stream->subtype));
if (new_stream)
- g_free (stream);
+ gst_qtdemux_stream_free (qtdemux, stream);
return TRUE;
}
too_many_streams:
@@ -9450,7 +9481,7 @@
break;
++sample_num;
}
- if (stream->n_samples > 0 && stream->stbl_index > 0) {
+ if (stream->n_samples > 0 && stream->stbl_index >= 0) {
stream->first_duration = stream->samples[0].duration;
GST_LOG_OBJECT (qtdemux, "stream %d first duration %u",
stream->track_id, stream->first_duration);
@@ -10610,32 +10641,44 @@
if (rdrf) {
guint32 ref_type;
guint8 *ref_data;
+ guint ref_len;
- ref_type = QT_FOURCC ((guint8 *) rdrf->data + 12);
- ref_data = (guint8 *) rdrf->data + 20;
- if (ref_type == FOURCC_alis) {
- guint record_len, record_version, fn_len;
+ ref_len = QT_UINT32 ((guint8 *) rdrf->data);
+ if (ref_len > 20) {
+ ref_type = QT_FOURCC ((guint8 *) rdrf->data + 12);
+ ref_data = (guint8 *) rdrf->data + 20;
+ if (ref_type == FOURCC_alis) {
+ guint record_len, record_version, fn_len;
- /* MacOSX alias record, google for alias-layout.txt */
- record_len = QT_UINT16 (ref_data + 4);
- record_version = QT_UINT16 (ref_data + 4 + 2);
- fn_len = QT_UINT8 (ref_data + 50);
- if (record_len > 50 && record_version == 2 && fn_len > 0) {
- ref.location = g_strndup ((gchar *) ref_data + 51, fn_len);
+ if (ref_len > 70) {
+ /* MacOSX alias record, google for alias-layout.txt */
+ record_len = QT_UINT16 (ref_data + 4);
+ record_version = QT_UINT16 (ref_data + 4 + 2);
+ fn_len = QT_UINT8 (ref_data + 50);
+ if (record_len > 50 && record_version == 2 && fn_len > 0) {
+ ref.location = g_strndup ((gchar *) ref_data + 51, fn_len);
+ }
+ } else {
+ GST_WARNING_OBJECT (qtdemux, "Invalid rdrf/alis size (%u < 70)",
+ ref_len);
+ }
+ } else if (ref_type == FOURCC_url_) {
+ ref.location = g_strndup ((gchar *) ref_data, ref_len - 8);
+ } else {
+ GST_DEBUG_OBJECT (qtdemux,
+ "unknown rdrf reference type %" GST_FOURCC_FORMAT,
+ GST_FOURCC_ARGS (ref_type));
}
- } else if (ref_type == FOURCC_url_) {
- ref.location = g_strdup ((gchar *) ref_data);
+ if (ref.location != NULL) {
+ GST_INFO_OBJECT (qtdemux, "New location: %s", ref.location);
+ redirects =
+ g_list_prepend (redirects, g_memdup (&ref, sizeof (ref)));
+ } else {
+ GST_WARNING_OBJECT (qtdemux,
+ "Failed to extract redirect location from rdrf atom");
+ }
} else {
- GST_DEBUG_OBJECT (qtdemux,
- "unknown rdrf reference type %" GST_FOURCC_FORMAT,
- GST_FOURCC_ARGS (ref_type));
- }
- if (ref.location != NULL) {
- GST_INFO_OBJECT (qtdemux, "New location: %s", ref.location);
- redirects = g_list_prepend (redirects, g_memdup (&ref, sizeof (ref)));
- } else {
- GST_WARNING_OBJECT (qtdemux,
- "Failed to extract redirect location from rdrf atom");
+ GST_WARNING_OBJECT (qtdemux, "Invalid rdrf size (%u < 20)", ref_len);
}
}
diff --git a/gst/law/Makefile.am b/gst/law/Makefile.am
index 430fd6d..2591c78 100644
--- a/gst/law/Makefile.am
+++ b/gst/law/Makefile.am
@@ -15,28 +15,3 @@
libgstmulaw_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
noinst_HEADERS = mulaw-conversion.h alaw-encode.h alaw-decode.h mulaw-encode.h mulaw-decode.h
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstalaw -:SHARED libgstalaw \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstalaw_la_SOURCES) \
- $(nodist_libgstalaw_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstalaw_la_CFLAGS) \
- -:LDFLAGS $(libgstalaw_la_LDFLAGS) \
- $(libgstalaw_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- -:SHARED libgstmulaw \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstmulaw_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstmulaw_la_CFLAGS) \
- -:LDFLAGS $(libgstmulaw_la_LDFLAGS) \
- $(libgstmulaw_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/law/Makefile.in b/gst/law/Makefile.in
index a1ab32d..55888c5 100644
--- a/gst/law/Makefile.in
+++ b/gst/law/Makefile.in
@@ -953,31 +953,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstalaw -:SHARED libgstalaw \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstalaw_la_SOURCES) \
- $(nodist_libgstalaw_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstalaw_la_CFLAGS) \
- -:LDFLAGS $(libgstalaw_la_LDFLAGS) \
- $(libgstalaw_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- -:SHARED libgstmulaw \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstmulaw_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstmulaw_la_CFLAGS) \
- -:LDFLAGS $(libgstmulaw_la_LDFLAGS) \
- $(libgstmulaw_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/law/mulaw-decode.c b/gst/law/mulaw-decode.c
index cc3efa1..02ccd4b 100644
--- a/gst/law/mulaw-decode.c
+++ b/gst/law/mulaw-decode.c
@@ -47,7 +47,7 @@
enum
{
- ARG_0
+ PROP_0
};
#define gst_mulawdec_parent_class parent_class
diff --git a/gst/law/mulaw-encode.c b/gst/law/mulaw-encode.c
index ed943af..e92e5e6 100644
--- a/gst/law/mulaw-encode.c
+++ b/gst/law/mulaw-encode.c
@@ -43,7 +43,7 @@
enum
{
- ARG_0
+ PROP_0
};
static gboolean gst_mulawenc_start (GstAudioEncoder * audioenc);
diff --git a/gst/level/Makefile.am b/gst/level/Makefile.am
index cca655c..dbcb393 100644
--- a/gst/level/Makefile.am
+++ b/gst/level/Makefile.am
@@ -7,18 +7,3 @@
libgstlevel_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
noinst_HEADERS = gstlevel.h
-
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstlevel -:SHARED libgstlevel \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstlevel_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstlevel_la_CFLAGS) \
- -:LDFLAGS $(libgstlevel_la_LDFLAGS) \
- $(libgstlevel_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/level/Makefile.in b/gst/level/Makefile.in
index b0da788..42d3aaa 100644
--- a/gst/level/Makefile.in
+++ b/gst/level/Makefile.in
@@ -882,20 +882,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstlevel -:SHARED libgstlevel \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstlevel_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstlevel_la_CFLAGS) \
- -:LDFLAGS $(libgstlevel_la_LDFLAGS) \
- $(libgstlevel_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/level/gstlevel.c b/gst/level/gstlevel.c
index a308267..1141db6 100644
--- a/gst/level/gstlevel.c
+++ b/gst/level/gstlevel.c
@@ -168,7 +168,7 @@
static void gst_level_post_message (GstLevel * filter);
static gboolean gst_level_sink_event (GstBaseTransform * trans,
GstEvent * event);
-
+static void gst_level_recalc_interval_frames (GstLevel * level);
static void
gst_level_class_init (GstLevelClass * klass)
@@ -299,10 +299,10 @@
break;
case PROP_INTERVAL:
filter->interval = g_value_get_uint64 (value);
+ /* Not exactly thread-safe, but property does not advertise that it
+ * can be changed at runtime anyway */
if (GST_AUDIO_INFO_RATE (&filter->info)) {
- filter->interval_frames =
- GST_CLOCK_TIME_TO_FRAMES (filter->interval,
- GST_AUDIO_INFO_RATE (&filter->info));
+ gst_level_recalc_interval_frames (filter);
}
break;
case PROP_PEAK_TTL:
@@ -431,13 +431,36 @@
}
*/
+static void
+gst_level_recalc_interval_frames (GstLevel * level)
+{
+ GstClockTime interval = level->interval;
+ guint sample_rate = GST_AUDIO_INFO_RATE (&level->info);
+ guint interval_frames;
+
+ interval_frames = GST_CLOCK_TIME_TO_FRAMES (interval, sample_rate);
+
+ if (interval_frames == 0) {
+ GST_WARNING_OBJECT (level, "interval %" GST_TIME_FORMAT " is too small, "
+ "should be at least %" GST_TIME_FORMAT " for sample rate %u",
+ GST_TIME_ARGS (interval),
+ GST_TIME_ARGS (GST_FRAMES_TO_CLOCK_TIME (1, sample_rate)), sample_rate);
+ interval_frames = 1;
+ }
+
+ level->interval_frames = interval_frames;
+
+ GST_INFO_OBJECT (level, "interval_frames now %u for interval "
+ "%" GST_TIME_FORMAT " and sample rate %u", interval_frames,
+ GST_TIME_ARGS (interval), sample_rate);
+}
static gboolean
gst_level_set_caps (GstBaseTransform * trans, GstCaps * in, GstCaps * out)
{
GstLevel *filter = GST_LEVEL (trans);
GstAudioInfo info;
- gint i, channels, rate;
+ gint i, channels;
if (!gst_audio_info_from_caps (&info, in))
return FALSE;
@@ -466,7 +489,6 @@
filter->info = info;
channels = GST_AUDIO_INFO_CHANNELS (&info);
- rate = GST_AUDIO_INFO_RATE (&info);
/* allocate channel variable arrays */
g_free (filter->CS);
@@ -489,7 +511,7 @@
filter->decay_peak_age[i] = G_GUINT64_CONSTANT (0);
}
- filter->interval_frames = GST_CLOCK_TIME_TO_FRAMES (filter->interval, rate);
+ gst_level_recalc_interval_frames (filter);
return TRUE;
}
diff --git a/gst/matroska/Makefile.am b/gst/matroska/Makefile.am
index 3b19d4d..960a831 100644
--- a/gst/matroska/Makefile.am
+++ b/gst/matroska/Makefile.am
@@ -42,18 +42,3 @@
$(LIBM)
libgstmatroska_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstmatroska_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
-
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstmatroska -:SHARED libgstmatroska \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstmatroska_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(CPPFLAGS) $(libgstmatroska_la_CFLAGS) \
- -:LDFLAGS $(libgstmatroska_la_LDFLAGS) \
- $(libgstmatroska_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/matroska/Makefile.in b/gst/matroska/Makefile.in
index b315ac5..8d4847e 100644
--- a/gst/matroska/Makefile.in
+++ b/gst/matroska/Makefile.in
@@ -1001,20 +1001,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstmatroska -:SHARED libgstmatroska \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstmatroska_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(CPPFLAGS) $(libgstmatroska_la_CFLAGS) \
- -:LDFLAGS $(libgstmatroska_la_LDFLAGS) \
- $(libgstmatroska_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/matroska/matroska-demux.c b/gst/matroska/matroska-demux.c
index 251adff..0ee13b3 100644
--- a/gst/matroska/matroska-demux.c
+++ b/gst/matroska/matroska-demux.c
@@ -80,10 +80,10 @@
enum
{
- ARG_0,
- ARG_METADATA,
- ARG_STREAMINFO,
- ARG_MAX_GAP_TIME
+ PROP_0,
+ PROP_METADATA,
+ PROP_STREAMINFO,
+ PROP_MAX_GAP_TIME
};
#define DEFAULT_MAX_GAP_TIME (2 * GST_SECOND)
@@ -208,7 +208,7 @@
gobject_class->get_property = gst_matroska_demux_get_property;
gobject_class->set_property = gst_matroska_demux_set_property;
- g_object_class_install_property (gobject_class, ARG_MAX_GAP_TIME,
+ g_object_class_install_property (gobject_class, PROP_MAX_GAP_TIME,
g_param_spec_uint64 ("max-gap-time", "Maximum gap time",
"The demuxer sends out segment events for skipping "
"gaps longer than this (0 = disabled).", 0, G_MAXUINT64,
@@ -401,7 +401,6 @@
GstFlowReturn ret;
guint32 id, riff_fourcc = 0;
guint16 riff_audio_fmt = 0;
- GstTagList *list = NULL;
GstEvent *stream_start;
gchar *codec = NULL;
gchar *stream_id;
@@ -434,6 +433,7 @@
context->alignment = 1;
context->dts_only = FALSE;
context->intra_only = FALSE;
+ context->tags = gst_tag_list_new_empty ();
demux->common.num_streams++;
g_assert (demux->common.src->len == demux->common.num_streams);
@@ -1094,7 +1094,9 @@
context->codec_priv_size, &codec, &riff_fourcc);
if (codec) {
- list = gst_tag_list_new (GST_TAG_VIDEO_CODEC, codec, NULL);
+ gst_tag_list_add (context->tags, GST_TAG_MERGE_REPLACE,
+ GST_TAG_VIDEO_CODEC, codec, NULL);
+ context->tags_changed = TRUE;
g_free (codec);
}
break;
@@ -1111,7 +1113,9 @@
&codec, &riff_audio_fmt);
if (codec) {
- list = gst_tag_list_new (GST_TAG_AUDIO_CODEC, codec, NULL);
+ gst_tag_list_add (context->tags, GST_TAG_MERGE_REPLACE,
+ GST_TAG_AUDIO_CODEC, codec, NULL);
+ context->tags_changed = TRUE;
g_free (codec);
}
break;
@@ -1147,13 +1151,11 @@
if (context->language) {
const gchar *lang;
- if (!list)
- list = gst_tag_list_new_empty ();
-
/* Matroska contains ISO 639-2B codes, we want ISO 639-1 */
lang = gst_tag_get_language_code (context->language);
- gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
+ gst_tag_list_add (context->tags, GST_TAG_MERGE_REPLACE,
GST_TAG_LANGUAGE_CODE, (lang) ? lang : context->language, NULL);
+ context->tags_changed = TRUE;
}
if (caps == NULL) {
@@ -1203,8 +1205,6 @@
GST_INFO_OBJECT (demux, "Adding pad '%s' with caps %" GST_PTR_FORMAT,
padname, caps);
- context->pending_tags = list;
-
gst_pad_set_element_private (context->pad, context);
gst_pad_use_fixed_caps (context->pad);
@@ -1240,6 +1240,29 @@
gst_pad_push_event (context->pad, stream_start);
gst_pad_set_caps (context->pad, context->caps);
+
+ if (demux->common.global_tags) {
+ GstEvent *tag_event;
+
+ gst_tag_list_add (demux->common.global_tags, GST_TAG_MERGE_REPLACE,
+ GST_TAG_CONTAINER_FORMAT, "Matroska", NULL);
+ GST_DEBUG_OBJECT (context->pad, "Sending global_tags %p: %" GST_PTR_FORMAT,
+ demux->common.global_tags, demux->common.global_tags);
+
+ tag_event =
+ gst_event_new_tag (gst_tag_list_copy (demux->common.global_tags));
+
+ gst_pad_push_event (context->pad, tag_event);
+ }
+
+ if (G_UNLIKELY (context->tags_changed)) {
+ GST_DEBUG_OBJECT (context->pad, "Sending tags %p: %"
+ GST_PTR_FORMAT, context->tags, context->tags);
+ gst_pad_push_event (context->pad,
+ gst_event_new_tag (gst_tag_list_copy (context->tags)));
+ context->tags_changed = FALSE;
+ }
+
gst_element_add_pad (GST_ELEMENT (demux), context->pad);
gst_flow_combiner_add_pad (demux->flowcombiner, context->pad);
@@ -1424,14 +1447,15 @@
{
gint i;
- if (G_UNLIKELY (demux->common.global_tags != NULL)) {
+ if (G_UNLIKELY (demux->common.global_tags_changed)) {
GstEvent *tag_event;
gst_tag_list_add (demux->common.global_tags, GST_TAG_MERGE_REPLACE,
GST_TAG_CONTAINER_FORMAT, "Matroska", NULL);
GST_DEBUG_OBJECT (demux, "Sending global_tags %p : %" GST_PTR_FORMAT,
demux->common.global_tags, demux->common.global_tags);
- tag_event = gst_event_new_tag (demux->common.global_tags);
+ tag_event =
+ gst_event_new_tag (gst_tag_list_copy (demux->common.global_tags));
for (i = 0; i < demux->common.src->len; i++) {
GstMatroskaTrackContext *stream;
@@ -1441,7 +1465,7 @@
}
gst_event_unref (tag_event);
- demux->common.global_tags = NULL;
+ demux->common.global_tags_changed = FALSE;
}
g_assert (demux->common.src->len == demux->common.num_streams);
@@ -1450,13 +1474,13 @@
stream = g_ptr_array_index (demux->common.src, i);
- if (G_UNLIKELY (stream->pending_tags != NULL)) {
- GST_DEBUG_OBJECT (demux, "Sending pending_tags %p for pad %s:%s : %"
- GST_PTR_FORMAT, stream->pending_tags,
- GST_DEBUG_PAD_NAME (stream->pad), stream->pending_tags);
+ if (G_UNLIKELY (stream->tags_changed)) {
+ GST_DEBUG_OBJECT (demux, "Sending tags %p for pad %s:%s : %"
+ GST_PTR_FORMAT, stream->tags,
+ GST_DEBUG_PAD_NAME (stream->pad), stream->tags);
gst_pad_push_event (stream->pad,
- gst_event_new_tag (stream->pending_tags));
- stream->pending_tags = NULL;
+ gst_event_new_tag (gst_tag_list_copy (stream->tags)));
+ stream->tags_changed = FALSE;
}
}
}
@@ -1501,6 +1525,7 @@
/* update the time */
gst_matroska_read_common_reset_streams (&demux->common, entry->time, TRUE);
+ gst_flow_combiner_reset (demux->flowcombiner);
demux->common.segment.position = entry->time;
demux->seek_block = entry->block;
demux->seek_first = TRUE;
@@ -3654,7 +3679,8 @@
}
}
/* combine flows */
- ret = gst_flow_combiner_update_flow (demux->flowcombiner, ret);
+ ret = gst_flow_combiner_update_pad_flow (demux->flowcombiner,
+ stream->pad, ret);
next_lace:
size -= lace_size[n];
@@ -3680,7 +3706,8 @@
stream->eos = TRUE;
ret = GST_FLOW_OK;
/* combine flows */
- ret = gst_flow_combiner_update_flow (demux->flowcombiner, ret);
+ ret = gst_flow_combiner_update_pad_flow (demux->flowcombiner, stream->pad,
+ ret);
goto done;
}
invalid_lacing:
@@ -4350,8 +4377,11 @@
g_assert (event);
/* unlikely to fail, since we managed to seek to this point */
- if (!gst_matroska_demux_handle_seek_event (demux, NULL, event))
+ if (!gst_matroska_demux_handle_seek_event (demux, NULL, event)) {
+ gst_event_unref (event);
goto seek_failed;
+ }
+ gst_event_unref (event);
/* resume data handling, main thread clear to seek again */
GST_OBJECT_LOCK (demux);
demux->common.state = GST_MATROSKA_READ_STATE_DATA;
@@ -4743,6 +4773,7 @@
GST_OBJECT_LOCK (demux);
gst_matroska_read_common_reset_streams (&demux->common,
GST_CLOCK_TIME_NONE, TRUE);
+ gst_flow_combiner_reset (demux->flowcombiner);
dur = demux->common.segment.duration;
gst_segment_init (&demux->common.segment, GST_FORMAT_TIME);
demux->common.segment.duration = dur;
@@ -5714,7 +5745,7 @@
demux = GST_MATROSKA_DEMUX (object);
switch (prop_id) {
- case ARG_MAX_GAP_TIME:
+ case PROP_MAX_GAP_TIME:
GST_OBJECT_LOCK (demux);
demux->max_gap_time = g_value_get_uint64 (value);
GST_OBJECT_UNLOCK (demux);
@@ -5735,7 +5766,7 @@
demux = GST_MATROSKA_DEMUX (object);
switch (prop_id) {
- case ARG_MAX_GAP_TIME:
+ case PROP_MAX_GAP_TIME:
GST_OBJECT_LOCK (demux);
g_value_set_uint64 (value, demux->max_gap_time);
GST_OBJECT_UNLOCK (demux);
diff --git a/gst/matroska/matroska-ids.c b/gst/matroska/matroska-ids.c
index 444084b..65af1cc 100644
--- a/gst/matroska/matroska-ids.c
+++ b/gst/matroska/matroska-ids.c
@@ -331,8 +331,8 @@
g_array_free (track->encodings, TRUE);
}
- if (track->pending_tags)
- gst_tag_list_unref (track->pending_tags);
+ if (track->tags)
+ gst_tag_list_unref (track->tags);
if (track->index_table)
g_array_free (track->index_table, TRUE);
diff --git a/gst/matroska/matroska-ids.h b/gst/matroska/matroska-ids.h
index dc098dd..41315a0 100644
--- a/gst/matroska/matroska-ids.h
+++ b/gst/matroska/matroska-ids.h
@@ -540,8 +540,10 @@
GstMatroskaTrackContext *context,
GstBuffer **buffer);
- /* Tags to send after newsegment event */
- GstTagList *pending_tags;
+ /* List of tags for this stream */
+ GstTagList *tags;
+ /* Tags changed and should be pushed again */
+ gboolean tags_changed;
/* A GArray of GstMatroskaTrackEncoding structures which contain the
* encoding (compression/encryption) settings for this track, if any */
diff --git a/gst/matroska/matroska-mux.c b/gst/matroska/matroska-mux.c
index 1ccc325..0e28ed3 100644
--- a/gst/matroska/matroska-mux.c
+++ b/gst/matroska/matroska-mux.c
@@ -64,11 +64,11 @@
enum
{
- ARG_0,
- ARG_WRITING_APP,
- ARG_DOCTYPE_VERSION,
- ARG_MIN_INDEX_INTERVAL,
- ARG_STREAMABLE
+ PROP_0,
+ PROP_WRITING_APP,
+ PROP_DOCTYPE_VERSION,
+ PROP_MIN_INDEX_INTERVAL,
+ PROP_STREAMABLE
};
#define DEFAULT_DOCTYPE_VERSION 2
@@ -311,21 +311,21 @@
gobject_class->get_property = gst_matroska_mux_get_property;
gobject_class->set_property = gst_matroska_mux_set_property;
- g_object_class_install_property (gobject_class, ARG_WRITING_APP,
+ g_object_class_install_property (gobject_class, PROP_WRITING_APP,
g_param_spec_string ("writing-app", "Writing application.",
"The name the application that creates the matroska file.",
NULL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
- g_object_class_install_property (gobject_class, ARG_DOCTYPE_VERSION,
+ g_object_class_install_property (gobject_class, PROP_DOCTYPE_VERSION,
g_param_spec_int ("version", "DocType version",
"This parameter determines what Matroska features can be used.",
1, 2, DEFAULT_DOCTYPE_VERSION,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
- g_object_class_install_property (gobject_class, ARG_MIN_INDEX_INTERVAL,
+ g_object_class_install_property (gobject_class, PROP_MIN_INDEX_INTERVAL,
g_param_spec_int64 ("min-index-interval", "Minimum time between index "
"entries", "An index entry is created every so many nanoseconds.",
0, G_MAXINT64, DEFAULT_MIN_INDEX_INTERVAL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
- g_object_class_install_property (gobject_class, ARG_STREAMABLE,
+ g_object_class_install_property (gobject_class, PROP_STREAMABLE,
g_param_spec_boolean ("streamable", "Determines whether output should "
"be streamable", "If set to true, the output should be as if it is "
"to be streamed and hence no indexes written or duration written.",
@@ -3724,7 +3724,7 @@
mux = GST_MATROSKA_MUX (object);
switch (prop_id) {
- case ARG_WRITING_APP:
+ case PROP_WRITING_APP:
if (!g_value_get_string (value)) {
GST_WARNING_OBJECT (mux, "writing-app property can not be NULL");
break;
@@ -3732,13 +3732,13 @@
g_free (mux->writing_app);
mux->writing_app = g_value_dup_string (value);
break;
- case ARG_DOCTYPE_VERSION:
+ case PROP_DOCTYPE_VERSION:
mux->doctype_version = g_value_get_int (value);
break;
- case ARG_MIN_INDEX_INTERVAL:
+ case PROP_MIN_INDEX_INTERVAL:
mux->min_index_interval = g_value_get_int64 (value);
break;
- case ARG_STREAMABLE:
+ case PROP_STREAMABLE:
mux->streamable = g_value_get_boolean (value);
break;
default:
@@ -3757,16 +3757,16 @@
mux = GST_MATROSKA_MUX (object);
switch (prop_id) {
- case ARG_WRITING_APP:
+ case PROP_WRITING_APP:
g_value_set_string (value, mux->writing_app);
break;
- case ARG_DOCTYPE_VERSION:
+ case PROP_DOCTYPE_VERSION:
g_value_set_int (value, mux->doctype_version);
break;
- case ARG_MIN_INDEX_INTERVAL:
+ case PROP_MIN_INDEX_INTERVAL:
g_value_set_int64 (value, mux->min_index_interval);
break;
- case ARG_STREAMABLE:
+ case PROP_STREAMABLE:
g_value_set_boolean (value, mux->streamable);
break;
default:
diff --git a/gst/matroska/matroska-parse.c b/gst/matroska/matroska-parse.c
index 1e86a25..53b4eac 100644
--- a/gst/matroska/matroska-parse.c
+++ b/gst/matroska/matroska-parse.c
@@ -79,9 +79,7 @@
enum
{
- ARG_0,
- ARG_METADATA,
- ARG_STREAMINFO
+ PROP_0
};
static GstStaticPadTemplate sink_templ = GST_STATIC_PAD_TEMPLATE ("sink",
@@ -1071,6 +1069,27 @@
return ret;
}
+static void
+gst_matroska_parse_send_tags (GstMatroskaParse * parse)
+{
+ if (G_UNLIKELY (parse->common.global_tags_changed)) {
+ GstEvent *tag_event;
+ gst_tag_list_add (parse->common.global_tags, GST_TAG_MERGE_REPLACE,
+ GST_TAG_CONTAINER_FORMAT, "Matroska", NULL);
+ GST_DEBUG_OBJECT (parse, "Sending global_tags %p : %" GST_PTR_FORMAT,
+ parse->common.global_tags, parse->common.global_tags);
+
+ /* Send a copy as we want to keep our local ref writable to add more tags
+ * if any are found */
+ tag_event =
+ gst_event_new_tag (gst_tag_list_copy (parse->common.global_tags));
+
+ gst_pad_push_event (parse->srcpad, tag_event);
+
+ parse->common.global_tags_changed = FALSE;
+ }
+}
+
/* returns FALSE if there are no pads to deliver event to,
* otherwise TRUE (whatever the outcome of event sending),
* takes ownership of the passed event! */
@@ -2567,6 +2586,8 @@
if (!parse->common.segmentinfo_parsed) {
ret = gst_matroska_read_common_parse_info (&parse->common,
GST_ELEMENT_CAST (parse), &ebml);
+ if (ret == GST_FLOW_OK)
+ gst_matroska_parse_send_tags (parse);
}
gst_matroska_parse_accumulate_streamheader (parse, ebml.buf);
break;
@@ -2660,6 +2681,8 @@
if (!parse->common.attachments_parsed) {
ret = gst_matroska_read_common_parse_attachments (&parse->common,
GST_ELEMENT_CAST (parse), &ebml);
+ if (ret == GST_FLOW_OK)
+ gst_matroska_parse_send_tags (parse);
}
gst_matroska_parse_output (parse, ebml.buf, FALSE);
break;
@@ -2667,6 +2690,8 @@
GST_READ_CHECK (gst_matroska_parse_take (parse, read, &ebml));
ret = gst_matroska_read_common_parse_metadata (&parse->common,
GST_ELEMENT_CAST (parse), &ebml);
+ if (ret == GST_FLOW_OK)
+ gst_matroska_parse_send_tags (parse);
gst_matroska_parse_accumulate_streamheader (parse, ebml.buf);
break;
case GST_MATROSKA_ID_CHAPTERS:
diff --git a/gst/matroska/matroska-read-common.c b/gst/matroska/matroska-read-common.c
index 59feb9c..0b8e700 100644
--- a/gst/matroska/matroska-read-common.c
+++ b/gst/matroska/matroska-read-common.c
@@ -452,6 +452,7 @@
} else {
common->global_tags = taglist;
}
+ common->global_tags_changed = TRUE;
}
gint64
@@ -2209,71 +2210,67 @@
return;
for (i = 0; i < vallen; i++) {
- GValue val = { 0 };
const GValue *val_ref;
val_ref = gst_tag_list_get_value_index (list, tag, i);
if (val_ref == NULL)
continue;
- g_value_init (&val, G_VALUE_TYPE (val_ref));
- g_value_copy (val_ref, &val);
/* TODO: use the optional ctx->target_type somehow */
if (strcmp (tag, GST_TAG_TITLE) == 0) {
if (ctx->target_type_value >= 70 && !ctx->audio_only) {
gst_tag_list_add_value (ctx->result, GST_TAG_MERGE_APPEND,
- GST_TAG_SHOW_NAME, &val);
+ GST_TAG_SHOW_NAME, val_ref);
continue;
} else if (ctx->target_type_value >= 50) {
gst_tag_list_add_value (ctx->result, GST_TAG_MERGE_APPEND,
- GST_TAG_ALBUM, &val);
+ GST_TAG_ALBUM, val_ref);
continue;
}
} else if (strcmp (tag, GST_TAG_TITLE_SORTNAME) == 0) {
if (ctx->target_type_value >= 70 && !ctx->audio_only) {
gst_tag_list_add_value (ctx->result, GST_TAG_MERGE_APPEND,
- GST_TAG_SHOW_SORTNAME, &val);
+ GST_TAG_SHOW_SORTNAME, val_ref);
continue;
} else if (ctx->target_type_value >= 50) {
gst_tag_list_add_value (ctx->result, GST_TAG_MERGE_APPEND,
- GST_TAG_ALBUM_SORTNAME, &val);
+ GST_TAG_ALBUM_SORTNAME, val_ref);
continue;
}
} else if (strcmp (tag, GST_TAG_ARTIST) == 0) {
if (ctx->target_type_value >= 50) {
gst_tag_list_add_value (ctx->result, GST_TAG_MERGE_APPEND,
- GST_TAG_ALBUM_ARTIST, &val);
+ GST_TAG_ALBUM_ARTIST, val_ref);
continue;
}
} else if (strcmp (tag, GST_TAG_ARTIST_SORTNAME) == 0) {
if (ctx->target_type_value >= 50) {
gst_tag_list_add_value (ctx->result, GST_TAG_MERGE_APPEND,
- GST_TAG_ALBUM_ARTIST_SORTNAME, &val);
+ GST_TAG_ALBUM_ARTIST_SORTNAME, val_ref);
continue;
}
} else if (strcmp (tag, GST_TAG_TRACK_COUNT) == 0) {
if (ctx->target_type_value >= 60) {
gst_tag_list_add_value (ctx->result, GST_TAG_MERGE_APPEND,
- GST_TAG_ALBUM_VOLUME_COUNT, &val);
+ GST_TAG_ALBUM_VOLUME_COUNT, val_ref);
continue;
}
} else if (strcmp (tag, GST_TAG_TRACK_NUMBER) == 0) {
if (ctx->target_type_value >= 60 && !ctx->audio_only) {
gst_tag_list_add_value (ctx->result, GST_TAG_MERGE_APPEND,
- GST_TAG_SHOW_SEASON_NUMBER, &val);
+ GST_TAG_SHOW_SEASON_NUMBER, val_ref);
continue;
} else if (ctx->target_type_value >= 50 && !ctx->audio_only) {
gst_tag_list_add_value (ctx->result, GST_TAG_MERGE_APPEND,
- GST_TAG_SHOW_EPISODE_NUMBER, &val);
+ GST_TAG_SHOW_EPISODE_NUMBER, val_ref);
continue;
} else if (ctx->target_type_value >= 50) {
gst_tag_list_add_value (ctx->result, GST_TAG_MERGE_APPEND,
- GST_TAG_ALBUM_VOLUME_NUMBER, &val);
+ GST_TAG_ALBUM_VOLUME_NUMBER, val_ref);
continue;
}
}
- gst_tag_list_add_value (ctx->result, GST_TAG_MERGE_APPEND, tag, &val);
- g_value_unset (&val);
+ gst_tag_list_add_value (ctx->result, GST_TAG_MERGE_APPEND, tag, val_ref);
}
}
@@ -2387,16 +2384,13 @@
GstMatroskaTrackContext *stream = g_ptr_array_index (common->src, j);
if (stream->uid == tgt) {
- if (stream->pending_tags == NULL)
- stream->pending_tags = gst_tag_list_new_empty ();
-
- gst_tag_list_insert (stream->pending_tags, taglist,
- GST_TAG_MERGE_REPLACE);
+ gst_tag_list_insert (stream->tags, taglist, GST_TAG_MERGE_REPLACE);
+ stream->tags_changed = TRUE;
found = TRUE;
}
}
if (!found) {
- GST_WARNING_OBJECT (common->sinkpad,
+ GST_FIXME_OBJECT (common->sinkpad,
"Found track-specific tag(s), but track %" G_GUINT64_FORMAT
" is not known (yet?)", tgt);
}
@@ -2923,6 +2917,7 @@
ctx->chapters_parsed = FALSE;
/* tags */
+ ctx->global_tags_changed = FALSE;
g_list_foreach (ctx->tags_parsed,
(GFunc) gst_matroska_read_common_free_parsed_el, NULL);
g_list_free (ctx->tags_parsed);
diff --git a/gst/matroska/matroska-read-common.h b/gst/matroska/matroska-read-common.h
index de21026..3be9542 100644
--- a/gst/matroska/matroska-read-common.h
+++ b/gst/matroska/matroska-read-common.h
@@ -90,6 +90,7 @@
GstSegment segment;
GstTagList *global_tags;
+ gboolean global_tags_changed;
/* pull mode caching */
GstBuffer *cached_buffer;
diff --git a/gst/monoscope/Makefile.am b/gst/monoscope/Makefile.am
index aac7b66..36d2ed6 100644
--- a/gst/monoscope/Makefile.am
+++ b/gst/monoscope/Makefile.am
@@ -8,17 +8,3 @@
libgstmonoscope_la_LIBADD = $(GST_LIBS) $(GST_BASE_LIBS)
libgstmonoscope_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstmonoscope_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstmonoscope -:SHARED libgstmonoscope \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstmonoscope_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstmonoscope_la_CFLAGS) \
- -:LDFLAGS $(libgstmonoscope_la_LDFLAGS) \
- $(libgstmonoscope_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/monoscope/Makefile.in b/gst/monoscope/Makefile.in
index 57760fc..098ce20 100644
--- a/gst/monoscope/Makefile.in
+++ b/gst/monoscope/Makefile.in
@@ -899,20 +899,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstmonoscope -:SHARED libgstmonoscope \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstmonoscope_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstmonoscope_la_CFLAGS) \
- -:LDFLAGS $(libgstmonoscope_la_LDFLAGS) \
- $(libgstmonoscope_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/multifile/Makefile.am b/gst/multifile/Makefile.am
index d15bc1f..f7aa42a 100644
--- a/gst/multifile/Makefile.am
+++ b/gst/multifile/Makefile.am
@@ -30,17 +30,3 @@
test_splitmux_part_reader_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(GIO_CFLAGS)
test_splitmux_part_reader_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstvideo-@GST_API_VERSION@ $(GST_BASE_LIBS) $(GST_LIBS) $(GIO_LIBS)
test_splitmux_part_reader_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstmultifile -:SHARED libgstmultifile \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstmultifile_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstmultifile_la_CFLAGS) \
- -:LDFLAGS $(libgstmultifile_la_LDFLAGS) \
- $(libgstmultifile_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/multifile/Makefile.in b/gst/multifile/Makefile.in
index 050e09d..7289eec 100644
--- a/gst/multifile/Makefile.in
+++ b/gst/multifile/Makefile.in
@@ -1086,20 +1086,6 @@
uninstall-am uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstmultifile -:SHARED libgstmultifile \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstmultifile_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstmultifile_la_CFLAGS) \
- -:LDFLAGS $(libgstmultifile_la_LDFLAGS) \
- $(libgstmultifile_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/multifile/gstmultifilesink.c b/gst/multifile/gstmultifilesink.c
index 85d1cd3..c035f55 100644
--- a/gst/multifile/gstmultifilesink.c
+++ b/gst/multifile/gstmultifilesink.c
@@ -28,6 +28,14 @@
*
* Write incoming data to a series of sequentially-named files.
*
+ * This element is usually used with data where each buffer is an
+ * independent unit of data in its own right (e.g. raw video buffers or
+ * encoded JPEG or PNG images) or with streamable container formats such
+ * as MPEG-TS or MPEG-PS.
+ *
+ * It is not possible to use this element to create independently playable
+ * mp4 files, use the splitmuxsink element for that instead.
+ *
* The filename property should contain a string with a \%d placeholder that will
* be substituted with the index for each filename.
*
@@ -136,8 +144,7 @@
PROP_POST_MESSAGES,
PROP_NEXT_FILE,
PROP_MAX_FILES,
- PROP_MAX_FILE_SIZE,
- PROP_LAST
+ PROP_MAX_FILE_SIZE
};
static void gst_multi_file_sink_finalize (GObject * object);
@@ -434,6 +441,30 @@
gst_message_new_element (GST_OBJECT_CAST (multifilesink), s));
}
+static void
+gst_multi_file_sink_post_message_from_time (GstMultiFileSink * multifilesink,
+ GstClockTime timestamp, GstClockTime duration, const char *filename)
+{
+ GstClockTime running_time, stream_time;
+ guint64 offset, offset_end;
+ GstSegment *segment;
+ GstFormat format;
+
+ if (!multifilesink->post_messages)
+ return;
+
+ segment = &GST_BASE_SINK (multifilesink)->segment;
+ format = segment->format;
+
+ offset = -1;
+ offset_end = -1;
+
+ running_time = gst_segment_to_running_time (segment, format, timestamp);
+ stream_time = gst_segment_to_stream_time (segment, format, timestamp);
+
+ gst_multi_file_sink_post_message_full (multifilesink, timestamp, duration,
+ offset, offset_end, running_time, stream_time, filename);
+}
static void
gst_multi_file_sink_post_message (GstMultiFileSink * multifilesink,
@@ -810,13 +841,12 @@
offset = offset_end = -1;
filename = g_strdup_printf (multifilesink->filename,
multifilesink->index);
- gst_multi_file_sink_post_message_full (multifilesink, timestamp,
- duration, offset, offset_end, running_time, stream_time, filename);
-
- g_free (filename);
gst_multi_file_sink_close_file (multifilesink, NULL);
+ gst_multi_file_sink_post_message_full (multifilesink, timestamp,
+ duration, offset, offset_end, running_time, stream_time, filename);
+ g_free (filename);
}
if (multifilesink->file == NULL) {
@@ -826,6 +856,20 @@
break;
}
+ case GST_EVENT_EOS:
+ if (multifilesink->file) {
+ gchar *filename;
+
+ filename = g_strdup_printf (multifilesink->filename,
+ multifilesink->index);
+
+ gst_multi_file_sink_close_file (multifilesink, NULL);
+
+ gst_multi_file_sink_post_message_from_time (multifilesink,
+ GST_BASE_SINK (multifilesink)->segment.position, -1, filename);
+ g_free (filename);
+ }
+ break;
default:
break;
}
diff --git a/gst/multifile/gstmultifilesrc.c b/gst/multifile/gstmultifilesrc.c
index c1c1cf1..01dae98 100644
--- a/gst/multifile/gstmultifilesrc.c
+++ b/gst/multifile/gstmultifilesrc.c
@@ -72,13 +72,13 @@
enum
{
- ARG_0,
- ARG_LOCATION,
- ARG_INDEX,
- ARG_START_INDEX,
- ARG_STOP_INDEX,
- ARG_CAPS,
- ARG_LOOP
+ PROP_0,
+ PROP_LOCATION,
+ PROP_INDEX,
+ PROP_START_INDEX,
+ PROP_STOP_INDEX,
+ PROP_CAPS,
+ PROP_LOOP
};
#define DEFAULT_LOCATION "%05d"
@@ -143,35 +143,35 @@
gobject_class->set_property = gst_multi_file_src_set_property;
gobject_class->get_property = gst_multi_file_src_get_property;
- g_object_class_install_property (gobject_class, ARG_LOCATION,
+ g_object_class_install_property (gobject_class, PROP_LOCATION,
g_param_spec_string ("location", "File Location",
"Pattern to create file names of input files. File names are "
"created by calling sprintf() with the pattern and the current "
"index.", DEFAULT_LOCATION,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
- g_object_class_install_property (gobject_class, ARG_INDEX,
+ g_object_class_install_property (gobject_class, PROP_INDEX,
g_param_spec_int ("index", "File Index",
"Index to use with location property to create file names. The "
"index is incremented by one for each buffer read.",
0, INT_MAX, DEFAULT_INDEX,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
- g_object_class_install_property (gobject_class, ARG_START_INDEX,
+ g_object_class_install_property (gobject_class, PROP_START_INDEX,
g_param_spec_int ("start-index", "Start Index",
"Start value of index. The initial value of index can be set "
"either by setting index or start-index. When the end of the loop "
"is reached, the index will be set to the value start-index.",
0, INT_MAX, DEFAULT_INDEX,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
- g_object_class_install_property (gobject_class, ARG_STOP_INDEX,
+ g_object_class_install_property (gobject_class, PROP_STOP_INDEX,
g_param_spec_int ("stop-index", "Stop Index",
"Stop value of index. The special value -1 means no stop.",
-1, INT_MAX, DEFAULT_INDEX,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
- g_object_class_install_property (gobject_class, ARG_CAPS,
+ g_object_class_install_property (gobject_class, PROP_CAPS,
g_param_spec_boxed ("caps", "Caps",
"Caps describing the format of the data.",
GST_TYPE_CAPS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
- g_object_class_install_property (gobject_class, ARG_LOOP,
+ g_object_class_install_property (gobject_class, PROP_LOOP,
g_param_spec_boolean ("loop", "Loop",
"Whether to repeat from the beginning when all files have been read.",
FALSE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
@@ -295,19 +295,19 @@
GstMultiFileSrc *src = GST_MULTI_FILE_SRC (object);
switch (prop_id) {
- case ARG_LOCATION:
+ case PROP_LOCATION:
gst_multi_file_src_set_location (src, g_value_get_string (value));
break;
- case ARG_INDEX:
+ case PROP_INDEX:
src->index = g_value_get_int (value);
break;
- case ARG_START_INDEX:
+ case PROP_START_INDEX:
src->start_index = g_value_get_int (value);
break;
- case ARG_STOP_INDEX:
+ case PROP_STOP_INDEX:
src->stop_index = g_value_get_int (value);
break;
- case ARG_CAPS:
+ case PROP_CAPS:
{
GstStructure *st = NULL;
const GstCaps *caps = gst_value_get_caps (value);
@@ -333,7 +333,7 @@
}
}
break;
- case ARG_LOOP:
+ case PROP_LOOP:
src->loop = g_value_get_boolean (value);
break;
default:
@@ -349,22 +349,22 @@
GstMultiFileSrc *src = GST_MULTI_FILE_SRC (object);
switch (prop_id) {
- case ARG_LOCATION:
+ case PROP_LOCATION:
g_value_set_string (value, src->filename);
break;
- case ARG_INDEX:
+ case PROP_INDEX:
g_value_set_int (value, src->index);
break;
- case ARG_START_INDEX:
+ case PROP_START_INDEX:
g_value_set_int (value, src->start_index);
break;
- case ARG_STOP_INDEX:
+ case PROP_STOP_INDEX:
g_value_set_int (value, src->stop_index);
break;
- case ARG_CAPS:
+ case PROP_CAPS:
gst_value_set_caps (value, src->caps);
break;
- case ARG_LOOP:
+ case PROP_LOOP:
g_value_set_boolean (value, src->loop);
break;
default:
diff --git a/gst/multifile/gstsplitmuxsink.c b/gst/multifile/gstsplitmuxsink.c
index 3c3b3c2..96f4d1d 100644
--- a/gst/multifile/gstsplitmuxsink.c
+++ b/gst/multifile/gstsplitmuxsink.c
@@ -248,6 +248,7 @@
{
GstSplitMuxSink *splitmux = GST_SPLITMUX_SINK (object);
g_cond_clear (&splitmux->data_cond);
+ g_mutex_clear (&splitmux->lock);
if (splitmux->provided_sink)
gst_object_unref (splitmux->provided_sink);
if (splitmux->provided_muxer)
@@ -1176,7 +1177,7 @@
mq_stream_ctx_ref (ctx);
ctx->sink_pad_block_id =
- gst_pad_add_probe (res, GST_PAD_PROBE_TYPE_DATA_DOWNSTREAM,
+ gst_pad_add_probe (mq_sink, GST_PAD_PROBE_TYPE_DATA_DOWNSTREAM,
(GstPadProbeCallback) handle_mq_input, ctx, (GDestroyNotify)
_pad_block_destroy_sink_notify);
@@ -1294,7 +1295,7 @@
create_element (splitmux, "mp4mux", "muxer")) == NULL)
goto fail;
} else {
- if (!gst_bin_add (GST_BIN (splitmux), splitmux->provided_muxer)) {
+ if (!gst_bin_add (GST_BIN (splitmux), provided_muxer)) {
g_warning ("Could not add muxer element - splitmuxsink will not work");
gst_object_unref (provided_muxer);
goto fail;
diff --git a/gst/multifile/gstsplitmuxsrc.c b/gst/multifile/gstsplitmuxsrc.c
index ea3ea75..9e310c1 100644
--- a/gst/multifile/gstsplitmuxsrc.c
+++ b/gst/multifile/gstsplitmuxsrc.c
@@ -557,8 +557,8 @@
error:
/* Fall through */
- GST_ELEMENT_ERROR (splitmux, RESOURCE, OPEN_READ,
- ("Error reading part file %s", GST_STR_NULL (reader->path)), (NULL));
+ GST_ELEMENT_ERROR (splitmux, RESOURCE, OPEN_READ, (NULL),
+ ("Error reading part file %s", GST_STR_NULL (reader->path)));
flushing:
gst_pad_pause_task (pad);
gst_object_unref (splitmux);
diff --git a/gst/multipart/Makefile.am b/gst/multipart/Makefile.am
index 124c67f..5ab670e 100644
--- a/gst/multipart/Makefile.am
+++ b/gst/multipart/Makefile.am
@@ -7,18 +7,3 @@
libgstmultipart_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
noinst_HEADERS = multipartdemux.h multipartmux.h
-
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstmultipart -:SHARED libgstmultipart \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstmultipart_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstmultipart_la_CFLAGS) \
- -:LDFLAGS $(libgstmultipart_la_LDFLAGS) \
- $(libgstmultipart_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/multipart/Makefile.in b/gst/multipart/Makefile.in
index d951ad7..e46539a 100644
--- a/gst/multipart/Makefile.in
+++ b/gst/multipart/Makefile.in
@@ -900,20 +900,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstmultipart -:SHARED libgstmultipart \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstmultipart_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstmultipart_la_CFLAGS) \
- -:LDFLAGS $(libgstmultipart_la_LDFLAGS) \
- $(libgstmultipart_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/multipart/multipartmux.c b/gst/multipart/multipartmux.c
index 6814293..2ac67c1 100644
--- a/gst/multipart/multipartmux.c
+++ b/gst/multipart/multipartmux.c
@@ -47,8 +47,8 @@
enum
{
- ARG_0,
- ARG_BOUNDARY
+ PROP_0,
+ PROP_BOUNDARY
/* FILL ME */
};
@@ -114,7 +114,7 @@
gobject_class->get_property = gst_multipart_mux_get_property;
gobject_class->set_property = gst_multipart_mux_set_property;
- g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_BOUNDARY,
+ g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BOUNDARY,
g_param_spec_string ("boundary", "Boundary", "Boundary string",
DEFAULT_BOUNDARY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
@@ -153,8 +153,8 @@
multipart_mux->collect = gst_collect_pads_new ();
gst_collect_pads_set_event_function (multipart_mux->collect,
- (GstCollectPadsEventFunction) GST_DEBUG_FUNCPTR (gst_multipart_mux_sink_event),
- multipart_mux);
+ (GstCollectPadsEventFunction)
+ GST_DEBUG_FUNCPTR (gst_multipart_mux_sink_event), multipart_mux);
gst_collect_pads_set_function (multipart_mux->collect,
(GstCollectPadsFunction) GST_DEBUG_FUNCPTR (gst_multipart_mux_collected),
multipart_mux);
@@ -600,7 +600,7 @@
mux = GST_MULTIPART_MUX (object);
switch (prop_id) {
- case ARG_BOUNDARY:
+ case PROP_BOUNDARY:
g_value_set_string (value, mux->boundary);
break;
default:
@@ -618,7 +618,7 @@
mux = GST_MULTIPART_MUX (object);
switch (prop_id) {
- case ARG_BOUNDARY:
+ case PROP_BOUNDARY:
g_free (mux->boundary);
mux->boundary = g_strdup (g_value_get_string (value));
break;
diff --git a/gst/replaygain/Makefile.am b/gst/replaygain/Makefile.am
index d352f18..87f64c3 100644
--- a/gst/replaygain/Makefile.am
+++ b/gst/replaygain/Makefile.am
@@ -21,18 +21,3 @@
gstrgvolume.h \
replaygain.h \
rganalysis.h
-
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstreplaygain -:SHARED libgstreplaygain \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstreplaygain_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstreplaygain_la_CFLAGS) \
- -:LDFLAGS $(libgstreplaygain_la_LDFLAGS) \
- $(libgstreplaygain_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/replaygain/Makefile.in b/gst/replaygain/Makefile.in
index 947f1b9..a071266 100644
--- a/gst/replaygain/Makefile.in
+++ b/gst/replaygain/Makefile.in
@@ -938,20 +938,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstreplaygain -:SHARED libgstreplaygain \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstreplaygain_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstreplaygain_la_CFLAGS) \
- -:LDFLAGS $(libgstreplaygain_la_LDFLAGS) \
- $(libgstreplaygain_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/rtp/Makefile.am b/gst/rtp/Makefile.am
index b4a621b..441c52f 100644
--- a/gst/rtp/Makefile.am
+++ b/gst/rtp/Makefile.am
@@ -180,17 +180,3 @@
gstrtpstreamdepay.h
EXTRA_DIST = dboolhuff.LICENSE
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstrtp -:SHARED libgstrtp \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstrtp_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstrtp_la_CFLAGS) \
- -:LDFLAGS $(libgstrtp_la_LDFLAGS) \
- $(libgstrtp_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/rtp/Makefile.in b/gst/rtp/Makefile.in
index 931b9ac..06bc305 100644
--- a/gst/rtp/Makefile.in
+++ b/gst/rtp/Makefile.in
@@ -1734,20 +1734,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstrtp -:SHARED libgstrtp \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstrtp_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstrtp_la_CFLAGS) \
- -:LDFLAGS $(libgstrtp_la_LDFLAGS) \
- $(libgstrtp_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/rtp/gstrtpL16depay.c b/gst/rtp/gstrtpL16depay.c
index 5bbbedc..667a64a 100644
--- a/gst/rtp/gstrtpL16depay.c
+++ b/gst/rtp/gstrtpL16depay.c
@@ -27,7 +27,7 @@
* <refsect2>
* <title>Example pipeline</title>
* |[
- * gst-launch udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1, payload=(int)96' ! rtpL16depay ! pulsesink
+ * gst-launch-1.0 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1, payload=(int)96' ! rtpL16depay ! pulsesink
* ]| This example pipeline will depayload an RTP raw audio stream. Refer to
* the rtpL16pay example to create the RTP stream.
* </refsect2>
diff --git a/gst/rtp/gstrtpL16pay.c b/gst/rtp/gstrtpL16pay.c
index 5d92c25..a8ed36f 100644
--- a/gst/rtp/gstrtpL16pay.c
+++ b/gst/rtp/gstrtpL16pay.c
@@ -27,7 +27,7 @@
* <refsect2>
* <title>Example pipeline</title>
* |[
- * gst-launch -v audiotestsrc ! audioconvert ! rtpL16pay ! udpsink
+ * gst-launch-1.0 -v audiotestsrc ! audioconvert ! rtpL16pay ! udpsink
* ]| This example pipeline will payload raw audio. Refer to
* the rtpL16depay example to depayload and play the RTP stream.
* </refsect2>
diff --git a/gst/rtp/gstrtpL24depay.c b/gst/rtp/gstrtpL24depay.c
index 1925ce0..7b5ef0c 100644
--- a/gst/rtp/gstrtpL24depay.c
+++ b/gst/rtp/gstrtpL24depay.c
@@ -27,7 +27,7 @@
* <refsect2>
* <title>Example pipeline</title>
* |[
- * gst-launch udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L24, encoding-params=(string)1, channels=(int)1, payload=(int)96' ! rtpL24depay ! pulsesink
+ * gst-launch-1.0 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L24, encoding-params=(string)1, channels=(int)1, payload=(int)96' ! rtpL24depay ! pulsesink
* ]| This example pipeline will depayload an RTP raw audio stream. Refer to
* the rtpL24pay example to create the RTP stream.
* </refsect2>
diff --git a/gst/rtp/gstrtpL24pay.c b/gst/rtp/gstrtpL24pay.c
index d2612e9..1bb37cb 100644
--- a/gst/rtp/gstrtpL24pay.c
+++ b/gst/rtp/gstrtpL24pay.c
@@ -27,7 +27,7 @@
* <refsect2>
* <title>Example pipeline</title>
* |[
- * gst-launch -v audiotestsrc ! audioconvert ! rtpL24pay ! udpsink
+ * gst-launch-1.0 -v audiotestsrc ! audioconvert ! rtpL24pay ! udpsink
* ]| This example pipeline will payload raw audio. Refer to
* the rtpL24depay example to depayload and play the RTP stream.
* </refsect2>
diff --git a/gst/rtp/gstrtpac3pay.c b/gst/rtp/gstrtpac3pay.c
index a6ef914..e283afd 100644
--- a/gst/rtp/gstrtpac3pay.c
+++ b/gst/rtp/gstrtpac3pay.c
@@ -27,7 +27,7 @@
* <refsect2>
* <title>Example pipeline</title>
* |[
- * gst-launch -v audiotestsrc ! avenc_ac3 ! rtpac3pay ! udpsink
+ * gst-launch-1.0 -v audiotestsrc ! avenc_ac3 ! rtpac3pay ! udpsink
* ]| This example pipeline will encode and payload AC3 stream. Refer to
* the rtpac3depay example to depayload and decode the RTP stream.
* </refsect2>
diff --git a/gst/rtp/gstrtpamrdepay.c b/gst/rtp/gstrtpamrdepay.c
index 1156ac9..4b53843 100644
--- a/gst/rtp/gstrtpamrdepay.c
+++ b/gst/rtp/gstrtpamrdepay.c
@@ -61,7 +61,7 @@
enum
{
- ARG_0
+ PROP_0
};
/* input is an RTP packet
@@ -74,8 +74,7 @@
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
- "clock-rate = (int) 8000, "
- "encoding-name = (string) \"AMR\", "
+ "clock-rate = (int) 8000, " "encoding-name = (string) \"AMR\", "
/* This is the default, so the peer doesn't have to specify it
* "encoding-params = (string) \"1\", " */
/* NOTE that all values must be strings in orde to be able to do SDP <->
@@ -94,8 +93,7 @@
*/
"application/x-rtp, "
"media = (string) \"audio\", "
- "clock-rate = (int) 16000, "
- "encoding-name = (string) \"AMR-WB\", "
+ "clock-rate = (int) 16000, " "encoding-name = (string) \"AMR-WB\", "
/* This is the default, so the peer doesn't have to specify it
* "encoding-params = (string) \"1\", " */
/* NOTE that all values must be strings in orde to be able to do SDP <->
diff --git a/gst/rtp/gstrtpamrpay.c b/gst/rtp/gstrtpamrpay.c
index ed319cc..ead9f94 100644
--- a/gst/rtp/gstrtpamrpay.c
+++ b/gst/rtp/gstrtpamrpay.c
@@ -27,7 +27,7 @@
* <refsect2>
* <title>Example pipeline</title>
* |[
- * gst-launch -v audiotestsrc ! amrnbenc ! rtpamrpay ! udpsink
+ * gst-launch-1.0 -v audiotestsrc ! amrnbenc ! rtpamrpay ! udpsink
* ]| This example pipeline will encode and payload an AMR stream. Refer to
* the rtpamrdepay example to depayload and decode the RTP stream.
* </refsect2>
diff --git a/gst/rtp/gstrtpceltdepay.c b/gst/rtp/gstrtpceltdepay.c
index 7203f10..5e15cc6 100644
--- a/gst/rtp/gstrtpceltdepay.c
+++ b/gst/rtp/gstrtpceltdepay.c
@@ -44,7 +44,7 @@
enum
{
- ARG_0
+ PROP_0
};
static GstStaticPadTemplate gst_rtp_celt_depay_sink_template =
diff --git a/gst/rtp/gstrtpdvdepay.c b/gst/rtp/gstrtpdvdepay.c
index 7cb1ff4..f13b696 100644
--- a/gst/rtp/gstrtpdvdepay.c
+++ b/gst/rtp/gstrtpdvdepay.c
@@ -45,7 +45,7 @@
enum
{
- ARG_0,
+ PROP_0,
};
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
diff --git a/gst/rtp/gstrtpg723depay.c b/gst/rtp/gstrtpg723depay.c
index b822688..fb726f2 100644
--- a/gst/rtp/gstrtpg723depay.c
+++ b/gst/rtp/gstrtpg723depay.c
@@ -45,7 +45,7 @@
enum
{
- ARG_0
+ PROP_0
};
/* input is an RTP packet
diff --git a/gst/rtp/gstrtpg726depay.c b/gst/rtp/gstrtpg726depay.c
index 45d99dc..0b4a041 100644
--- a/gst/rtp/gstrtpg726depay.c
+++ b/gst/rtp/gstrtpg726depay.c
@@ -50,8 +50,7 @@
enum
{
PROP_0,
- PROP_FORCE_AAL2,
- PROP_LAST
+ PROP_FORCE_AAL2
};
static GstStaticPadTemplate gst_rtp_g726_depay_sink_template =
diff --git a/gst/rtp/gstrtpg726pay.c b/gst/rtp/gstrtpg726pay.c
index a1c8f5d..da9fe25 100644
--- a/gst/rtp/gstrtpg726pay.c
+++ b/gst/rtp/gstrtpg726pay.c
@@ -38,8 +38,7 @@
enum
{
PROP_0,
- PROP_FORCE_AAL2,
- PROP_LAST
+ PROP_FORCE_AAL2
};
static GstStaticPadTemplate gst_rtp_g726_pay_sink_template =
diff --git a/gst/rtp/gstrtpg729depay.c b/gst/rtp/gstrtpg729depay.c
index f763e4c..9910add 100644
--- a/gst/rtp/gstrtpg729depay.c
+++ b/gst/rtp/gstrtpg729depay.c
@@ -43,7 +43,7 @@
enum
{
- ARG_0
+ PROP_0
};
/* input is an RTP packet
diff --git a/gst/rtp/gstrtpgstpay.c b/gst/rtp/gstrtpgstpay.c
index 08794d1..6167008 100644
--- a/gst/rtp/gstrtpgstpay.c
+++ b/gst/rtp/gstrtpgstpay.c
@@ -76,8 +76,7 @@
enum
{
PROP_0,
- PROP_CONFIG_INTERVAL,
- PROP_LAST
+ PROP_CONFIG_INTERVAL
};
#define DEFAULT_CONFIG_INTERVAL 0
diff --git a/gst/rtp/gstrtph263depay.c b/gst/rtp/gstrtph263depay.c
index 83e9012..a63b0b1 100644
--- a/gst/rtp/gstrtph263depay.c
+++ b/gst/rtp/gstrtph263depay.c
@@ -55,9 +55,16 @@
"media = (string) \"video\", "
"payload = (int) " GST_RTP_PAYLOAD_H263_STRING ", "
"clock-rate = (int) 90000; "
+ /* optional SDP attribute:
+ * "a-framesize = (string) \"1234-1234\", "
+ */
"application/x-rtp, "
"media = (string) \"video\", "
- "clock-rate = (int) 90000, " "encoding-name = (string) \"H263\"")
+ "clock-rate = (int) 90000, " "encoding-name = (string) \"H263\""
+ /* optional SDP attribute:
+ * "a-framesize = (string) \"1234-1234\", "
+ */
+ )
);
#define gst_rtp_h263_depay_parent_class parent_class
@@ -129,20 +136,67 @@
G_OBJECT_CLASS (parent_class)->finalize (object);
}
+static gboolean
+gst_rtp_h263_parse_framesize (GstRTPBaseDepayload * filter,
+ const gchar * media_attr, GstCaps * srccaps)
+{
+ gchar *dimension, *endptr;
+ gint width, height;
+ GstStructure *d;
+
+ width = g_ascii_strtoull (media_attr, &endptr, 10);
+ if (width <= 0) {
+ GST_ERROR_OBJECT (filter,
+ "Framesize media attribute width out of valid range");
+ return FALSE;
+ } else if (*endptr != '-') {
+ GST_ERROR_OBJECT (filter,
+ "Framesize media attribute has invalid dimension separator");
+ return FALSE;
+ }
+
+ dimension = endptr + 1;
+ height = g_ascii_strtoull (dimension, &endptr, 10);
+ if (height <= 0) {
+ GST_ERROR_OBJECT (filter,
+ "Framesize media attribute height out of valid range");
+ return FALSE;
+ } else if (*endptr != '\0') {
+ GST_ERROR_OBJECT (filter,
+ "Framesize media attribute unexpectedly has trailing characters");
+ return FALSE;
+ }
+
+ d = gst_caps_get_structure (srccaps, 0);
+ gst_structure_set (d, "width", G_TYPE_INT, width, "height", G_TYPE_INT,
+ height, NULL);
+
+ return TRUE;
+}
+
gboolean
gst_rtp_h263_depay_setcaps (GstRTPBaseDepayload * filter, GstCaps * caps)
{
GstCaps *srccaps;
GstStructure *structure = gst_caps_get_structure (caps, 0);
gint clock_rate;
+ const gchar *framesize;
+
+ srccaps = gst_caps_new_simple ("video/x-h263",
+ "variant", G_TYPE_STRING, "itu",
+ "h263version", G_TYPE_STRING, "h263", NULL);
if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
clock_rate = 90000; /* default */
filter->clock_rate = clock_rate;
- srccaps = gst_caps_new_simple ("video/x-h263",
- "variant", G_TYPE_STRING, "itu",
- "h263version", G_TYPE_STRING, "h263", NULL);
+ framesize = gst_structure_get_string (structure, "a-framesize");
+ if (framesize != NULL) {
+ if (!gst_rtp_h263_parse_framesize (filter, framesize, srccaps)) {
+ return FALSE;
+ }
+ }
+
gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (filter), srccaps);
gst_caps_unref (srccaps);
diff --git a/gst/rtp/gstrtph263pay.c b/gst/rtp/gstrtph263pay.c
index c664c5b..0b7d24e 100644
--- a/gst/rtp/gstrtph263pay.c
+++ b/gst/rtp/gstrtph263pay.c
@@ -461,13 +461,43 @@
static gboolean
gst_rtp_h263_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
+ GstStructure *s = gst_caps_get_structure (caps, 0);
+ gint width, height;
+ gchar *framesize = NULL;
gboolean res;
+ if (gst_structure_has_field (s, "width") &&
+ gst_structure_has_field (s, "height")) {
+ if (!gst_structure_get_int (s, "width", &width) || width <= 0) {
+ goto invalid_dimension;
+ }
+
+ if (!gst_structure_get_int (s, "height", &height) || height <= 0) {
+ goto invalid_dimension;
+ }
+
+ framesize = g_strdup_printf ("%d-%d", width, height);
+ }
+
payload->pt = GST_RTP_PAYLOAD_H263;
gst_rtp_base_payload_set_options (payload, "video", TRUE, "H263", 90000);
- res = gst_rtp_base_payload_set_outcaps (payload, NULL);
+
+ if (framesize != NULL) {
+ res = gst_rtp_base_payload_set_outcaps (payload,
+ "a-framesize", G_TYPE_STRING, framesize, NULL);
+ } else {
+ res = gst_rtp_base_payload_set_outcaps (payload, NULL);
+ }
+ g_free (framesize);
return res;
+
+ /* ERRORS */
+invalid_dimension:
+ {
+ GST_ERROR_OBJECT (payload, "Invalid width/height from caps");
+ return FALSE;
+ }
}
static void
diff --git a/gst/rtp/gstrtph264pay.c b/gst/rtp/gstrtph264pay.c
index ba5e4b3..665814a 100644
--- a/gst/rtp/gstrtph264pay.c
+++ b/gst/rtp/gstrtph264pay.c
@@ -72,8 +72,7 @@
{
PROP_0,
PROP_SPROP_PARAMETER_SETS,
- PROP_CONFIG_INTERVAL,
- PROP_LAST
+ PROP_CONFIG_INTERVAL
};
#define IS_ACCESS_UNIT(x) (((x) > 0x00) && ((x) < 0x06))
diff --git a/gst/rtp/gstrtpjpegdepay.c b/gst/rtp/gstrtpjpegdepay.c
index 5463283..57055e8 100644
--- a/gst/rtp/gstrtpjpegdepay.c
+++ b/gst/rtp/gstrtpjpegdepay.c
@@ -50,7 +50,6 @@
/*
* "a-framerate = (string) 0.00, "
* "x-framerate = (string) 0.00, "
- * "a-framesize = (string) 1234-1234, "
* "x-dimensions = (string) \"1234,1234\", "
*/
"application/x-rtp, "
@@ -61,7 +60,6 @@
/*
* "a-framerate = (string) 0.00, "
* "x-framerate = (string) 0.00, "
- * "a-framesize = (string) 1234-1234, "
* "x-dimensions = (string) \"1234,1234\""
*/
)
@@ -458,15 +456,6 @@
}
}
- if ((media_attr = gst_structure_get_string (structure, "a-framesize"))) {
- gint w, h;
-
- if (sscanf (media_attr, "%d-%d", &w, &h) == 2) {
- rtpjpegdepay->media_width = w;
- rtpjpegdepay->media_height = h;
- }
- }
-
/* try to get a framerate */
media_attr = gst_structure_get_string (structure, "a-framerate");
if (!media_attr)
diff --git a/gst/rtp/gstrtpjpegpay.c b/gst/rtp/gstrtpjpegpay.c
index 3e820a0..bf4ed55 100644
--- a/gst/rtp/gstrtpjpegpay.c
+++ b/gst/rtp/gstrtpjpegpay.c
@@ -114,8 +114,7 @@
{
PROP_0,
PROP_JPEG_QUALITY,
- PROP_JPEG_TYPE,
- PROP_LAST
+ PROP_JPEG_TYPE
};
enum
@@ -296,7 +295,6 @@
gint num = 0, denom;
gchar *rate = NULL;
gchar *dim = NULL;
- gchar *size;
pay = GST_RTP_JPEG_PAY (basepayload);
@@ -331,8 +329,6 @@
rate = g_strdup_printf ("%f", framerate);
}
- size = g_strdup_printf ("%d-%d", width, height);
-
if (pay->width == 0) {
GST_DEBUG_OBJECT (pay,
"width or height are greater than 2040, adding x-dimensions to caps");
@@ -341,24 +337,21 @@
if (rate != NULL && dim != NULL) {
res = gst_rtp_base_payload_set_outcaps (basepayload, "a-framerate",
- G_TYPE_STRING, rate, "a-framesize", G_TYPE_STRING, size,
- "x-dimensions", G_TYPE_STRING, dim, NULL);
+ G_TYPE_STRING, rate, "x-dimensions", G_TYPE_STRING, dim, NULL);
} else if (rate != NULL && dim == NULL) {
res = gst_rtp_base_payload_set_outcaps (basepayload, "a-framerate",
- G_TYPE_STRING, rate, "a-framesize", G_TYPE_STRING, size, NULL);
+ G_TYPE_STRING, rate, NULL);
} else if (rate == NULL && dim != NULL) {
res = gst_rtp_base_payload_set_outcaps (basepayload, "x-dimensions",
- G_TYPE_STRING, dim, "a-framesize", G_TYPE_STRING, size, NULL);
+ G_TYPE_STRING, dim, NULL);
} else {
- res = gst_rtp_base_payload_set_outcaps (basepayload, "a-framesize",
- G_TYPE_STRING, size, NULL);
+ res = gst_rtp_base_payload_set_outcaps (basepayload, NULL);
}
if (dim != NULL)
g_free (dim);
if (rate != NULL)
g_free (rate);
- g_free (size);
return res;
diff --git a/gst/rtp/gstrtpmp4vpay.c b/gst/rtp/gstrtpmp4vpay.c
index 56cf9f7..1ed1ac5 100644
--- a/gst/rtp/gstrtpmp4vpay.c
+++ b/gst/rtp/gstrtpmp4vpay.c
@@ -58,8 +58,8 @@
enum
{
- ARG_0,
- ARG_CONFIG_INTERVAL
+ PROP_0,
+ PROP_CONFIG_INTERVAL
};
@@ -103,7 +103,7 @@
"Payload MPEG-4 video as RTP packets (RFC 3016)",
"Wim Taymans <wim.taymans@gmail.com>");
- g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_CONFIG_INTERVAL,
+ g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_CONFIG_INTERVAL,
g_param_spec_uint ("config-interval", "Config Send Interval",
"Send Config Insertion Interval in seconds (configuration headers "
"will be multiplexed in the data stream when detected.) (0 = disabled)",
@@ -584,7 +584,7 @@
rtpmp4vpay = GST_RTP_MP4V_PAY (object);
switch (prop_id) {
- case ARG_CONFIG_INTERVAL:
+ case PROP_CONFIG_INTERVAL:
rtpmp4vpay->config_interval = g_value_get_uint (value);
break;
default:
@@ -601,7 +601,7 @@
rtpmp4vpay = GST_RTP_MP4V_PAY (object);
switch (prop_id) {
- case ARG_CONFIG_INTERVAL:
+ case PROP_CONFIG_INTERVAL:
g_value_set_uint (value, rtpmp4vpay->config_interval);
break;
default:
diff --git a/gst/rtp/gstrtppcmadepay.c b/gst/rtp/gstrtppcmadepay.c
index d4fb2af..8f96250 100644
--- a/gst/rtp/gstrtppcmadepay.c
+++ b/gst/rtp/gstrtppcmadepay.c
@@ -36,7 +36,7 @@
enum
{
- ARG_0
+ PROP_0
};
static GstStaticPadTemplate gst_rtp_pcma_depay_sink_template =
diff --git a/gst/rtp/gstrtppcmudepay.c b/gst/rtp/gstrtppcmudepay.c
index 14367a1..e75c282 100644
--- a/gst/rtp/gstrtppcmudepay.c
+++ b/gst/rtp/gstrtppcmudepay.c
@@ -36,7 +36,7 @@
enum
{
- ARG_0
+ PROP_0
};
static GstStaticPadTemplate gst_rtp_pcmu_depay_sink_template =
diff --git a/gst/rtp/gstrtpqcelpdepay.c b/gst/rtp/gstrtpqcelpdepay.c
index b52451c..fc88f4a 100644
--- a/gst/rtp/gstrtpqcelpdepay.c
+++ b/gst/rtp/gstrtpqcelpdepay.c
@@ -45,7 +45,7 @@
enum
{
- ARG_0
+ PROP_0
};
static GstStaticPadTemplate gst_rtp_qcelp_depay_sink_template =
diff --git a/gst/rtp/gstrtpspeexdepay.c b/gst/rtp/gstrtpspeexdepay.c
index 34aef39..efe7357 100644
--- a/gst/rtp/gstrtpspeexdepay.c
+++ b/gst/rtp/gstrtpspeexdepay.c
@@ -36,7 +36,7 @@
enum
{
- ARG_0
+ PROP_0
};
static GstStaticPadTemplate gst_rtp_speex_depay_sink_template =
diff --git a/gst/rtp/gstrtpvp8depay.c b/gst/rtp/gstrtpvp8depay.c
index fed9779..cafdad5 100644
--- a/gst/rtp/gstrtpvp8depay.c
+++ b/gst/rtp/gstrtpvp8depay.c
@@ -24,14 +24,20 @@
#include "gstrtpvp8depay.h"
+#include <gst/video/video.h>
+
+#include <stdio.h>
+
GST_DEBUG_CATEGORY_STATIC (gst_rtp_vp8_depay_debug);
#define GST_CAT_DEFAULT gst_rtp_vp8_depay_debug
static void gst_rtp_vp8_depay_dispose (GObject * object);
static GstBuffer *gst_rtp_vp8_depay_process (GstRTPBaseDepayload * depayload,
GstBuffer * buf);
-static gboolean gst_rtp_vp8_depay_set_caps (GstRTPBaseDepayload * depayload,
- GstCaps * caps);
+static GstStateChangeReturn gst_rtp_vp8_depay_change_state (GstElement *
+ element, GstStateChange transition);
+static gboolean gst_rtp_vp8_depay_handle_event (GstRTPBaseDepayload * depay,
+ GstEvent * event);
G_DEFINE_TYPE (GstRtpVP8Depay, gst_rtp_vp8_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
@@ -78,8 +84,10 @@
object_class->dispose = gst_rtp_vp8_depay_dispose;
+ element_class->change_state = gst_rtp_vp8_depay_change_state;
+
depay_class->process = gst_rtp_vp8_depay_process;
- depay_class->set_caps = gst_rtp_vp8_depay_set_caps;
+ depay_class->handle_event = gst_rtp_vp8_depay_handle_event;
GST_DEBUG_CATEGORY_INIT (gst_rtp_vp8_depay_debug, "rtpvp8depay", 0,
"VP8 Video RTP Depayloader");
@@ -177,11 +185,53 @@
/* mark keyframes */
out = gst_buffer_make_writable (out);
- if ((flag0 & 0x01))
+ if ((flag0 & 0x01)) {
GST_BUFFER_FLAG_SET (out, GST_BUFFER_FLAG_DELTA_UNIT);
- else
+
+ if (!self->caps_sent) {
+ gst_buffer_unref (out);
+ out = NULL;
+ GST_INFO_OBJECT (self, "Dropping inter-frame before intra-frame");
+ gst_pad_push_event (GST_RTP_BASE_DEPAYLOAD_SINKPAD (depay),
+ gst_video_event_new_upstream_force_key_unit (GST_CLOCK_TIME_NONE,
+ TRUE, 0));
+ }
+ } else {
+ GstMapInfo info;
+
GST_BUFFER_FLAG_UNSET (out, GST_BUFFER_FLAG_DELTA_UNIT);
+ if (gst_buffer_map (out, &info, GST_MAP_READ)) {
+ guint profile, width, height;
+
+ profile = (flag0 & 0x0e) >> 1;
+ width = GST_READ_UINT16_LE (info.data + 6) & 0x3fff;
+ height = GST_READ_UINT16_LE (info.data + 8) & 0x3fff;
+ gst_buffer_unmap (out, &info);
+
+ if (G_UNLIKELY (self->last_width != width ||
+ self->last_height != height || self->last_profile != profile)) {
+ gchar profile_str[3];
+ GstCaps *srccaps;
+
+ snprintf (profile_str, 3, "%u", profile);
+ srccaps = gst_caps_new_simple ("video/x-vp8",
+ "framerate", GST_TYPE_FRACTION, 0, 1,
+ "height", G_TYPE_INT, height,
+ "width", G_TYPE_INT, width,
+ "profile", G_TYPE_STRING, profile_str, NULL);
+
+ gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depay), srccaps);
+ gst_caps_unref (srccaps);
+
+ self->caps_sent = TRUE;
+ self->last_width = width;
+ self->last_height = height;
+ self->last_profile = profile;
+ }
+ }
+ }
+
return out;
}
@@ -197,17 +247,45 @@
goto done;
}
-static gboolean
-gst_rtp_vp8_depay_set_caps (GstRTPBaseDepayload * depayload, GstCaps * caps)
+static GstStateChangeReturn
+gst_rtp_vp8_depay_change_state (GstElement * element, GstStateChange transition)
{
- GstCaps *srccaps = gst_caps_new_simple ("video/x-vp8",
- "framerate", GST_TYPE_FRACTION, 0, 1,
- NULL);
+ GstRtpVP8Depay *self = GST_RTP_VP8_DEPAY (element);
- gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
- gst_caps_unref (srccaps);
+ switch (transition) {
+ case GST_STATE_CHANGE_READY_TO_PAUSED:
+ self->last_profile = -1;
+ self->last_height = -1;
+ self->last_width = -1;
+ self->caps_sent = FALSE;
+ break;
+ default:
+ break;
+ }
- return TRUE;
+ return
+ GST_ELEMENT_CLASS (gst_rtp_vp8_depay_parent_class)->change_state (element,
+ transition);
+}
+
+static gboolean
+gst_rtp_vp8_depay_handle_event (GstRTPBaseDepayload * depay, GstEvent * event)
+{
+ GstRtpVP8Depay *self = GST_RTP_VP8_DEPAY (depay);
+
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_FLUSH_STOP:
+ self->last_profile = -1;
+ self->last_height = -1;
+ self->last_width = -1;
+ break;
+ default:
+ break;
+ }
+
+ return
+ GST_RTP_BASE_DEPAYLOAD_CLASS
+ (gst_rtp_vp8_depay_parent_class)->handle_event (depay, event);
}
gboolean
diff --git a/gst/rtp/gstrtpvp8depay.h b/gst/rtp/gstrtpvp8depay.h
index f10b563..258546a 100644
--- a/gst/rtp/gstrtpvp8depay.h
+++ b/gst/rtp/gstrtpvp8depay.h
@@ -53,6 +53,11 @@
GstRTPBaseDepayload parent;
GstAdapter *adapter;
gboolean started;
+
+ gboolean caps_sent;
+ gint last_profile;
+ gint last_width;
+ gint last_height;
};
GType gst_rtp_vp8_depay_get_type (void);
diff --git a/gst/rtpmanager/Makefile.am b/gst/rtpmanager/Makefile.am
index bd8a1e3..ea593ef 100644
--- a/gst/rtpmanager/Makefile.am
+++ b/gst/rtpmanager/Makefile.am
@@ -38,18 +38,3 @@
$(GST_BASE_LIBS) $(GST_LIBS_LIBS)
libgstrtpmanager_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstrtpmanager_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
-
-Android.mk: Makefile.am
- androgenizer \
- -:PROJECT libgstrtpmanager -:SHARED libgstrtpmanager \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstrtpmanager_la_SOURCES) \
- $(nodist_libgstrtpmanager_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstrtpmanager_la_CFLAGS) \
- -:LDFLAGS $(libgstrtpmanager_la_LDFLAGS) \
- $(libgstrtpmanager_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/rtpmanager/Makefile.in b/gst/rtpmanager/Makefile.in
index 37852c9..5e809b6 100644
--- a/gst/rtpmanager/Makefile.in
+++ b/gst/rtpmanager/Makefile.in
@@ -1042,21 +1042,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am
- androgenizer \
- -:PROJECT libgstrtpmanager -:SHARED libgstrtpmanager \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstrtpmanager_la_SOURCES) \
- $(nodist_libgstrtpmanager_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstrtpmanager_la_CFLAGS) \
- -:LDFLAGS $(libgstrtpmanager_la_LDFLAGS) \
- $(libgstrtpmanager_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/rtpmanager/gstrtpbin.c b/gst/rtpmanager/gstrtpbin.c
index 6ba8f7a..f0b8b4e 100644
--- a/gst/rtpmanager/gstrtpbin.c
+++ b/gst/rtpmanager/gstrtpbin.c
@@ -304,15 +304,7 @@
PROP_BUFFER_MODE,
PROP_USE_PIPELINE_CLOCK,
PROP_DO_SYNC_EVENT,
- PROP_DO_RETRANSMISSION,
- PROP_LAST
-};
-
-enum
-{
- GST_RTP_BIN_RTCP_SYNC_ALWAYS,
- GST_RTP_BIN_RTCP_SYNC_INITIAL,
- GST_RTP_BIN_RTCP_SYNC_RTP
+ PROP_DO_RETRANSMISSION
};
#define GST_RTP_BIN_RTCP_SYNC_TYPE (gst_rtp_bin_rtcp_sync_get_type())
diff --git a/gst/rtpmanager/gstrtpbin.h b/gst/rtpmanager/gstrtpbin.h
index 4366535..fcad2a5 100644
--- a/gst/rtpmanager/gstrtpbin.h
+++ b/gst/rtpmanager/gstrtpbin.h
@@ -36,6 +36,13 @@
#define GST_IS_RTP_BIN_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_BIN))
+typedef enum
+{
+ GST_RTP_BIN_RTCP_SYNC_ALWAYS,
+ GST_RTP_BIN_RTCP_SYNC_INITIAL,
+ GST_RTP_BIN_RTCP_SYNC_RTP
+} GstRTCPSync;
+
typedef struct _GstRtpBin GstRtpBin;
typedef struct _GstRtpBinClass GstRtpBinClass;
typedef struct _GstRtpBinPrivate GstRtpBinPrivate;
diff --git a/gst/rtpmanager/gstrtpjitterbuffer.c b/gst/rtpmanager/gstrtpjitterbuffer.c
index 999b24c..547bef5 100644
--- a/gst/rtpmanager/gstrtpjitterbuffer.c
+++ b/gst/rtpmanager/gstrtpjitterbuffer.c
@@ -128,12 +128,14 @@
#define DEFAULT_MODE RTP_JITTER_BUFFER_MODE_SLAVE
#define DEFAULT_PERCENT 0
#define DEFAULT_DO_RETRANSMISSION FALSE
+#define DEFAULT_RTX_NEXT_SEQNUM TRUE
#define DEFAULT_RTX_DELAY -1
#define DEFAULT_RTX_MIN_DELAY 0
#define DEFAULT_RTX_DELAY_REORDER 3
#define DEFAULT_RTX_RETRY_TIMEOUT -1
#define DEFAULT_RTX_MIN_RETRY_TIMEOUT -1
#define DEFAULT_RTX_RETRY_PERIOD -1
+#define DEFAULT_RTX_MAX_RETRIES -1
#define DEFAULT_AUTO_RTX_DELAY (20 * GST_MSECOND)
#define DEFAULT_AUTO_RTX_TIMEOUT (40 * GST_MSECOND)
@@ -148,14 +150,15 @@
PROP_MODE,
PROP_PERCENT,
PROP_DO_RETRANSMISSION,
+ PROP_RTX_NEXT_SEQNUM,
PROP_RTX_DELAY,
PROP_RTX_MIN_DELAY,
PROP_RTX_DELAY_REORDER,
PROP_RTX_RETRY_TIMEOUT,
PROP_RTX_MIN_RETRY_TIMEOUT,
PROP_RTX_RETRY_PERIOD,
- PROP_STATS,
- PROP_LAST
+ PROP_RTX_MAX_RETRIES,
+ PROP_STATS
};
#define JBUF_LOCK(priv) (g_mutex_lock (&(priv)->jbuf_lock))
@@ -244,12 +247,14 @@
gint64 ts_offset;
gboolean do_lost;
gboolean do_retransmission;
+ gboolean rtx_next_seqnum;
gint rtx_delay;
guint rtx_min_delay;
gint rtx_delay_reorder;
gint rtx_retry_timeout;
gint rtx_min_retry_timeout;
gint rtx_retry_period;
+ gint rtx_max_retries;
/* the last seqnum we pushed out */
guint32 last_popped_seqnum;
@@ -531,6 +536,23 @@
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
+ * GstRtpJitterBuffer:rtx-next-seqnum
+ *
+ * Estimate when the next packet should arrive and schedule a retransmission
+ * request for it.
+ * This is, when packet N arrives, a GstRTPRetransmission event is schedule
+ * for packet N+1. So it will be requested if it does not arrive at the expected time.
+ * The expected time is calculated using the dts of N and the packet spacing.
+ *
+ * Since: 1.6
+ */
+ g_object_class_install_property (gobject_class, PROP_RTX_NEXT_SEQNUM,
+ g_param_spec_boolean ("rtx-next-seqnum", "RTX next seqnum",
+ "Estimate when the next packet should arrive and schedule a "
+ "retransmission request for it.",
+ DEFAULT_RTX_NEXT_SEQNUM, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ /**
* GstRtpJitterBuffer:rtx-delay:
*
* When a packet did not arrive at the expected time, wait this extra amount
@@ -624,15 +646,56 @@
"(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_PERIOD,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
+ * GstRtpJitterBuffer:rtx-max-retries:
+ *
+ * The maximum number of retries to request a retransmission.
+ *
+ * This implies that as maximum (rtx-max-retries + 1) retransmissions will be requested.
+ * When -1 is used, the number of retransmission request will not be limited.
+ *
+ * Since: 1.6
+ */
+ g_object_class_install_property (gobject_class, PROP_RTX_MAX_RETRIES,
+ g_param_spec_int ("rtx-max-retries", "RTX Max Retries",
+ "The maximum number of retries to request a retransmission. "
+ "(-1 not limited)", -1, G_MAXINT, DEFAULT_RTX_MAX_RETRIES,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ /**
* GstRtpJitterBuffer:stats:
*
* Various jitterbuffer statistics. This property returns a GstStructure
* with name application/x-rtp-jitterbuffer-stats with the following fields:
*
- * "rtx-count" G_TYPE_UINT64 The number of retransmissions requested
- * "rtx-success-count" G_TYPE_UINT64 The number of successful retransmissions
- * "rtx-per-packet" G_TYPE_DOUBLE Average number of RTX per packet
- * "rtx-rtt" G_TYPE_UINT64 Average round trip time per RTX
+ * <itemizedlist>
+ * <listitem>
+ * <para>
+ * #guint64
+ * <classname>"rtx-count"</classname>:
+ * the number of retransmissions requested.
+ * </para>
+ * </listitem>
+ * <listitem>
+ * <para>
+ * #guint64
+ * <classname>"rtx-success-count"</classname>:
+ * the number of successful retransmissions.
+ * </para>
+ * </listitem>
+ * <listitem>
+ * <para>
+ * #gdouble
+ * <classname>"rtx-per-packet"</classname>:
+ * average number of RTX per packet.
+ * </para>
+ * </listitem>
+ * <listitem>
+ * <para>
+ * #guint64
+ * <classname>"rtx-rtt"</classname>:
+ * average round trip time per RTX.
+ * </para>
+ * </listitem>
+ * </itemizedlist>
*
* Since: 1.4
*/
@@ -750,12 +813,14 @@
priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
priv->do_lost = DEFAULT_DO_LOST;
priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
+ priv->rtx_next_seqnum = DEFAULT_RTX_NEXT_SEQNUM;
priv->rtx_delay = DEFAULT_RTX_DELAY;
priv->rtx_min_delay = DEFAULT_RTX_MIN_DELAY;
priv->rtx_delay_reorder = DEFAULT_RTX_DELAY_REORDER;
priv->rtx_retry_timeout = DEFAULT_RTX_RETRY_TIMEOUT;
priv->rtx_min_retry_timeout = DEFAULT_RTX_MIN_RETRY_TIMEOUT;
priv->rtx_retry_period = DEFAULT_RTX_RETRY_PERIOD;
+ priv->rtx_max_retries = DEFAULT_RTX_MAX_RETRIES;
priv->last_dts = -1;
priv->last_rtptime = -1;
@@ -1849,11 +1914,13 @@
GstClockTime delay;
if (priv->rtx_delay == -1) {
- if (priv->avg_jitter == 0)
+ if (priv->avg_jitter == 0 && priv->packet_spacing == 0) {
delay = DEFAULT_AUTO_RTX_DELAY;
- else
- /* jitter is in nanoseconds, 2x jitter is a good margin */
- delay = priv->avg_jitter * 2;
+ } else {
+ /* jitter is in nanoseconds, maximum of 2x jitter and half the
+ * packet spacing is a good margin */
+ delay = MAX (priv->avg_jitter * 2, priv->packet_spacing / 2);
+ }
} else {
delay = priv->rtx_delay * GST_MSECOND;
}
@@ -1910,7 +1977,7 @@
}
do_next_seqnum = do_next_seqnum && priv->packet_spacing > 0
- && priv->do_retransmission;
+ && priv->do_retransmission && priv->rtx_next_seqnum;
if (timer && timer->type != TIMER_TYPE_DEADLINE) {
if (timer->num_rtx_retry > 0) {
@@ -1987,9 +2054,28 @@
if (priv->ips_rtptime != rtptime) {
/* rtptime changed, check dts diff */
if (priv->ips_dts != -1 && dts != -1 && dts > priv->ips_dts) {
- priv->packet_spacing = dts - priv->ips_dts;
+ GstClockTime new_packet_spacing = dts - priv->ips_dts;
+ GstClockTime old_packet_spacing = priv->packet_spacing;
+
+ /* Biased towards bigger packet spacings to prevent
+ * too many unneeded retransmission requests for next
+ * packets that just arrive a little later than we would
+ * expect */
+ if (old_packet_spacing > new_packet_spacing)
+ priv->packet_spacing =
+ (new_packet_spacing + 3 * old_packet_spacing) / 4;
+ else if (old_packet_spacing > 0)
+ priv->packet_spacing =
+ (3 * new_packet_spacing + old_packet_spacing) / 4;
+ else
+ priv->packet_spacing = new_packet_spacing;
+
GST_DEBUG_OBJECT (jitterbuffer,
- "new packet spacing %" GST_TIME_FORMAT,
+ "new packet spacing %" GST_TIME_FORMAT
+ " old packet spacing %" GST_TIME_FORMAT
+ " combined to %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (new_packet_spacing),
+ GST_TIME_ARGS (old_packet_spacing),
GST_TIME_ARGS (priv->packet_spacing));
}
priv->ips_rtptime = rtptime;
@@ -2004,6 +2090,7 @@
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
GstClockTime total_duration, duration, expected_dts;
TimerType type;
+ guint lost_packets = 0;
GST_DEBUG_OBJECT (jitterbuffer,
"dts %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
@@ -2025,7 +2112,6 @@
if (total_duration > priv->latency_ns) {
GstClockTime gap_time;
- guint lost_packets;
gap_time = total_duration - priv->latency_ns;
@@ -2052,7 +2138,7 @@
priv->last_in_dts += gap_time;
}
- expected_dts = priv->last_in_dts + duration;
+ expected_dts = priv->last_in_dts + (lost_packets + 1) * duration;
if (priv->do_retransmission) {
TimerData *timer;
@@ -2063,7 +2149,7 @@
GstClockTime timeout = timer->timeout;
timer->duration = duration;
- if (timeout > expected_dts) {
+ if (timeout > (expected_dts + timer->rtx_retry)) {
GstClockTime delay = timeout - expected_dts - timer->rtx_retry;
reschedule_timer (jitterbuffer, timer, timer->seqnum, expected_dts,
delay, TRUE);
@@ -2300,6 +2386,7 @@
(GFunc) free_item_and_retain_events, &events);
rtp_jitter_buffer_reset_skew (priv->jbuf);
remove_all_timers (jitterbuffer);
+ priv->discont = TRUE;
priv->last_popped_seqnum = -1;
priv->next_seqnum = seqnum;
do_next_seqnum = TRUE;
@@ -2886,8 +2973,9 @@
GST_TIME_FORMAT ", retry %" GST_TIME_FORMAT ", num_retry %u",
GST_TIME_ARGS (timer->rtx_base), GST_TIME_ARGS (timer->rtx_delay),
GST_TIME_ARGS (timer->rtx_retry), timer->num_rtx_retry);
-
- if (timer->rtx_retry + timer->rtx_delay > rtx_retry_period) {
+ if ((priv->rtx_max_retries != -1
+ && timer->num_rtx_retry >= priv->rtx_max_retries)
+ || (timer->rtx_retry + timer->rtx_delay > rtx_retry_period)) {
GST_DEBUG_OBJECT (jitterbuffer, "reschedule as LOST timer");
/* too many retransmission request, we now convert the timer
* to a lost timer, leave the num_rtx_retry as it is for stats */
@@ -3587,6 +3675,11 @@
priv->do_retransmission = g_value_get_boolean (value);
JBUF_UNLOCK (priv);
break;
+ case PROP_RTX_NEXT_SEQNUM:
+ JBUF_LOCK (priv);
+ priv->rtx_next_seqnum = g_value_get_boolean (value);
+ JBUF_UNLOCK (priv);
+ break;
case PROP_RTX_DELAY:
JBUF_LOCK (priv);
priv->rtx_delay = g_value_get_int (value);
@@ -3617,6 +3710,11 @@
priv->rtx_retry_period = g_value_get_int (value);
JBUF_UNLOCK (priv);
break;
+ case PROP_RTX_MAX_RETRIES:
+ JBUF_LOCK (priv);
+ priv->rtx_max_retries = g_value_get_int (value);
+ JBUF_UNLOCK (priv);
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
@@ -3678,6 +3776,11 @@
g_value_set_boolean (value, priv->do_retransmission);
JBUF_UNLOCK (priv);
break;
+ case PROP_RTX_NEXT_SEQNUM:
+ JBUF_LOCK (priv);
+ g_value_set_boolean (value, priv->rtx_next_seqnum);
+ JBUF_UNLOCK (priv);
+ break;
case PROP_RTX_DELAY:
JBUF_LOCK (priv);
g_value_set_int (value, priv->rtx_delay);
@@ -3708,6 +3811,11 @@
g_value_set_int (value, priv->rtx_retry_period);
JBUF_UNLOCK (priv);
break;
+ case PROP_RTX_MAX_RETRIES:
+ JBUF_LOCK (priv);
+ g_value_set_int (value, priv->rtx_max_retries);
+ JBUF_UNLOCK (priv);
+ break;
case PROP_STATS:
g_value_take_boxed (value,
gst_rtp_jitter_buffer_create_stats (jitterbuffer));
diff --git a/gst/rtpmanager/gstrtpmux.c b/gst/rtpmanager/gstrtpmux.c
index 5cddd04..db3585b 100644
--- a/gst/rtpmanager/gstrtpmux.c
+++ b/gst/rtpmanager/gstrtpmux.c
@@ -35,7 +35,7 @@
* <refsect2>
* <title>Example pipelines</title>
* |[
- * gst-launch rtpmux name=mux ! udpsink host=127.0.0.1 port=8888 \
+ * gst-launch-1.0 rtpmux name=mux ! udpsink host=127.0.0.1 port=8888 \
* alsasrc ! alawenc ! rtppcmapay ! \
* application/x-rtp, payload=8, rate=8000 ! mux.sink_0 \
* audiotestsrc is-live=1 ! \
@@ -63,7 +63,7 @@
enum
{
- ARG_0,
+ PROP_0,
PROP_TIMESTAMP_OFFSET,
PROP_SEQNUM_OFFSET,
PROP_SEQNUM,
diff --git a/gst/rtpmanager/gstrtprtxqueue.c b/gst/rtpmanager/gstrtprtxqueue.c
index 209aa76..8e98543 100644
--- a/gst/rtpmanager/gstrtprtxqueue.c
+++ b/gst/rtpmanager/gstrtprtxqueue.c
@@ -44,8 +44,7 @@
{
PROP_0,
PROP_MAX_SIZE_TIME,
- PROP_MAX_SIZE_PACKETS,
- PROP_LAST
+ PROP_MAX_SIZE_PACKETS
};
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
@@ -64,6 +63,8 @@
GstEvent * event);
static GstFlowReturn gst_rtp_rtx_queue_chain (GstPad * pad, GstObject * parent,
GstBuffer * buffer);
+static GstFlowReturn gst_rtp_rtx_queue_chain_list (GstPad * pad,
+ GstObject * parent, GstBufferList * list);
static GstStateChangeReturn gst_rtp_rtx_queue_change_state (GstElement *
element, GstStateChange transition);
@@ -159,6 +160,8 @@
GST_PAD_SET_PROXY_ALLOCATION (rtx->sinkpad);
gst_pad_set_chain_function (rtx->sinkpad,
GST_DEBUG_FUNCPTR (gst_rtp_rtx_queue_chain));
+ gst_pad_set_chain_list_function (rtx->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_rtp_rtx_queue_chain_list));
gst_element_add_pad (GST_ELEMENT (rtx), rtx->sinkpad);
rtx->queue = g_queue_new ();
@@ -246,6 +249,16 @@
gst_pad_push (rtx->srcpad, buffer);
}
+/* Must be called with rtx->lock */
+static void
+shrink_queue (GstRTPRtxQueue * rtx)
+{
+ if (rtx->max_size_packets) {
+ while (g_queue_get_length (rtx->queue) > rtx->max_size_packets)
+ gst_buffer_unref (g_queue_pop_tail (rtx->queue));
+ }
+}
+
static GstFlowReturn
gst_rtp_rtx_queue_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
{
@@ -257,11 +270,7 @@
g_mutex_lock (&rtx->lock);
g_queue_push_head (rtx->queue, gst_buffer_ref (buffer));
-
- if (rtx->max_size_packets) {
- while (g_queue_get_length (rtx->queue) > rtx->max_size_packets)
- gst_buffer_unref (g_queue_pop_tail (rtx->queue));
- }
+ shrink_queue (rtx);
pending = rtx->pending;
rtx->pending = NULL;
@@ -275,6 +284,42 @@
return ret;
}
+static gboolean
+push_to_queue (GstBuffer ** buffer, guint idx, gpointer user_data)
+{
+ GQueue *queue = user_data;
+
+ g_queue_push_head (queue, gst_buffer_ref (*buffer));
+
+ return TRUE;
+}
+
+static GstFlowReturn
+gst_rtp_rtx_queue_chain_list (GstPad * pad, GstObject * parent,
+ GstBufferList * list)
+{
+ GstRTPRtxQueue *rtx;
+ GstFlowReturn ret;
+ GList *pending;
+
+ rtx = GST_RTP_RTX_QUEUE (parent);
+
+ g_mutex_lock (&rtx->lock);
+ gst_buffer_list_foreach (list, push_to_queue, rtx->queue);
+ shrink_queue (rtx);
+
+ pending = rtx->pending;
+ rtx->pending = NULL;
+ g_mutex_unlock (&rtx->lock);
+
+ g_list_foreach (pending, (GFunc) do_push, rtx);
+ g_list_free (pending);
+
+ ret = gst_pad_push_list (rtx->srcpad, list);
+
+ return ret;
+}
+
static void
gst_rtp_rtx_queue_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
diff --git a/gst/rtpmanager/gstrtprtxreceive.c b/gst/rtpmanager/gstrtprtxreceive.c
index 32933a1..71b8023 100644
--- a/gst/rtpmanager/gstrtprtxreceive.c
+++ b/gst/rtpmanager/gstrtprtxreceive.c
@@ -133,8 +133,7 @@
PROP_PAYLOAD_TYPE_MAP,
PROP_NUM_RTX_REQUESTS,
PROP_NUM_RTX_PACKETS,
- PROP_NUM_RTX_ASSOC_PACKETS,
- PROP_LAST
+ PROP_NUM_RTX_ASSOC_PACKETS
};
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
@@ -310,11 +309,11 @@
guint ssrc = 0;
gpointer ssrc2 = 0;
- /* retrieve seqnum of the packet that need to be restransmisted */
+ /* retrieve seqnum of the packet that need to be retransmitted */
if (!gst_structure_get_uint (s, "seqnum", &seqnum))
seqnum = -1;
- /* retrieve ssrc of the packet that need to be restransmisted
+ /* retrieve ssrc of the packet that need to be retransmitted
* it's usefull when reconstructing the original packet from the rtx packet */
if (!gst_structure_get_uint (s, "ssrc", &ssrc))
ssrc = -1;
@@ -329,7 +328,7 @@
++rtx->num_rtx_requests;
/* First, we lookup in our map to see if we have already associate this
- * master stream ssrc with its retransmisted stream.
+ * master stream ssrc with its retransmitted stream.
* Every ssrc are unique so we can use the same hash table
* for both retrieving the ssrc1 from ssrc2 and also ssrc2 from ssrc1
*/
@@ -389,17 +388,19 @@
retransmit:
/* the request has not been already considered
* insert it for the first time */
- GST_DEBUG_OBJECT (rtx,
- "packet number %" G_GUINT32_FORMAT " of master stream %"
- G_GUINT32_FORMAT " needs to be retransmited", seqnum, ssrc);
g_hash_table_insert (rtx->seqnum_ssrc1_map,
GUINT_TO_POINTER (seqnum),
ssrc_assoc_new (ssrc, rtx->last_time));
}
}
+ GST_DEBUG_OBJECT (rtx,
+ "packet number %" G_GUINT32_FORMAT " of master stream %"
+ G_GUINT32_FORMAT " needs to be retransmitted", seqnum, ssrc);
+
GST_OBJECT_UNLOCK (rtx);
}
+
/* Transfer event upstream so that the request can acutally by translated
* through gstrtpsession through the network */
res = gst_pad_event_default (pad, parent, event);
@@ -532,13 +533,13 @@
} else {
SsrcAssoc *assoc;
- /* the current retransmisted packet has its rtx stream not already
+ /* the current retransmitted packet has its rtx stream not already
* associated to a master stream, so retrieve it from our request
* history */
if (g_hash_table_lookup_extended (rtx->seqnum_ssrc1_map,
GUINT_TO_POINTER (orign_seqnum), NULL, (gpointer *) & assoc)) {
GST_DEBUG_OBJECT (rtx,
- "associate retransmisted stream %" G_GUINT32_FORMAT
+ "associate retransmitted stream %" G_GUINT32_FORMAT
" to master stream %" G_GUINT32_FORMAT " thanks to packet %"
G_GUINT16_FORMAT "", ssrc, assoc->ssrc, orign_seqnum);
ssrc1 = GUINT_TO_POINTER (assoc->ssrc);
diff --git a/gst/rtpmanager/gstrtprtxsend.c b/gst/rtpmanager/gstrtprtxsend.c
index 922fad3..d9cc69f 100644
--- a/gst/rtpmanager/gstrtprtxsend.c
+++ b/gst/rtpmanager/gstrtprtxsend.c
@@ -61,8 +61,7 @@
PROP_MAX_SIZE_TIME,
PROP_MAX_SIZE_PACKETS,
PROP_NUM_RTX_REQUESTS,
- PROP_NUM_RTX_PACKETS,
- PROP_LAST
+ PROP_NUM_RTX_PACKETS
};
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
@@ -86,6 +85,8 @@
GstEvent * event);
static GstFlowReturn gst_rtp_rtx_send_chain (GstPad * pad, GstObject * parent,
GstBuffer * buffer);
+static GstFlowReturn gst_rtp_rtx_send_chain_list (GstPad * pad,
+ GstObject * parent, GstBufferList * list);
static void gst_rtp_rtx_send_src_loop (GstRtpRtxSend * rtx);
static gboolean gst_rtp_rtx_send_activate_mode (GstPad * pad,
@@ -119,7 +120,7 @@
typedef struct
{
guint32 rtx_ssrc;
- guint16 next_seqnum;
+ guint16 seqnum_base, next_seqnum;
gint clock_rate;
/* history of rtp packets */
@@ -132,7 +133,7 @@
SSRCRtxData *data = g_slice_new0 (SSRCRtxData);
data->rtx_ssrc = rtx_ssrc;
- data->next_seqnum = g_random_int_range (0, G_MAXUINT16);
+ data->next_seqnum = data->seqnum_base = g_random_int_range (0, G_MAXUINT16);
data->queue = g_sequence_new ((GDestroyNotify) buffer_queue_item_free);
return data;
@@ -258,6 +259,8 @@
GST_DEBUG_FUNCPTR (gst_rtp_rtx_send_sink_event));
gst_pad_set_chain_function (rtx->sinkpad,
GST_DEBUG_FUNCPTR (gst_rtp_rtx_send_chain));
+ gst_pad_set_chain_list_function (rtx->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_rtp_rtx_send_chain_list));
gst_element_add_pad (GST_ELEMENT (rtx), rtx->sinkpad);
rtx->queue = gst_data_queue_new (gst_rtp_rtx_send_queue_check_full, NULL,
@@ -426,6 +429,9 @@
gst_rtp_buffer_set_padding (&new_rtp, FALSE);
gst_rtp_buffer_unmap (&new_rtp);
+ /* Copy over timestamps */
+ gst_buffer_copy_into (new_buffer, buffer, GST_BUFFER_COPY_TIMESTAMPS, 0, -1);
+
return new_buffer;
}
@@ -456,11 +462,11 @@
guint ssrc = 0;
GstBuffer *rtx_buf = NULL;
- /* retrieve seqnum of the packet that need to be restransmisted */
+ /* retrieve seqnum of the packet that need to be retransmitted */
if (!gst_structure_get_uint (s, "seqnum", &seqnum))
seqnum = -1;
- /* retrieve ssrc of the packet that need to be restransmisted */
+ /* retrieve ssrc of the packet that need to be retransmitted */
if (!gst_structure_get_uint (s, "ssrc", &ssrc))
ssrc = -1;
@@ -586,22 +592,52 @@
GstCaps *caps;
GstStructure *s;
guint ssrc;
+ gint payload;
+ gpointer rtx_payload;
SSRCRtxData *data;
gst_event_parse_caps (event, &caps);
- g_assert (gst_caps_is_fixed (caps));
s = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_uint (s, "ssrc", &ssrc))
ssrc = -1;
+ if (!gst_structure_get_int (s, "payload", &payload))
+ payload = -1;
+
+ if (payload == -1)
+ GST_WARNING_OBJECT (rtx, "No payload in caps");
GST_OBJECT_LOCK (rtx);
data = gst_rtp_rtx_send_get_ssrc_data (rtx, ssrc);
+ if (!g_hash_table_lookup_extended (rtx->rtx_pt_map,
+ GUINT_TO_POINTER (payload), NULL, &rtx_payload))
+ rtx_payload = GINT_TO_POINTER (-1);
+
+ if (GPOINTER_TO_INT (rtx_payload) == -1 && payload != -1)
+ GST_WARNING_OBJECT (rtx, "Payload %d not in rtx-pt-map", payload);
+
+ GST_DEBUG_OBJECT (rtx,
+ "got caps for payload: %d->%d, ssrc: %u->%d: %" GST_PTR_FORMAT,
+ payload, GPOINTER_TO_INT (rtx_payload), ssrc, data->rtx_ssrc, caps);
+
gst_structure_get_int (s, "clock-rate", &data->clock_rate);
+ /* The session might need to know the RTX ssrc */
+ caps = gst_caps_copy (caps);
+ gst_caps_set_simple (caps, "rtx-ssrc", G_TYPE_UINT, data->rtx_ssrc,
+ "rtx-seqnum-offset", G_TYPE_UINT, data->seqnum_base, NULL);
+
+ if (GPOINTER_TO_INT (rtx_payload) != -1)
+ gst_caps_set_simple (caps, "rtx-payload", G_TYPE_INT,
+ GPOINTER_TO_INT (rtx_payload), NULL);
+
GST_DEBUG_OBJECT (rtx, "got clock-rate from caps: %d for ssrc: %u",
data->clock_rate, ssrc);
GST_OBJECT_UNLOCK (rtx);
+
+ gst_event_unref (event);
+ event = gst_event_new_caps (caps);
+ gst_caps_unref (caps);
break;
}
default:
@@ -640,11 +676,10 @@
return (guint32) gst_util_uint64_scale_int (result, 1000, data->clock_rate);
}
-static GstFlowReturn
-gst_rtp_rtx_send_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
+/* Must be called with lock */
+static void
+process_buffer (GstRtpRtxSend * rtx, GstBuffer * buffer)
{
- GstRtpRtxSend *rtx = GST_RTP_RTX_SEND (parent);
- GstFlowReturn ret = GST_FLOW_ERROR;
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
BufferQueueItem *item;
SSRCRtxData *data;
@@ -660,7 +695,9 @@
rtptime = gst_rtp_buffer_get_timestamp (&rtp);
gst_rtp_buffer_unmap (&rtp);
- GST_OBJECT_LOCK (rtx);
+ GST_LOG_OBJECT (rtx,
+ "Processing buffer seqnum: %" G_GUINT16_FORMAT ", ssrc: %"
+ G_GUINT32_FORMAT, seqnum, ssrc);
/* do not store the buffer if it's payload type is unknown */
if (g_hash_table_contains (rtx->rtx_pt_map, GUINT_TO_POINTER (payload_type))) {
@@ -683,14 +720,41 @@
g_sequence_remove (g_sequence_get_begin_iter (data->queue));
}
}
+}
+static GstFlowReturn
+gst_rtp_rtx_send_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
+{
+ GstRtpRtxSend *rtx = GST_RTP_RTX_SEND (parent);
+ GstFlowReturn ret;
+
+ GST_OBJECT_LOCK (rtx);
+ process_buffer (rtx, buffer);
+ GST_OBJECT_UNLOCK (rtx);
+ ret = gst_pad_push (rtx->srcpad, buffer);
+
+ return ret;
+}
+
+static gboolean
+process_buffer_from_list (GstBuffer ** buffer, guint idx, gpointer user_data)
+{
+ process_buffer (user_data, *buffer);
+ return TRUE;
+}
+
+static GstFlowReturn
+gst_rtp_rtx_send_chain_list (GstPad * pad, GstObject * parent,
+ GstBufferList * list)
+{
+ GstRtpRtxSend *rtx = GST_RTP_RTX_SEND (parent);
+ GstFlowReturn ret;
+
+ GST_OBJECT_LOCK (rtx);
+ gst_buffer_list_foreach (list, process_buffer_from_list, rtx);
GST_OBJECT_UNLOCK (rtx);
- GST_LOG_OBJECT (rtx,
- "push seqnum: %" G_GUINT16_FORMAT ", ssrc: %" G_GUINT32_FORMAT, seqnum,
- ssrc);
-
- ret = gst_pad_push (rtx->srcpad, buffer);
+ ret = gst_pad_push_list (rtx->srcpad, list);
return ret;
}
diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c
index 3a968a5..8838aaf 100644
--- a/gst/rtpmanager/gstrtpsession.c
+++ b/gst/rtpmanager/gstrtpsession.c
@@ -191,8 +191,8 @@
LAST_SIGNAL
};
-#define DEFAULT_BANDWIDTH RTP_STATS_BANDWIDTH
-#define DEFAULT_RTCP_FRACTION (RTP_STATS_BANDWIDTH * RTP_STATS_RTCP_FRACTION)
+#define DEFAULT_BANDWIDTH 0
+#define DEFAULT_RTCP_FRACTION RTP_STATS_RTCP_FRACTION
#define DEFAULT_RTCP_RR_BANDWIDTH -1
#define DEFAULT_RTCP_RS_BANDWIDTH -1
#define DEFAULT_SDES NULL
@@ -216,8 +216,7 @@
PROP_USE_PIPELINE_CLOCK,
PROP_RTCP_MIN_INTERVAL,
PROP_PROBATION,
- PROP_STATS,
- PROP_LAST
+ PROP_STATS
};
#define GST_RTP_SESSION_GET_PRIVATE(obj) \
@@ -911,7 +910,7 @@
GST_RTP_SESSION_LOCK (rtpsession);
while (rtpsession->priv->wait_send) {
- GST_LOG_OBJECT (rtpsession, "waiting for RTP thread");
+ GST_LOG_OBJECT (rtpsession, "waiting for getting started");
GST_RTP_SESSION_WAIT (rtpsession);
GST_LOG_OBJECT (rtpsession, "signaled...");
}
@@ -1721,6 +1720,14 @@
GST_LOG_OBJECT (rtpsession, "received RTP packet");
+ GST_RTP_SESSION_LOCK (rtpsession);
+ if (rtpsession->priv->wait_send) {
+ GST_LOG_OBJECT (rtpsession, "signal RTCP thread");
+ rtpsession->priv->wait_send = FALSE;
+ GST_RTP_SESSION_SIGNAL (rtpsession);
+ }
+ GST_RTP_SESSION_UNLOCK (rtpsession);
+
/* get NTP time when this packet was captured, this depends on the timestamp. */
timestamp = GST_BUFFER_TIMESTAMP (buffer);
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
@@ -1765,6 +1772,18 @@
GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_SEGMENT:
+ /* Make sure that the sync_src pad has caps before the segment event.
+ * Otherwise we might get a segment event before caps from the receive
+ * RTCP pad, and then later when receiving RTCP packets will set caps.
+ * This will results in a sticky event misordering warning
+ */
+ if (!gst_pad_has_current_caps (rtpsession->sync_src)) {
+ GstCaps *caps = gst_caps_new_empty_simple ("application/x-rtcp");
+ gst_pad_set_caps (rtpsession->sync_src, caps);
+ gst_caps_unref (caps);
+ }
+ /* fall through */
default:
ret = gst_pad_push_event (rtpsession->sync_src, event);
break;
@@ -1790,6 +1809,14 @@
GST_LOG_OBJECT (rtpsession, "received RTCP packet");
+ GST_RTP_SESSION_LOCK (rtpsession);
+ if (rtpsession->priv->wait_send) {
+ GST_LOG_OBJECT (rtpsession, "signal RTCP thread");
+ rtpsession->priv->wait_send = FALSE;
+ GST_RTP_SESSION_SIGNAL (rtpsession);
+ }
+ GST_RTP_SESSION_UNLOCK (rtpsession);
+
current_time = gst_clock_get_time (priv->sysclock);
get_current_times (rtpsession, NULL, &ntpnstime);
diff --git a/gst/rtpmanager/rtpjitterbuffer.c b/gst/rtpmanager/rtpjitterbuffer.c
index 1292d7d..f41b799 100644
--- a/gst/rtpmanager/rtpjitterbuffer.c
+++ b/gst/rtpmanager/rtpjitterbuffer.c
@@ -445,7 +445,7 @@
/* need resync, lock on to time and gstrtptime if we can, otherwise we
* do with the previous values */
if (G_UNLIKELY (jbuf->need_resync && time != -1)) {
- GST_WARNING ("resync to time %" GST_TIME_FORMAT ", rtptime %"
+ GST_INFO ("resync to time %" GST_TIME_FORMAT ", rtptime %"
GST_TIME_FORMAT, GST_TIME_ARGS (time), GST_TIME_ARGS (gstrtptime));
rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, FALSE);
send_diff = 0;
diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c
index 486ac3a..0acac15 100644
--- a/gst/rtpmanager/rtpsession.c
+++ b/gst/rtpmanager/rtpsession.c
@@ -55,8 +55,8 @@
};
#define DEFAULT_INTERNAL_SOURCE NULL
-#define DEFAULT_BANDWIDTH RTP_STATS_BANDWIDTH
-#define DEFAULT_RTCP_FRACTION (RTP_STATS_RTCP_FRACTION * RTP_STATS_BANDWIDTH)
+#define DEFAULT_BANDWIDTH 0.0
+#define DEFAULT_RTCP_FRACTION RTP_STATS_RTCP_FRACTION
#define DEFAULT_RTCP_RR_BANDWIDTH -1
#define DEFAULT_RTCP_RS_BANDWIDTH -1
#define DEFAULT_RTCP_MTU 1400
@@ -88,8 +88,7 @@
PROP_RTCP_FEEDBACK_RETENTION_WINDOW,
PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
PROP_PROBATION,
- PROP_STATS,
- PROP_LAST
+ PROP_STATS
};
/* update average packet size */
@@ -2701,6 +2700,15 @@
rtp_source_update_caps (source, caps);
g_object_unref (source);
}
+
+ if (gst_structure_get_uint (s, "rtx-ssrc", &ssrc)) {
+ source =
+ obtain_internal_source (sess, ssrc, &created, GST_CLOCK_TIME_NONE);
+ if (source) {
+ rtp_source_update_caps (source, caps);
+ g_object_unref (source);
+ }
+ }
RTP_SESSION_UNLOCK (sess);
}
}
@@ -2793,9 +2801,8 @@
g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
(GHFunc) add_bitrates, &bandwidth);
- bandwidth /= 8.0;
}
- if (bandwidth < 8000)
+ if (bandwidth < RTP_STATS_BANDWIDTH)
bandwidth = RTP_STATS_BANDWIDTH;
rtp_stats_set_bandwidths (&sess->stats, bandwidth,
@@ -3555,7 +3562,7 @@
/* Apply the rules from RFC 4585 section 3.5.3 */
if (stats->min_interval != 0 && !sess->first_rtcp) {
GstClockTime T_rr_current_interval =
- g_random_double_range (0.5, 1.5) * stats->min_interval;
+ g_random_double_range (0.5, 1.5) * stats->min_interval * GST_SECOND;
/* This will caused the RTCP to be suppressed if no FB packets are added */
if (sess->last_rtcp_send_time + T_rr_current_interval > new_send_time) {
@@ -3848,7 +3855,7 @@
/* RFC 4585 section 3.5.2 step 2 */
if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) {
GST_LOG_OBJECT (sess, "already have next early rtcp time");
- ret = TRUE;
+ ret = (current_time + max_delay > sess->next_early_rtcp_time);
goto end;
}
@@ -3874,6 +3881,11 @@
GST_TIME_ARGS (sess->next_rtcp_check_time));
ret = TRUE;
} else {
+ GST_LOG_OBJECT (sess,
+ "can't allow early feedback, next scheduled time is too late %"
+ GST_TIME_FORMAT " + %" GST_TIME_FORMAT " < %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (current_time), GST_TIME_ARGS (max_delay),
+ GST_TIME_ARGS (sess->next_rtcp_check_time));
ret = FALSE;
}
goto end;
@@ -3896,8 +3908,12 @@
/* RFC 4585 section 3.5.2 step 3 */
if (current_time + T_dither_max > sess->next_rtcp_check_time) {
- GST_LOG_OBJECT (sess, "don't send because of dither");
- ret = FALSE;
+ GST_LOG_OBJECT (sess,
+ "don't send because of dither, next scheduled time is soon %"
+ GST_TIME_FORMAT " + %" GST_TIME_FORMAT " > %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (current_time), GST_TIME_ARGS (T_dither_max),
+ GST_TIME_ARGS (sess->next_rtcp_check_time));
+ ret = TRUE;
goto end;
}
@@ -3912,7 +3928,11 @@
GST_TIME_ARGS (sess->next_rtcp_check_time));
ret = TRUE;
} else {
- GST_LOG_OBJECT (sess, "can't allow early feedback");
+ GST_LOG_OBJECT (sess,
+ "can't allow early feedback, next scheduled time is too late %"
+ GST_TIME_FORMAT " + %" GST_TIME_FORMAT " < %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (current_time), GST_TIME_ARGS (max_delay),
+ GST_TIME_ARGS (sess->next_rtcp_check_time));
ret = FALSE;
}
goto end;
@@ -3936,8 +3956,10 @@
sess->next_rtcp_check_time = sess->last_rtcp_send_time + 2 * T_rr;
sess->last_rtcp_send_time += T_rr;
- GST_LOG_OBJECT (sess, "next early RTCP time %" GST_TIME_FORMAT,
- GST_TIME_ARGS (sess->next_early_rtcp_time));
+ GST_LOG_OBJECT (sess, "next early RTCP time %" GST_TIME_FORMAT
+ ", next regular RTCP time %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (sess->next_early_rtcp_time),
+ GST_TIME_ARGS (sess->next_rtcp_check_time));
RTP_SESSION_UNLOCK (sess);
/* notify app of need to send packet early
diff --git a/gst/rtpmanager/rtpsource.c b/gst/rtpmanager/rtpsource.c
index 6b0cec5..d97adc4 100644
--- a/gst/rtpmanager/rtpsource.c
+++ b/gst/rtpmanager/rtpsource.c
@@ -50,8 +50,7 @@
PROP_IS_SENDER,
PROP_SDES,
PROP_STATS,
- PROP_PROBATION,
- PROP_LAST
+ PROP_PROBATION
};
/* GObject vmethods */
@@ -748,6 +747,7 @@
GstStructure *s;
guint val;
gint ival;
+ gboolean rtx;
/* nothing changed, return */
if (caps == NULL || src->caps == caps)
@@ -755,11 +755,14 @@
s = gst_caps_get_structure (caps, 0);
- if (gst_structure_get_int (s, "payload", &ival))
+ rtx = (gst_structure_get_uint (s, "rtx-ssrc", &val) && val == src->ssrc);
+
+ if (gst_structure_get_int (s, rtx ? "rtx-payload" : "payload", &ival))
src->payload = ival;
else
src->payload = -1;
- GST_DEBUG ("got payload %d", src->payload);
+
+ GST_DEBUG ("got %spayload %d", rtx ? "rtx " : "", src->payload);
if (gst_structure_get_int (s, "clock-rate", &ival))
src->clock_rate = ival;
@@ -768,12 +771,14 @@
GST_DEBUG ("got clock-rate %d", src->clock_rate);
- if (gst_structure_get_uint (s, "seqnum-offset", &val))
+ if (gst_structure_get_uint (s, rtx ? "rtx-seqnum-offset" : "seqnum-offset",
+ &val))
src->seqnum_offset = val;
else
src->seqnum_offset = -1;
- GST_DEBUG ("got seqnum-offset %" G_GINT32_FORMAT, src->seqnum_offset);
+ GST_DEBUG ("got %sseqnum-offset %" G_GINT32_FORMAT, rtx ? "rtx " : "",
+ src->seqnum_offset);
gst_caps_replace (&src->caps, caps);
}
diff --git a/gst/rtpmanager/rtpstats.c b/gst/rtpmanager/rtpstats.c
index 4da164c..56e58b7 100644
--- a/gst/rtpmanager/rtpstats.c
+++ b/gst/rtpmanager/rtpstats.c
@@ -166,10 +166,10 @@
/* no bandwidth for RTCP, return NONE to signal that we don't want to send
* RTCP packets */
- if (rtcp_bw <= 0.00001)
+ if (rtcp_bw <= 0.0001)
return GST_CLOCK_TIME_NONE;
- avg_rtcp_size = stats->avg_rtcp_packet_size;
+ avg_rtcp_size = 8.0 * stats->avg_rtcp_packet_size;
/*
* The effective number of sites times the average packet size is
* the total number of octets sent when each site sends a report.
@@ -249,7 +249,7 @@
if (rtcp_bw <= 0.0001)
return GST_CLOCK_TIME_NONE;
- avg_rtcp_size = stats->avg_rtcp_packet_size;
+ avg_rtcp_size = 8.0 * stats->avg_rtcp_packet_size;
/*
* The effective number of sites times the average packet size is
* the total number of octets sent when each site sends a report.
diff --git a/gst/rtsp/Makefile.am b/gst/rtsp/Makefile.am
index 59030d7..267e49a 100644
--- a/gst/rtsp/Makefile.am
+++ b/gst/rtsp/Makefile.am
@@ -14,17 +14,3 @@
gstrtsp.h \
gstrtpdec.h \
gstrtspext.h
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstrtsp -:SHARED libgstrtsp \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstrtsp_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstrtsp_la_CFLAGS) \
- -:LDFLAGS $(libgstrtsp_la_LDFLAGS) \
- $(libgstrtsp_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/rtsp/Makefile.in b/gst/rtsp/Makefile.in
index cb211a9..c2958e1 100644
--- a/gst/rtsp/Makefile.in
+++ b/gst/rtsp/Makefile.in
@@ -919,20 +919,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstrtsp -:SHARED libgstrtsp \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstrtsp_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstrtsp_la_CFLAGS) \
- -:LDFLAGS $(libgstrtsp_la_LDFLAGS) \
- $(libgstrtsp_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c
index c786024..397ef31 100644
--- a/gst/rtsp/gstrtspsrc.c
+++ b/gst/rtsp/gstrtspsrc.c
@@ -232,8 +232,7 @@
PROP_SDES,
PROP_TLS_VALIDATION_FLAGS,
PROP_TLS_DATABASE,
- PROP_DO_RETRANSMISSION,
- PROP_LAST
+ PROP_DO_RETRANSMISSION
};
#define GST_TYPE_RTSP_NAT_METHOD (gst_rtsp_nat_method_get_type())
@@ -338,6 +337,31 @@
G_DEFINE_TYPE_WITH_CODE (GstRTSPSrc, gst_rtspsrc, GST_TYPE_BIN,
G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtspsrc_uri_handler_init));
+#ifndef GST_DISABLE_GST_DEBUG
+static inline const char *
+cmd_to_string (guint cmd)
+{
+ switch (cmd) {
+ case CMD_OPEN:
+ return "OPEN";
+ case CMD_PLAY:
+ return "PLAY";
+ case CMD_PAUSE:
+ return "PAUSE";
+ case CMD_CLOSE:
+ return "CLOSE";
+ case CMD_WAIT:
+ return "WAIT";
+ case CMD_RECONNECT:
+ return "RECONNECT";
+ case CMD_LOOP:
+ return "LOOP";
+ }
+
+ return "unknown";
+}
+#endif
+
static gboolean
default_select_stream (GstRTSPSrc * src, guint id, GstCaps * caps)
{
@@ -1636,10 +1660,9 @@
gst_sdp_message_free (src->sdp);
src->sdp = NULL;
}
- if (src->start_segment) {
- gst_event_unref (src->start_segment);
- src->start_segment = NULL;
- }
+
+ src->need_segment = FALSE;
+
if (src->provided_clock) {
gst_object_unref (src->provided_clock);
src->provided_clock = NULL;
@@ -2385,16 +2408,6 @@
}
static gboolean
-gst_rtspsrc_do_seek (GstRTSPSrc * src, GstSegment * segment)
-{
- src->state = GST_RTSP_STATE_SEEKING;
- /* PLAY will add the range header now. */
- src->need_range = TRUE;
-
- return TRUE;
-}
-
-static gboolean
gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
{
gdouble rate;
@@ -2483,7 +2496,10 @@
}
src->skip = skip;
- gst_rtspsrc_do_seek (src, &seeksegment);
+ src->state = GST_RTSP_STATE_SEEKING;
+
+ /* PLAY will add the range header now. */
+ src->need_range = TRUE;
/* and continue playing */
if (playing)
@@ -2691,9 +2707,8 @@
seekable = seekable && src->seekable && src->segment.duration &&
GST_CLOCK_TIME_IS_VALID (src->segment.duration);
- /* FIXME ?? should we have 0 and segment.duration here; see demuxers */
- gst_query_set_seeking (query, GST_FORMAT_TIME, seekable,
- src->segment.start, src->segment.stop);
+ gst_query_set_seeking (query, GST_FORMAT_TIME, seekable, 0,
+ src->segment.duration);
res = TRUE;
}
break;
@@ -3185,9 +3200,13 @@
{
GList *walk;
guint signal_id;
+ gboolean do_retransmission = FALSE;
if (transport->trans != GST_RTSP_TRANS_RTP)
return;
+ if (transport->profile != GST_RTSP_PROFILE_AVPF &&
+ transport->profile != GST_RTSP_PROFILE_SAVPF)
+ return;
signal_id = g_signal_lookup ("request-aux-receiver",
G_OBJECT_TYPE (src->manager));
@@ -3203,6 +3222,7 @@
/* build the retransmission payload type map */
for (walk = src->streams; walk; walk = g_list_next (walk)) {
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
+ gboolean do_retransmission_stream = FALSE;
int i;
if (stream->rtx_pt_map)
@@ -3226,21 +3246,37 @@
if (rtx_pt != 0) {
gst_structure_set (stream->rtx_pt_map, stream_pt_s, G_TYPE_UINT,
rtx_pt, NULL);
+ do_retransmission_stream = TRUE;
}
}
}
}
- GST_DEBUG_OBJECT (src, "built retransmission payload map for stream "
- "id %i: %" GST_PTR_FORMAT, stream->id, stream->rtx_pt_map);
+ if (do_retransmission_stream) {
+ GST_DEBUG_OBJECT (src, "built retransmission payload map for stream "
+ "id %i: %" GST_PTR_FORMAT, stream->id, stream->rtx_pt_map);
+ do_retransmission = TRUE;
+ } else {
+ GST_DEBUG_OBJECT (src, "no retransmission payload map for stream "
+ "id %i", stream->id);
+ gst_structure_free (stream->rtx_pt_map);
+ stream->rtx_pt_map = NULL;
+ }
}
- g_object_set (src->manager, "do-retransmission", TRUE, NULL);
+ if (do_retransmission) {
+ GST_DEBUG_OBJECT (src, "Enabling retransmissions");
- /* enable RFC4588 retransmission handling by setting rtprtxreceive
- * as the "aux" element of rtpbin */
- g_signal_connect (src->manager, "request-aux-receiver",
- (GCallback) request_aux_receiver, src);
+ g_object_set (src->manager, "do-retransmission", TRUE, NULL);
+
+ /* enable RFC4588 retransmission handling by setting rtprtxreceive
+ * as the "aux" element of rtpbin */
+ g_signal_connect (src->manager, "request-aux-receiver",
+ (GCallback) request_aux_receiver, src);
+ } else {
+ GST_DEBUG_OBJECT (src,
+ "Not enabling retransmissions as no stream had a retransmission payload map");
+ }
}
/* try to get and configure a manager */
@@ -4524,7 +4560,6 @@
guint size;
GstBuffer *buf;
gboolean is_rtcp;
- GstEvent *event;
channel = message->type_data.data.channel;
@@ -4581,15 +4616,12 @@
gchar *uri;
GList *streams;
guint group_id = gst_util_group_id_next ();
- GstSegment segment;
/* generate an SHA256 sum of the URI */
cs = g_checksum_new (G_CHECKSUM_SHA256);
uri = src->conninfo.location;
g_checksum_update (cs, (const guchar *) uri, strlen (uri));
- gst_segment_init (&segment, GST_FORMAT_TIME);
-
for (streams = src->streams; streams; streams = g_list_next (streams)) {
GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
GstCaps *caps;
@@ -4630,25 +4662,16 @@
gst_pad_send_event (ostream->channelpad[1],
gst_event_new_caps (caps));
}
+
+ gst_caps_unref (caps);
}
}
-
- /* Push a SEGMENT event if we don't have one pending, if we have one
- * pending we will just send that one a few lines below to all pads
- */
- if (!src->start_segment)
- gst_rtspsrc_stream_push_event (src, ostream,
- gst_event_new_segment (&segment));
}
g_checksum_free (cs);
gst_rtspsrc_activate_streams (src);
src->need_activate = FALSE;
- }
-
- if ((event = src->start_segment) != NULL) {
- src->start_segment = NULL;
- gst_rtspsrc_push_event (src, event);
+ src->need_segment = TRUE;
}
if (src->base_time == -1) {
@@ -4675,6 +4698,15 @@
GST_OBJECT_UNLOCK (src);
}
+ /* If needed send a new segment, don't forget we are live and buffer are
+ * timestamped with running time */
+ if (src->need_segment) {
+ GstSegment segment;
+ src->need_segment = FALSE;
+ gst_segment_init (&segment, GST_FORMAT_TIME);
+ gst_rtspsrc_push_event (src, gst_event_new_segment (&segment));
+ }
+
if (stream->discont && !is_rtcp) {
/* mark first RTP buffer as discont */
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
@@ -5173,7 +5205,7 @@
/* start new request */
gst_rtspsrc_loop_start_cmd (src, cmd);
- GST_DEBUG_OBJECT (src, "sending cmd %d", cmd);
+ GST_DEBUG_OBJECT (src, "sending cmd %s", cmd_to_string (cmd));
GST_OBJECT_LOCK (src);
old = src->pending_cmd;
@@ -5185,18 +5217,20 @@
src->pending_cmd = CMD_WAIT;
GST_OBJECT_UNLOCK (src);
/* cancel previous request */
- GST_DEBUG_OBJECT (src, "cancel previous request %d", old);
+ GST_DEBUG_OBJECT (src, "cancel previous request %s", cmd_to_string (old));
gst_rtspsrc_loop_cancel_cmd (src, old);
GST_OBJECT_LOCK (src);
}
src->pending_cmd = cmd;
/* interrupt if allowed */
if (src->busy_cmd & mask) {
- GST_DEBUG_OBJECT (src, "connection flush busy %d", src->busy_cmd);
+ GST_DEBUG_OBJECT (src, "connection flush busy %s",
+ cmd_to_string (src->busy_cmd));
gst_rtspsrc_connection_flush (src, TRUE);
flushed = TRUE;
} else {
- GST_DEBUG_OBJECT (src, "not interrupting busy cmd %d", src->busy_cmd);
+ GST_DEBUG_OBJECT (src, "not interrupting busy cmd %s",
+ cmd_to_string (src->busy_cmd));
}
if (src->task)
gst_task_start (src->task);
@@ -7516,9 +7550,7 @@
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_RANGE, hval);
/* store the newsegment event so it can be sent from the streaming thread. */
- if (src->start_segment)
- gst_event_unref (src->start_segment);
- src->start_segment = gst_event_new_segment (segment);
+ src->need_segment = TRUE;
}
if (segment->rate != 1.0) {
@@ -7883,7 +7915,7 @@
src->pending_cmd = CMD_LOOP;
else
src->pending_cmd = CMD_WAIT;
- GST_DEBUG_OBJECT (src, "got command %d", cmd);
+ GST_DEBUG_OBJECT (src, "got command %s", cmd_to_string (cmd));
/* we got the message command, so ensure communication is possible again */
gst_rtspsrc_connection_flush (src, FALSE);
diff --git a/gst/rtsp/gstrtspsrc.h b/gst/rtsp/gstrtspsrc.h
index 3d94dbb..730ea97 100644
--- a/gst/rtsp/gstrtspsrc.h
+++ b/gst/rtsp/gstrtspsrc.h
@@ -185,7 +185,7 @@
gboolean need_range;
gboolean skip;
gint free_channel;
- GstEvent *start_segment;
+ gboolean need_segment;
GstClockTime base_time;
/* UDP mode loop */
diff --git a/gst/shapewipe/Makefile.am b/gst/shapewipe/Makefile.am
index 76758ea..6af1662 100644
--- a/gst/shapewipe/Makefile.am
+++ b/gst/shapewipe/Makefile.am
@@ -8,18 +8,3 @@
libgstshapewipe_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
noinst_HEADERS = gstshapewipe.h
-
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstshapewhipe -:SHARED libgstshapewhipe \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstshapewhipe_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstshapewhipe_la_CFLAGS) \
- -:LDFLAGS $(libgstshapewhipe_la_LDFLAGS) \
- $(libgstshapewhipe_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/shapewipe/Makefile.in b/gst/shapewipe/Makefile.in
index ef7b160..1f2f0be 100644
--- a/gst/shapewipe/Makefile.in
+++ b/gst/shapewipe/Makefile.in
@@ -882,20 +882,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstshapewhipe -:SHARED libgstshapewhipe \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstshapewhipe_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstshapewhipe_la_CFLAGS) \
- -:LDFLAGS $(libgstshapewhipe_la_LDFLAGS) \
- $(libgstshapewhipe_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/smpte/Makefile.am b/gst/smpte/Makefile.am
index 22a19fe..b104134 100644
--- a/gst/smpte/Makefile.am
+++ b/gst/smpte/Makefile.am
@@ -9,17 +9,3 @@
-lgstvideo-$(GST_API_VERSION)
libgstsmpte_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstsmpte_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstsmpte -:SHARED libgstsmpte \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstsmpte_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstsmpte_la_CFLAGS) \
- -:LDFLAGS $(libgstsmpte_la_LDFLAGS) \
- $(libgstsmpte_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/smpte/Makefile.in b/gst/smpte/Makefile.in
index d5e66fb..ec22751 100644
--- a/gst/smpte/Makefile.in
+++ b/gst/smpte/Makefile.in
@@ -927,20 +927,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstsmpte -:SHARED libgstsmpte \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstsmpte_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstsmpte_la_CFLAGS) \
- -:LDFLAGS $(libgstsmpte_la_LDFLAGS) \
- $(libgstsmpte_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/smpte/gstsmpte.c b/gst/smpte/gstsmpte.c
index a423b80..722f146 100644
--- a/gst/smpte/gstsmpte.c
+++ b/gst/smpte/gstsmpte.c
@@ -94,8 +94,7 @@
PROP_BORDER,
PROP_DEPTH,
PROP_DURATION,
- PROP_INVERT,
- PROP_LAST,
+ PROP_INVERT
};
/* FIXME: should use video meta etc. */
diff --git a/gst/smpte/gstsmpte.h b/gst/smpte/gstsmpte.h
index 59a4830..3039942 100644
--- a/gst/smpte/gstsmpte.h
+++ b/gst/smpte/gstsmpte.h
@@ -61,10 +61,8 @@
gboolean invert;
/* negotiated format */
- gint format;
gint width;
gint height;
- gdouble fps;
gint fps_num;
gint fps_denom;
GstVideoInfo vinfo1;
diff --git a/gst/smpte/gstsmptealpha.c b/gst/smpte/gstsmptealpha.c
index ba934ac..38e1d6c 100644
--- a/gst/smpte/gstsmptealpha.c
+++ b/gst/smpte/gstsmptealpha.c
@@ -94,8 +94,7 @@
PROP_BORDER,
PROP_DEPTH,
PROP_POSITION,
- PROP_INVERT,
- PROP_LAST,
+ PROP_INVERT
};
#define AYUV_SIZE(w,h) ((w) * (h) * 4)
diff --git a/gst/spectrum/Makefile.am b/gst/spectrum/Makefile.am
index 2f465ca..22633f8 100644
--- a/gst/spectrum/Makefile.am
+++ b/gst/spectrum/Makefile.am
@@ -10,18 +10,3 @@
libgstspectrum_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
noinst_HEADERS = gstspectrum.h
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstspectrum -:SHARED libgstspectrum \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstspectrum_la_SOURCES) \
- $(nodist_libgstspectrum_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstspectrum_la_CFLAGS) \
- -:LDFLAGS $(libgstspectrum_la_LDFLAGS) \
- $(libgstspectrum_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/spectrum/Makefile.in b/gst/spectrum/Makefile.in
index 5089ba5..f799b1f 100644
--- a/gst/spectrum/Makefile.in
+++ b/gst/spectrum/Makefile.in
@@ -888,21 +888,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstspectrum -:SHARED libgstspectrum \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstspectrum_la_SOURCES) \
- $(nodist_libgstspectrum_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstspectrum_la_CFLAGS) \
- -:LDFLAGS $(libgstspectrum_la_LDFLAGS) \
- $(libgstspectrum_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/udp/Makefile.am b/gst/udp/Makefile.am
index 7c109d5..74579cd 100644
--- a/gst/udp/Makefile.am
+++ b/gst/udp/Makefile.am
@@ -12,18 +12,3 @@
EXTRA_DIST = README
CLEANFILES = $(BUILT_SOURCES)
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstudp -:SHARED libgstudp \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstudp_la_SOURCES) \
- $(nodist_libgstudp_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstudp_la_CFLAGS) \
- -:LDFLAGS $(libgstudp_la_LDFLAGS) \
- $(libgstudp_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/udp/Makefile.in b/gst/udp/Makefile.in
index d26f180..9fc150c 100644
--- a/gst/udp/Makefile.in
+++ b/gst/udp/Makefile.in
@@ -929,21 +929,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstudp -:SHARED libgstudp \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstudp_la_SOURCES) \
- $(nodist_libgstudp_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstudp_la_CFLAGS) \
- -:LDFLAGS $(libgstudp_la_LDFLAGS) \
- $(libgstudp_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/udp/gstmultiudpsink.c b/gst/udp/gstmultiudpsink.c
index c81dc62..2f1a8ec 100644
--- a/gst/udp/gstmultiudpsink.c
+++ b/gst/udp/gstmultiudpsink.c
@@ -112,8 +112,7 @@
PROP_SEND_DUPLICATES,
PROP_BUFFER_SIZE,
PROP_BIND_ADDRESS,
- PROP_BIND_PORT,
- PROP_LAST
+ PROP_BIND_PORT
};
static void gst_multiudpsink_finalize (GObject * object);
diff --git a/gst/udp/gstudpsrc.c b/gst/udp/gstudpsrc.c
index 81cb1f1..6e3126f 100644
--- a/gst/udp/gstudpsrc.c
+++ b/gst/udp/gstudpsrc.c
@@ -173,9 +173,7 @@
PROP_USED_SOCKET,
PROP_AUTO_MULTICAST,
PROP_REUSE,
- PROP_ADDRESS,
-
- PROP_LAST
+ PROP_ADDRESS
};
static void gst_udpsrc_uri_handler_init (gpointer g_iface, gpointer iface_data);
diff --git a/gst/videobox/Makefile.am b/gst/videobox/Makefile.am
index 32e1c5d..5a94562 100644
--- a/gst/videobox/Makefile.am
+++ b/gst/videobox/Makefile.am
@@ -19,18 +19,3 @@
noinst_HEADERS = gstvideobox.h
EXTRA_DIST += README
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstvideobox -:SHARED libgstvideobox \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstvideobox_la_SOURCES) \
- $(nodist_libgstvideobox_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstvideobox_la_CFLAGS) \
- -:LDFLAGS $(libgstvideobox_la_LDFLAGS) \
- $(libgstvideobox_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/videobox/Makefile.in b/gst/videobox/Makefile.in
index 8722cac..2042107 100644
--- a/gst/videobox/Makefile.in
+++ b/gst/videobox/Makefile.in
@@ -983,21 +983,6 @@
cp -p tmp-orc.c $(distdir)/$(ORC_SOURCE)-dist.c
cp -p $(ORC_SOURCE).h $(distdir)/$(ORC_SOURCE)-dist.h
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstvideobox -:SHARED libgstvideobox \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstvideobox_la_SOURCES) \
- $(nodist_libgstvideobox_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstvideobox_la_CFLAGS) \
- -:LDFLAGS $(libgstvideobox_la_LDFLAGS) \
- $(libgstvideobox_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/videocrop/Makefile.am b/gst/videocrop/Makefile.am
index 8491444..a9b69cb 100644
--- a/gst/videocrop/Makefile.am
+++ b/gst/videocrop/Makefile.am
@@ -11,18 +11,3 @@
libgstvideocrop_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
noinst_HEADERS = gstvideocrop.h gstaspectratiocrop.h
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstvideocrop -:SHARED libgstvideocrop \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstvideocrop_la_SOURCES) \
- $(nodist_libgstvideocrop_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstvideocrop_la_CFLAGS) \
- -:LDFLAGS $(libgstvideocrop_la_LDFLAGS) \
- $(libgstvideocrop_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/videocrop/Makefile.in b/gst/videocrop/Makefile.in
index 2893eb3..9ae403e 100644
--- a/gst/videocrop/Makefile.in
+++ b/gst/videocrop/Makefile.in
@@ -896,21 +896,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstvideocrop -:SHARED libgstvideocrop \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstvideocrop_la_SOURCES) \
- $(nodist_libgstvideocrop_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstvideocrop_la_CFLAGS) \
- -:LDFLAGS $(libgstvideocrop_la_LDFLAGS) \
- $(libgstvideocrop_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/videocrop/gstaspectratiocrop.c b/gst/videocrop/gstaspectratiocrop.c
index 62e9644..10d0499 100644
--- a/gst/videocrop/gstaspectratiocrop.c
+++ b/gst/videocrop/gstaspectratiocrop.c
@@ -50,8 +50,8 @@
enum
{
- ARG_0,
- ARG_ASPECT_RATIO_CROP,
+ PROP_0,
+ PROP_ASPECT_RATIO_CROP,
};
/* we support the same caps as videocrop (sync changes) */
@@ -183,7 +183,7 @@
gobject_class->get_property = gst_aspect_ratio_crop_get_property;
gobject_class->finalize = gst_aspect_ratio_crop_finalize;
- g_object_class_install_property (gobject_class, ARG_ASPECT_RATIO_CROP,
+ g_object_class_install_property (gobject_class, PROP_ASPECT_RATIO_CROP,
gst_param_spec_fraction ("aspect-ratio", "aspect-ratio",
"Target aspect-ratio of video", 0, 1, G_MAXINT, 1, 0, 1,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
@@ -448,7 +448,7 @@
GST_OBJECT_LOCK (aspect_ratio_crop);
switch (prop_id) {
- case ARG_ASPECT_RATIO_CROP:
+ case PROP_ASPECT_RATIO_CROP:
if (GST_VALUE_HOLDS_FRACTION (value)) {
aspect_ratio_crop->ar_num = gst_value_get_fraction_numerator (value);
aspect_ratio_crop->ar_denom =
@@ -479,7 +479,7 @@
GST_OBJECT_LOCK (aspect_ratio_crop);
switch (prop_id) {
- case ARG_ASPECT_RATIO_CROP:
+ case PROP_ASPECT_RATIO_CROP:
gst_value_set_fraction (value, aspect_ratio_crop->ar_num,
aspect_ratio_crop->ar_denom);
break;
diff --git a/gst/videocrop/gstvideocrop.c b/gst/videocrop/gstvideocrop.c
index 46e4483..99360d5 100644
--- a/gst/videocrop/gstvideocrop.c
+++ b/gst/videocrop/gstvideocrop.c
@@ -70,11 +70,11 @@
enum
{
- ARG_0,
- ARG_LEFT,
- ARG_RIGHT,
- ARG_TOP,
- ARG_BOTTOM
+ PROP_0,
+ PROP_LEFT,
+ PROP_RIGHT,
+ PROP_TOP,
+ PROP_BOTTOM
};
/* we support the same caps as aspectratiocrop (sync changes) */
@@ -177,19 +177,19 @@
gobject_class->set_property = gst_video_crop_set_property;
gobject_class->get_property = gst_video_crop_get_property;
- g_object_class_install_property (gobject_class, ARG_LEFT,
+ g_object_class_install_property (gobject_class, PROP_LEFT,
g_param_spec_int ("left", "Left",
"Pixels to crop at left (-1 to auto-crop)", -1, G_MAXINT, 0,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
- g_object_class_install_property (gobject_class, ARG_RIGHT,
+ g_object_class_install_property (gobject_class, PROP_RIGHT,
g_param_spec_int ("right", "Right",
"Pixels to crop at right (-1 to auto-crop)", -1, G_MAXINT, 0,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
- g_object_class_install_property (gobject_class, ARG_TOP,
+ g_object_class_install_property (gobject_class, PROP_TOP,
g_param_spec_int ("top", "Top",
"Pixels to crop at top (-1 to auto-crop)", -1, G_MAXINT, 0,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
- g_object_class_install_property (gobject_class, ARG_BOTTOM,
+ g_object_class_install_property (gobject_class, PROP_BOTTOM,
g_param_spec_int ("bottom", "Bottom",
"Pixels to crop at bottom (-1 to auto-crop)", -1, G_MAXINT, 0,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
@@ -739,19 +739,19 @@
GST_OBJECT_LOCK (video_crop);
switch (prop_id) {
- case ARG_LEFT:
+ case PROP_LEFT:
gst_video_crop_set_crop (video_crop, g_value_get_int (value),
&video_crop->prop_left);
break;
- case ARG_RIGHT:
+ case PROP_RIGHT:
gst_video_crop_set_crop (video_crop, g_value_get_int (value),
&video_crop->prop_right);
break;
- case ARG_TOP:
+ case PROP_TOP:
gst_video_crop_set_crop (video_crop, g_value_get_int (value),
&video_crop->prop_top);
break;
- case ARG_BOTTOM:
+ case PROP_BOTTOM:
gst_video_crop_set_crop (video_crop, g_value_get_int (value),
&video_crop->prop_bottom);
break;
@@ -759,9 +759,9 @@
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
- GST_LOG_OBJECT (video_crop, "l=%d,r=%d,b=%d,t=%d",
- video_crop->crop_left, video_crop->crop_right, video_crop->crop_bottom,
- video_crop->crop_top);
+ GST_LOG_OBJECT (video_crop, "l=%d,r=%d,b=%d,t=%d, need_update:%d",
+ video_crop->prop_left, video_crop->prop_right, video_crop->prop_bottom,
+ video_crop->prop_top, video_crop->need_update);
GST_OBJECT_UNLOCK (video_crop);
@@ -778,16 +778,16 @@
GST_OBJECT_LOCK (video_crop);
switch (prop_id) {
- case ARG_LEFT:
+ case PROP_LEFT:
g_value_set_int (value, video_crop->prop_left);
break;
- case ARG_RIGHT:
+ case PROP_RIGHT:
g_value_set_int (value, video_crop->prop_right);
break;
- case ARG_TOP:
+ case PROP_TOP:
g_value_set_int (value, video_crop->prop_top);
break;
- case ARG_BOTTOM:
+ case PROP_BOTTOM:
g_value_set_int (value, video_crop->prop_bottom);
break;
default:
diff --git a/gst/videofilter/Makefile.am b/gst/videofilter/Makefile.am
index 71755e6..a0cae46 100644
--- a/gst/videofilter/Makefile.am
+++ b/gst/videofilter/Makefile.am
@@ -21,18 +21,3 @@
gstvideoexample.c: $(srcdir)/make_filter $(srcdir)/gstvideotemplate.c
$(srcdir)/make_filter Videoexample $(srcdir)/gstvideotemplate.c
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstvideofilter -:SHARED libgstvideofilter \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstvideofilter_la_SOURCES) \
- $(nodist_libgstvideofilter_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstvideofilter_la_CFLAGS) \
- -:LDFLAGS $(libgstvideofilter_la_LDFLAGS) \
- $(libgstvideofilter_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/videofilter/Makefile.in b/gst/videofilter/Makefile.in
index 550adc0..6dd29ea 100644
--- a/gst/videofilter/Makefile.in
+++ b/gst/videofilter/Makefile.in
@@ -935,21 +935,6 @@
gstvideoexample.c: $(srcdir)/make_filter $(srcdir)/gstvideotemplate.c
$(srcdir)/make_filter Videoexample $(srcdir)/gstvideotemplate.c
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstvideofilter -:SHARED libgstvideofilter \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstvideofilter_la_SOURCES) \
- $(nodist_libgstvideofilter_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstvideofilter_la_CFLAGS) \
- -:LDFLAGS $(libgstvideofilter_la_LDFLAGS) \
- $(libgstvideofilter_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/videofilter/gstvideotemplate.c b/gst/videofilter/gstvideotemplate.c
index 3d60f55..660526a 100644
--- a/gst/videofilter/gstvideotemplate.c
+++ b/gst/videofilter/gstvideotemplate.c
@@ -68,7 +68,7 @@
enum
{
- ARG_0
+ PROP_0
/* FILL ME */
};
@@ -145,7 +145,7 @@
videofilter_class = GST_VIDEOFILTER_CLASS (g_class);
#if 0
- g_object_class_install_property (gobject_class, ARG_METHOD,
+ g_object_class_install_property (gobject_class, PROP_METHOD,
g_param_spec_enum ("method", "method", "method",
GST_TYPE_VIDEOTEMPLATE_METHOD, GST_VIDEOTEMPLATE_METHOD_1,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
@@ -182,7 +182,7 @@
GST_DEBUG ("gst_videotemplate_set_property");
switch (prop_id) {
#if 0
- case ARG_METHOD:
+ case PROP_METHOD:
src->method = g_value_get_enum (value);
break;
#endif
@@ -202,7 +202,7 @@
switch (prop_id) {
#if 0
- case ARG_METHOD:
+ case PROP_METHOD:
g_value_set_enum (value, src->method);
break;
#endif
diff --git a/gst/videomixer/Makefile.am b/gst/videomixer/Makefile.am
index f8499b6..4250342 100644
--- a/gst/videomixer/Makefile.am
+++ b/gst/videomixer/Makefile.am
@@ -22,18 +22,3 @@
blend.h \
videomixer2.h \
videomixer2pad.h
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstvideomixer -:SHARED libgstvideomixer \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstvideomixer_la_SOURCES) \
- $(nodist_libgstvideomixer_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstvideomixer_la_CFLAGS) \
- -:LDFLAGS $(libgstvideomixer_la_LDFLAGS) \
- $(libgstvideomixer_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/videomixer/Makefile.in b/gst/videomixer/Makefile.in
index 3a0af1c..955512f 100644
--- a/gst/videomixer/Makefile.in
+++ b/gst/videomixer/Makefile.in
@@ -997,21 +997,6 @@
cp -p tmp-orc.c $(distdir)/$(ORC_SOURCE)-dist.c
cp -p $(ORC_SOURCE).h $(distdir)/$(ORC_SOURCE)-dist.h
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstvideomixer -:SHARED libgstvideomixer \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstvideomixer_la_SOURCES) \
- $(nodist_libgstvideomixer_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstvideomixer_la_CFLAGS) \
- -:LDFLAGS $(libgstvideomixer_la_LDFLAGS) \
- $(libgstvideomixer_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/videomixer/videomixerorc-dist.c b/gst/videomixer/videomixerorc-dist.c
index 7ddc1b0..115b6fc 100644
--- a/gst/videomixer/videomixerorc-dist.c
+++ b/gst/videomixer/videomixerorc-dist.c
@@ -1713,7 +1713,7 @@
static const orc_uint8 bc[] = {
1, 7, 9, 28, 118, 105, 100, 101, 111, 95, 109, 105, 120, 101, 114, 95,
111, 114, 99, 95, 111, 118, 101, 114, 108, 97, 121, 95, 97, 114, 103,
- 98,
+ 98,
11, 4, 4, 12, 4, 4, 14, 4, 255, 255, 255, 255, 14, 4, 255, 0,
0, 0, 14, 4, 0, 255, 255, 255, 14, 2, 8, 0, 0, 0, 16, 2,
20, 4, 20, 2, 20, 1, 20, 8, 20, 8, 20, 8, 20, 4, 20, 8,
@@ -2283,7 +2283,7 @@
static const orc_uint8 bc[] = {
1, 7, 9, 28, 118, 105, 100, 101, 111, 95, 109, 105, 120, 101, 114, 95,
111, 114, 99, 95, 111, 118, 101, 114, 108, 97, 121, 95, 98, 103, 114,
- 97,
+ 97,
11, 4, 4, 12, 4, 4, 14, 4, 255, 255, 255, 255, 14, 4, 0, 0,
0, 255, 14, 4, 255, 255, 255, 0, 14, 4, 24, 0, 0, 0, 14, 2,
8, 0, 0, 0, 16, 2, 20, 4, 20, 4, 20, 2, 20, 1, 20, 8,
diff --git a/gst/wavenc/Makefile.am b/gst/wavenc/Makefile.am
index 98d6f66..a25dd31 100644
--- a/gst/wavenc/Makefile.am
+++ b/gst/wavenc/Makefile.am
@@ -15,17 +15,3 @@
libgstwavenc_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
noinst_HEADERS = gstwavenc.h
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstwavenc -:SHARED libgstwavenc \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstwavenc_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstwavenc_la_CFLAGS) \
- -:LDFLAGS $(libgstwavenc_la_LDFLAGS) \
- $(libgstwavenc_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/wavenc/Makefile.in b/gst/wavenc/Makefile.in
index cb2f961..bffe23a 100644
--- a/gst/wavenc/Makefile.in
+++ b/gst/wavenc/Makefile.in
@@ -892,20 +892,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstwavenc -:SHARED libgstwavenc \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstwavenc_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstwavenc_la_CFLAGS) \
- -:LDFLAGS $(libgstwavenc_la_LDFLAGS) \
- $(libgstwavenc_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/wavparse/Makefile.am b/gst/wavparse/Makefile.am
index 2a0aa40..d3f1818 100644
--- a/gst/wavparse/Makefile.am
+++ b/gst/wavparse/Makefile.am
@@ -17,17 +17,3 @@
libgstwavparse_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
noinst_HEADERS = gstwavparse.h
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstwavparse -:SHARED libgstwavparse \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstwavparse_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstwavparse_la_CFLAGS) \
- -:LDFLAGS $(libgstwavparse_la_LDFLAGS) \
- $(libgstwavparse_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/wavparse/Makefile.in b/gst/wavparse/Makefile.in
index b54ff0f..9c7ce5d 100644
--- a/gst/wavparse/Makefile.in
+++ b/gst/wavparse/Makefile.in
@@ -895,20 +895,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgstwavparse -:SHARED libgstwavparse \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgstwavparse_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstwavparse_la_CFLAGS) \
- -:LDFLAGS $(libgstwavparse_la_LDFLAGS) \
- $(libgstwavparse_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/wavparse/gstwavparse.c b/gst/wavparse/gstwavparse.c
index 41d5151..b0e3423 100644
--- a/gst/wavparse/gstwavparse.c
+++ b/gst/wavparse/gstwavparse.c
@@ -1261,8 +1261,20 @@
}
GST_INFO_OBJECT (wav,
- "Got TAG: %" GST_FOURCC_FORMAT ", offset %" G_GUINT64_FORMAT,
- GST_FOURCC_ARGS (tag), wav->offset);
+ "Got TAG: %" GST_FOURCC_FORMAT ", offset %" G_GUINT64_FORMAT ", size %"
+ G_GUINT32_FORMAT, GST_FOURCC_ARGS (tag), wav->offset, size);
+
+ /* Maximum valid size is INT_MAX */
+ if (size & 0x80000000) {
+ GST_WARNING_OBJECT (wav, "Invalid size, clipping to 0x7fffffff");
+ size = 0x7fffffff;
+ }
+
+ /* Clip to upstream size if known */
+ if (wav->datasize > 0 && size + wav->offset > wav->datasize) {
+ GST_WARNING_OBJECT (wav, "Clipping chunk size to file size");
+ size = wav->datasize - wav->offset;
+ }
/* wav is a st00pid format, we don't know for sure where data starts.
* So we have to go bit by bit until we find the 'data' header
@@ -1827,12 +1839,16 @@
s = gst_caps_get_structure (caps, 0);
if (!gst_structure_has_name (s, "audio/x-dts"))
return FALSE;
- if (prob >= GST_TYPE_FIND_LIKELY)
+ /* typefind behavior for DTS:
+ * MAXIMUM: multiple frame syncs detected, certainly DTS
+ * LIKELY: single frame sync at offset 0. Maybe DTS?
+ * POSSIBLE: single frame sync, not at offset 0. Highly unlikely
+ * to be DTS. */
+ if (prob > GST_TYPE_FIND_LIKELY)
return TRUE;
- /* DTS at non-0 offsets and without second sync may yield POSSIBLE .. */
- if (prob < GST_TYPE_FIND_POSSIBLE)
+ if (prob <= GST_TYPE_FIND_POSSIBLE)
return FALSE;
- /* .. in which case we want at least a valid-looking rate and channels */
+ /* for maybe, check for at least a valid-looking rate and channels */
if (!gst_structure_has_field (s, "channels"))
return FALSE;
/* and for extra assurance we could also check the rate from the DTS frame
diff --git a/gst/y4m/Makefile.am b/gst/y4m/Makefile.am
index ae424f9..844dadf 100644
--- a/gst/y4m/Makefile.am
+++ b/gst/y4m/Makefile.am
@@ -8,17 +8,3 @@
libgsty4menc_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
noinst_HEADERS = gsty4mencode.h
-
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgsty4menc -:SHARED libgsty4menc \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgsty4menc_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgsty4menc_la_CFLAGS) \
- -:LDFLAGS $(libgsty4menc_la_LDFLAGS) \
- $(libgsty4menc_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
diff --git a/gst/y4m/Makefile.in b/gst/y4m/Makefile.in
index 608bb2a..956ebd7 100644
--- a/gst/y4m/Makefile.in
+++ b/gst/y4m/Makefile.in
@@ -882,20 +882,6 @@
uninstall-pluginLTLIBRARIES
-Android.mk: Makefile.am $(BUILT_SOURCES)
- androgenizer \
- -:PROJECT libgsty4menc -:SHARED libgsty4menc \
- -:TAGS eng debug \
- -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
- -:SOURCES $(libgsty4menc_la_SOURCES) \
- -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgsty4menc_la_CFLAGS) \
- -:LDFLAGS $(libgsty4menc_la_LDFLAGS) \
- $(libgsty4menc_la_LIBADD) \
- -ldl \
- -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
- LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
- > $@
-
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
diff --git a/gst/y4m/gsty4mencode.c b/gst/y4m/gsty4mencode.c
index c4cbde6..7b39d5f 100644
--- a/gst/y4m/gsty4mencode.c
+++ b/gst/y4m/gsty4mencode.c
@@ -56,7 +56,7 @@
enum
{
- ARG_0
+ PROP_0
};
static GstStaticPadTemplate y4mencode_src_factory =
diff --git a/po/POTFILES.in b/po/POTFILES.in
index ccbef6f..fd21f95 100644
--- a/po/POTFILES.in
+++ b/po/POTFILES.in
@@ -10,6 +10,7 @@
gst/avi/gstavidemux.c
gst/avi/gstavimux.c
gst/isomp4/qtdemux.c
+gst/multifile/gstsplitmuxsrc.c
gst/rtsp/gstrtspsrc.c
gst/wavparse/gstwavparse.c
sys/oss4/oss4-sink.c
diff --git a/po/af.gmo b/po/af.gmo
index ea91b65..e28aa21 100644
--- a/po/af.gmo
+++ b/po/af.gmo
Binary files differ
diff --git a/po/af.po b/po/af.po
index afe7544..baa38ec 100644
--- a/po/af.po
+++ b/po/af.po
@@ -7,7 +7,7 @@
msgstr ""
"Project-Id-Version: gst-plugins 0.7.6\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2004-03-18 14:16+0200\n"
"Last-Translator: Petri Jooste <rkwjpj@puk.ac.za>\n"
"Language-Team: Afrikaans <i18n@af.org.za>\n"
@@ -86,6 +86,9 @@
msgid "This file contains too many streams. Only playing first %d"
msgstr ""
+msgid "Internal data flow error."
+msgstr ""
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -97,9 +100,6 @@
"plugin."
msgstr ""
-msgid "Internal data flow error."
-msgstr ""
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/az.gmo b/po/az.gmo
index 2deec12..bdd79be 100644
--- a/po/az.gmo
+++ b/po/az.gmo
Binary files differ
diff --git a/po/az.po b/po/az.po
index fd91884..77c0009 100644
--- a/po/az.po
+++ b/po/az.po
@@ -7,7 +7,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-0.8.0\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2004-03-19 18:29+0200\n"
"Last-Translator: Metin Amiroff <metin@karegen.com>\n"
"Language-Team: Azerbaijani <translation-team-az@lists.sourceforge.net>\n"
@@ -87,6 +87,9 @@
msgid "This file contains too many streams. Only playing first %d"
msgstr ""
+msgid "Internal data flow error."
+msgstr ""
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -98,9 +101,6 @@
"plugin."
msgstr ""
-msgid "Internal data flow error."
-msgstr ""
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/bg.gmo b/po/bg.gmo
index eddfd9e..e634ee2 100644
--- a/po/bg.gmo
+++ b/po/bg.gmo
Binary files differ
diff --git a/po/bg.po b/po/bg.po
index c9feb21..163f700 100644
--- a/po/bg.po
+++ b/po/bg.po
@@ -8,7 +8,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good 0.10.28.2\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2011-04-26 22:35+0300\n"
"Last-Translator: Alexander Shopov <ash@kambanaria.org>\n"
"Language-Team: Bulgarian <dict@fsa-bg.org>\n"
@@ -86,6 +86,9 @@
msgstr ""
"Този файл съдържа прекалено много потоци. Изпълняват се само първите %d"
+msgid "Internal data flow error."
+msgstr "Вътрешна грешка в потока с данни."
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -101,9 +104,6 @@
"Не е открит поддържан поток. Може би трябва да разрешите повече транспортни "
"протоколи или ви липсва правилната приставка на GStreamer за RTSP."
-msgid "Internal data flow error."
-msgstr "Вътрешна грешка в потока с данни."
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/ca.gmo b/po/ca.gmo
index d6cbcd9..8c45936 100644
--- a/po/ca.gmo
+++ b/po/ca.gmo
Binary files differ
diff --git a/po/ca.po b/po/ca.po
index 0db48c0..8a7b005 100644
--- a/po/ca.po
+++ b/po/ca.po
@@ -9,7 +9,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good 0.10.28.2\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2012-01-01 14:19+0100\n"
"Last-Translator: Gil Forcada <gforcada@gnome.org>\n"
"Language-Team: Catalan <ca@dodds.net>\n"
@@ -89,6 +89,9 @@
msgstr ""
"Aquest fitxer conté massa fluxos. Només s'estan reproduint els primers %d"
+msgid "Internal data flow error."
+msgstr "S'ha produït un error intern de flux de dades."
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -105,9 +108,6 @@
"protocols de transport o bé potser manca el connector d'extensió RTSP del "
"GStreamer correcte."
-msgid "Internal data flow error."
-msgstr "S'ha produït un error intern de flux de dades."
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/cs.gmo b/po/cs.gmo
index 1bfad59..0d55f2a 100644
--- a/po/cs.gmo
+++ b/po/cs.gmo
Binary files differ
diff --git a/po/cs.po b/po/cs.po
index 8ed6164..2fe592b 100644
--- a/po/cs.po
+++ b/po/cs.po
@@ -10,7 +10,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good-1.1.4\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2013-09-18 19:18+0200\n"
"Last-Translator: Marek Černocký <marek@manet.cz>\n"
"Language-Team: Czech <translation-team-cs@lists.sourceforge.net>\n"
@@ -88,6 +88,9 @@
msgid "This file contains too many streams. Only playing first %d"
msgstr "Tento soubor obsahuje příliš mnoho proudů. Přehrává se pouze první %d"
+msgid "Internal data flow error."
+msgstr "Vnitřní chyba datového toku."
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -104,9 +107,6 @@
"přenosových protokolů, jinak nejspíš bude scházet potřebný zásuvný modul "
"rozšíření GStreamer RTSP."
-msgid "Internal data flow error."
-msgstr "Vnitřní chyba datového toku."
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/da.gmo b/po/da.gmo
index e749006..54aae8c 100644
--- a/po/da.gmo
+++ b/po/da.gmo
Binary files differ
diff --git a/po/da.po b/po/da.po
index c2a4657..d421e02 100644
--- a/po/da.po
+++ b/po/da.po
@@ -11,7 +11,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good-1.3.90\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2014-06-29 23:54+0200\n"
"Last-Translator: Joe Hansen <joedalton2@yahoo.dk>\n"
"Language-Team: Danish <dansk@dansk-gruppen.dk>\n"
@@ -91,6 +91,9 @@
msgid "This file contains too many streams. Only playing first %d"
msgstr "Denne fil indeholder for mange strømme. Afspil først kun %d"
+msgid "Internal data flow error."
+msgstr "Intern datastrømsfejl."
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -107,9 +110,6 @@
"transportprotokoller eller mangler måske det korrekte udvidelsesmodul til "
"GStreamer RTSP."
-msgid "Internal data flow error."
-msgstr "Intern datastrømsfejl."
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/de.gmo b/po/de.gmo
index 33a2d18..a83cb8d 100644
--- a/po/de.gmo
+++ b/po/de.gmo
Binary files differ
diff --git a/po/de.po b/po/de.po
index 7cfb100..4516c43 100644
--- a/po/de.po
+++ b/po/de.po
@@ -14,7 +14,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good 1.3.90\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2014-06-29 21:20+0100\n"
"Last-Translator: Christian Kirbach <christian.kirbach@gmail.com>\n"
"Language-Team: German <translation-team-de@lists.sourceforge.net>\n"
@@ -98,6 +98,9 @@
"Diese Datei enthält zu viele Ströme. Es werden nur die ersten %d "
"wiedergegeben"
+msgid "Internal data flow error."
+msgstr "Interner Datenstromfehler."
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -114,9 +117,6 @@
"Transport-Protokolle erlauben oder es fehlt vielleicht das richtige Plugin "
"zur RTSP-Erweiterung."
-msgid "Internal data flow error."
-msgstr "Interner Datenstromfehler."
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/el.gmo b/po/el.gmo
index eb7c98b..9dfbe1a 100644
--- a/po/el.gmo
+++ b/po/el.gmo
Binary files differ
diff --git a/po/el.po b/po/el.po
index 76c0937..1957ae7 100644
--- a/po/el.po
+++ b/po/el.po
@@ -8,7 +8,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good 0.10.28.2\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2012-05-05 19:05+0100\n"
"Last-Translator: Savvas Radevic <vicedar@gmail.com>\n"
"Language-Team: Greek <team@lists.gnome.gr>\n"
@@ -87,6 +87,9 @@
msgid "This file contains too many streams. Only playing first %d"
msgstr "Αυτό το αρχείο περιέχει πολλές ροές. Αναπαράγεται μόνο η πρώτη %d"
+msgid "Internal data flow error."
+msgstr "Εσωτερικό σφάλμα ροής δεδομένων."
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -103,9 +106,6 @@
"επιτρέψετε περισσότερα πρωτόκολλα μεταγωγής ή αλλιώς να σας λείπει το σωστό "
"πρόσθετο RTSP του GStreamer."
-msgid "Internal data flow error."
-msgstr "Εσωτερικό σφάλμα ροής δεδομένων."
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/en_GB.gmo b/po/en_GB.gmo
index 87d70e2..f4bf50a 100644
--- a/po/en_GB.gmo
+++ b/po/en_GB.gmo
Binary files differ
diff --git a/po/en_GB.po b/po/en_GB.po
index 683a69b..1634c83 100644
--- a/po/en_GB.po
+++ b/po/en_GB.po
@@ -6,7 +6,7 @@
msgstr ""
"Project-Id-Version: gst-plugins 0.8.1\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2004-04-26 10:41-0400\n"
"Last-Translator: Gareth Owen <gowen72@yahoo.com>\n"
"Language-Team: English (British) <en_gb@li.org>\n"
@@ -86,6 +86,9 @@
msgid "This file contains too many streams. Only playing first %d"
msgstr ""
+msgid "Internal data flow error."
+msgstr ""
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -97,9 +100,6 @@
"plugin."
msgstr ""
-msgid "Internal data flow error."
-msgstr ""
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/eo.gmo b/po/eo.gmo
index 0f1a262..dc2ff77 100644
--- a/po/eo.gmo
+++ b/po/eo.gmo
Binary files differ
diff --git a/po/eo.po b/po/eo.po
index fc3cd03..2247ca9 100644
--- a/po/eo.po
+++ b/po/eo.po
@@ -7,7 +7,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good 0.10.28.2\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2011-06-04 21:48+0100\n"
"Last-Translator: Kristjan SCHMIDT <kristjan.schmidt@googlemail.com>\n"
"Language-Team: Esperanto <translation-team-eo@lists.sourceforge.net>\n"
@@ -84,6 +84,9 @@
msgid "This file contains too many streams. Only playing first %d"
msgstr ""
+msgid "Internal data flow error."
+msgstr ""
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -95,9 +98,6 @@
"plugin."
msgstr ""
-msgid "Internal data flow error."
-msgstr ""
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/es.gmo b/po/es.gmo
index d9efa50..e02f4c8 100644
--- a/po/es.gmo
+++ b/po/es.gmo
Binary files differ
diff --git a/po/es.po b/po/es.po
index 3aef96a..3a0887c 100644
--- a/po/es.po
+++ b/po/es.po
@@ -7,7 +7,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good 0.10.26.2\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2011-02-12 18:30+0100\n"
"Last-Translator: Jorge González González <aloriel@gmail.com>\n"
"Language-Team: Spanish <es@li.org>\n"
@@ -88,6 +88,9 @@
msgstr ""
"Este archivo contiene demasiados flujos. Sólo se reproducirá el primer %d"
+msgid "Internal data flow error."
+msgstr "Error en el flujo de datos interno."
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -104,9 +107,6 @@
"protocolos de transporte o de otra forma puede que le falte la extensión "
"correcta de RTSP de GStreamer."
-msgid "Internal data flow error."
-msgstr "Error en el flujo de datos interno."
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/eu.gmo b/po/eu.gmo
index 6de8c1c..df8ab28 100644
--- a/po/eu.gmo
+++ b/po/eu.gmo
Binary files differ
diff --git a/po/eu.po b/po/eu.po
index 354e437..9645966 100644
--- a/po/eu.po
+++ b/po/eu.po
@@ -8,7 +8,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good-0.10.18.2\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2010-03-25 12:37+0100\n"
"Last-Translator: Mikel Olasagasti Uranga <hey_neken@mundurat.net>\n"
"Language-Team: Basque <translation-team-eu@lists.sourceforge.net>\n"
@@ -92,6 +92,9 @@
msgstr ""
"Fitxategi horrek korronte gehiegi ditu. Erreproduzitu soilik lehen %d(r)ak"
+msgid "Internal data flow error."
+msgstr "Datu-fluxuaren barne-errorea."
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -107,9 +110,6 @@
"Ez da onartutako korronterik aurkitu. Garraioko protokolo gehiago baimentzea "
"behar da edo GStreamer RTSP hedapen egokia falta zaizu."
-msgid "Internal data flow error."
-msgstr "Datu-fluxuaren barne-errorea."
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/fi.gmo b/po/fi.gmo
index a0642ca..8e935b0 100644
--- a/po/fi.gmo
+++ b/po/fi.gmo
Binary files differ
diff --git a/po/fi.po b/po/fi.po
index 55b7f55..0075314 100644
--- a/po/fi.po
+++ b/po/fi.po
@@ -11,7 +11,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good 0.10.25.3\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2010-11-17 23:03+0200\n"
"Last-Translator: Tommi Vainikainen <Tommi.Vainikainen@iki.fi>\n"
"Language-Team: Finnish <translation-team-fi@lists.sourceforge.net>\n"
@@ -90,6 +90,9 @@
msgid "This file contains too many streams. Only playing first %d"
msgstr "Tiedosto sisältää useita virtoja. Soitetaan vain ensimmäiset %d"
+msgid "Internal data flow error."
+msgstr "Sisäisen tietovirran virhe."
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -106,9 +109,6 @@
"siirtoprotokollia, tai vaihtoehtoisesti sinulta saattaa puuttua oikea "
"GStreamer-RTSP-laajennusliitännäinen."
-msgid "Internal data flow error."
-msgstr "Sisäisen tietovirran virhe."
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/fr.gmo b/po/fr.gmo
index 5d6a39f..fdd1345 100644
--- a/po/fr.gmo
+++ b/po/fr.gmo
Binary files differ
diff --git a/po/fr.po b/po/fr.po
index 70342fd..cf5554f 100644
--- a/po/fr.po
+++ b/po/fr.po
@@ -9,7 +9,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good 0.10.28.2\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2011-04-28 09:22+0200\n"
"Last-Translator: Claude Paroz <claude@2xlibre.net>\n"
"Language-Team: French <traduc@traduc.org>\n"
@@ -89,6 +89,9 @@
msgid "This file contains too many streams. Only playing first %d"
msgstr "Ce fichier contient trop de flux. Seuls les %d premiers seront lus."
+msgid "Internal data flow error."
+msgstr "Erreur interne de flux de données."
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -105,9 +108,6 @@
"davantage de protocoles de transport ou il manque peut-être le bon greffon "
"d'extension GStreamer RTSP."
-msgid "Internal data flow error."
-msgstr "Erreur interne de flux de données."
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/gl.gmo b/po/gl.gmo
index 00208f3..933806d 100644
--- a/po/gl.gmo
+++ b/po/gl.gmo
Binary files differ
diff --git a/po/gl.po b/po/gl.po
index 2a01485..9b9ce58 100644
--- a/po/gl.po
+++ b/po/gl.po
@@ -7,7 +7,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good 1.0.3\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2012-12-15 03:46+0200\n"
"Last-Translator: Fran Dieguez <frandieguez@ubuntu.com>\n"
"Language-Team: Galician <proxecto@trasno.net>\n"
@@ -89,6 +89,9 @@
msgstr ""
"Este ficheiro contén demasiados fluxos. Só se reproducirá o primeiro %d"
+msgid "Internal data flow error."
+msgstr "Produciuse un erro interno no fluxo de datos."
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -105,9 +108,6 @@
"protocolos de transporte ou de outra forma pode que lle falte a extensión "
"correcta de RTSP de GStreamer."
-msgid "Internal data flow error."
-msgstr "Produciuse un erro interno no fluxo de datos."
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/gst-plugins-good-1.0.pot b/po/gst-plugins-good-1.0.pot
index b5e9e02..033c31a 100644
--- a/po/gst-plugins-good-1.0.pot
+++ b/po/gst-plugins-good-1.0.pot
@@ -7,7 +7,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good 1.5.0.1\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: YEAR-MO-DA HO:MI+ZONE\n"
"Last-Translator: FULL NAME <EMAIL@ADDRESS>\n"
"Language-Team: LANGUAGE <LL@li.org>\n"
@@ -20,10 +20,10 @@
msgid "Jack server not found"
msgstr ""
-#: ext/jpeg/gstjpegdec.c:921 ext/jpeg/gstjpegdec.c:1169
-#: ext/jpeg/gstjpegdec.c:1181 ext/jpeg/gstjpegdec.c:1209
-#: ext/jpeg/gstjpegdec.c:1218 ext/jpeg/gstjpegdec.c:1227
-#: ext/jpeg/gstjpegdec.c:1235
+#: ext/jpeg/gstjpegdec.c:921 ext/jpeg/gstjpegdec.c:1174
+#: ext/jpeg/gstjpegdec.c:1186 ext/jpeg/gstjpegdec.c:1215
+#: ext/jpeg/gstjpegdec.c:1224 ext/jpeg/gstjpegdec.c:1233
+#: ext/jpeg/gstjpegdec.c:1241
msgid "Failed to decode JPEG image"
msgstr ""
@@ -78,56 +78,56 @@
msgid "This file contains no playable streams."
msgstr ""
-#: gst/isomp4/qtdemux.c:649 gst/isomp4/qtdemux.c:5039
-#: gst/isomp4/qtdemux.c:5106 gst/isomp4/qtdemux.c:5345
+#: gst/isomp4/qtdemux.c:649 gst/isomp4/qtdemux.c:5060
+#: gst/isomp4/qtdemux.c:5127 gst/isomp4/qtdemux.c:5365
msgid "This file is invalid and cannot be played."
msgstr ""
-#: gst/isomp4/qtdemux.c:2324
+#: gst/isomp4/qtdemux.c:2329
msgid "Cannot play stream because it is encrypted with PlayReady DRM."
msgstr ""
-#: gst/isomp4/qtdemux.c:3028 gst/isomp4/qtdemux.c:6044
-#: gst/isomp4/qtdemux.c:6051 gst/isomp4/qtdemux.c:6705
-#: gst/isomp4/qtdemux.c:7132 gst/isomp4/qtdemux.c:7139
-#: gst/isomp4/qtdemux.c:9256
+#: gst/isomp4/qtdemux.c:3033 gst/isomp4/qtdemux.c:6072
+#: gst/isomp4/qtdemux.c:6079 gst/isomp4/qtdemux.c:6736
+#: gst/isomp4/qtdemux.c:7163 gst/isomp4/qtdemux.c:7170
+#: gst/isomp4/qtdemux.c:9287
msgid "This file is corrupt and cannot be played."
msgstr ""
-#: gst/isomp4/qtdemux.c:3261
+#: gst/isomp4/qtdemux.c:3275
msgid "Invalid atom size."
msgstr ""
-#: gst/isomp4/qtdemux.c:3337
+#: gst/isomp4/qtdemux.c:3351
msgid "This file is incomplete and cannot be played."
msgstr ""
-#: gst/isomp4/qtdemux.c:7908
+#: gst/isomp4/qtdemux.c:7939
msgid "The video in this file might not play correctly."
msgstr ""
-#: gst/isomp4/qtdemux.c:9297
+#: gst/isomp4/qtdemux.c:9328
#, c-format
msgid "This file contains too many streams. Only playing first %d"
msgstr ""
-#: gst/rtsp/gstrtspsrc.c:6665
+#: gst/multifile/gstsplitmuxsrc.c:548 gst/wavparse/gstwavparse.c:2248
+msgid "Internal data flow error."
+msgstr ""
+
+#: gst/rtsp/gstrtspsrc.c:6699
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
msgstr ""
-#: gst/rtsp/gstrtspsrc.c:6670
+#: gst/rtsp/gstrtspsrc.c:6704
msgid ""
"No supported stream was found. You might need to allow more transport "
"protocols or may otherwise be missing the right GStreamer RTSP extension "
"plugin."
msgstr ""
-#: gst/wavparse/gstwavparse.c:2232
-msgid "Internal data flow error."
-msgstr ""
-
#: sys/oss4/oss4-sink.c:493 sys/oss4/oss4-source.c:358
#: sys/oss/gstosssink.c:384
msgid ""
@@ -251,85 +251,85 @@
msgid "AUX 2 Out"
msgstr ""
-#: sys/v4l2/gstv4l2bufferpool.c:1648
+#: sys/v4l2/gstv4l2bufferpool.c:1657
#, c-format
msgid "Error reading %d bytes from device '%s'."
msgstr ""
-#: sys/v4l2/gstv4l2object.c:1064
+#: sys/v4l2/gstv4l2object.c:1086
#, c-format
msgid "Failed to enumerate possible video formats device '%s' can work with"
msgstr ""
-#: sys/v4l2/gstv4l2object.c:2424
+#: sys/v4l2/gstv4l2object.c:2453
#, c-format
msgid "Could not map buffers from device '%s'"
msgstr ""
-#: sys/v4l2/gstv4l2object.c:2432
+#: sys/v4l2/gstv4l2object.c:2461
#, c-format
msgid "The driver of device '%s' does not support the IO method %d"
msgstr ""
-#: sys/v4l2/gstv4l2object.c:2439
+#: sys/v4l2/gstv4l2object.c:2468
#, c-format
msgid "The driver of device '%s' does not support any known IO method."
msgstr ""
-#: sys/v4l2/gstv4l2object.c:2945 sys/v4l2/gstv4l2object.c:2956
+#: sys/v4l2/gstv4l2object.c:2974 sys/v4l2/gstv4l2object.c:2985
#, c-format
msgid "Device '%s' is busy"
msgstr ""
-#: sys/v4l2/gstv4l2object.c:2962 sys/v4l2/gstv4l2object.c:2974
+#: sys/v4l2/gstv4l2object.c:2991 sys/v4l2/gstv4l2object.c:3003
#, c-format
msgid "Device '%s' cannot capture at %dx%d"
msgstr ""
-#: sys/v4l2/gstv4l2object.c:2985
+#: sys/v4l2/gstv4l2object.c:3014
#, c-format
msgid "Device '%s' cannot capture in the specified format"
msgstr ""
-#: sys/v4l2/gstv4l2object.c:2998
+#: sys/v4l2/gstv4l2object.c:3027
#, c-format
msgid "Device '%s' does support non-contiguous planes"
msgstr ""
-#: sys/v4l2/gstv4l2object.c:3009
+#: sys/v4l2/gstv4l2object.c:3038
#, c-format
msgid "Could not get parameters on device '%s'"
msgstr ""
-#: sys/v4l2/gstv4l2object.c:3017
+#: sys/v4l2/gstv4l2object.c:3046
msgid "Video device did not accept new frame rate setting."
msgstr ""
-#: sys/v4l2/gstv4l2object.c:3122
+#: sys/v4l2/gstv4l2object.c:3151
msgid "Video device did not provide output format."
msgstr ""
-#: sys/v4l2/gstv4l2object.c:3128
+#: sys/v4l2/gstv4l2object.c:3157
msgid "Video device returned invalid dimensions."
msgstr ""
-#: sys/v4l2/gstv4l2object.c:3136
+#: sys/v4l2/gstv4l2object.c:3165
msgid "Video devices uses an unsupported interlacing method."
msgstr ""
-#: sys/v4l2/gstv4l2object.c:3143
+#: sys/v4l2/gstv4l2object.c:3172
msgid "Video devices uses an unsupported pixel format."
msgstr ""
-#: sys/v4l2/gstv4l2object.c:3565
+#: sys/v4l2/gstv4l2object.c:3584
msgid "Failed to configure internal buffer pool."
msgstr ""
-#: sys/v4l2/gstv4l2object.c:3571
+#: sys/v4l2/gstv4l2object.c:3590
msgid "Video device did not suggest any buffer size."
msgstr ""
-#: sys/v4l2/gstv4l2object.c:3586
+#: sys/v4l2/gstv4l2object.c:3605
msgid "No downstream pool to import from."
msgstr ""
@@ -367,7 +367,7 @@
msgid "Failed to allocated required memory."
msgstr ""
-#: sys/v4l2/gstv4l2src.c:509 sys/v4l2/gstv4l2videodec.c:581
+#: sys/v4l2/gstv4l2src.c:510 sys/v4l2/gstv4l2videodec.c:581
msgid "Failed to allocate required memory."
msgstr ""
diff --git a/po/hr.gmo b/po/hr.gmo
index b522a85..d31112d 100644
--- a/po/hr.gmo
+++ b/po/hr.gmo
Binary files differ
diff --git a/po/hr.po b/po/hr.po
index 419d159..e54d070 100644
--- a/po/hr.po
+++ b/po/hr.po
@@ -6,7 +6,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good 0.10.28.2\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2012-04-16 03:04+0200\n"
"Last-Translator: Tomislav Krznar <tomislav.krznar@gmail.com>\n"
"Language-Team: Croatian <lokalizacija@linux.hr>\n"
@@ -85,6 +85,9 @@
msgid "This file contains too many streams. Only playing first %d"
msgstr "Ova datoteka sadrži previše nizova. Reproduciram samo prvih %d"
+msgid "Internal data flow error."
+msgstr "Greška unutarnjeg toka podataka."
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -100,9 +103,6 @@
"Nisu pronađeni podržani nizovi. Morate dozvoliti više protokola prijenosa "
"ili vam možda nedostaje odgovarajući GStreamer RTSP priključak proširenja."
-msgid "Internal data flow error."
-msgstr "Greška unutarnjeg toka podataka."
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/hu.gmo b/po/hu.gmo
index c6a4f47..76de3cf 100644
--- a/po/hu.gmo
+++ b/po/hu.gmo
Binary files differ
diff --git a/po/hu.po b/po/hu.po
index bf03ea9..8da7fad 100644
--- a/po/hu.po
+++ b/po/hu.po
@@ -9,7 +9,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good 1.3.90\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2014-06-29 12:56+0200\n"
"Last-Translator: Balázs Úr <urbalazs@gmail.com>\n"
"Language-Team: Hungarian <translation-team-hu@lists.sourceforge.net>\n"
@@ -92,6 +92,9 @@
msgstr ""
"A fájl túl sok adatfolyamot tartalmaz. Csak az első %d kerül lejátszásra."
+msgid "Internal data flow error."
+msgstr "Belső adatfolyam-hiba."
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -107,9 +110,6 @@
"Nem található támogatott adatfolyam. Lehet, hogy több átviteli protokollt "
"kell engedélyezni, vagy hiányzik a megfelelő GStreamer RTSP bővítmény."
-msgid "Internal data flow error."
-msgstr "Belső adatfolyam-hiba."
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/id.gmo b/po/id.gmo
index ec600d2..2642ab4 100644
--- a/po/id.gmo
+++ b/po/id.gmo
Binary files differ
diff --git a/po/id.po b/po/id.po
index ed150c8..6579e9c 100644
--- a/po/id.po
+++ b/po/id.po
@@ -8,7 +8,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good 1.3.90\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2014-07-06 21:37+0700\n"
"Last-Translator: Andhika Padmawan <andhika.padmawan@gmail.com>\n"
"Language-Team: Indonesian <translation-team-id@lists.sourceforge.net>\n"
@@ -87,6 +87,9 @@
msgid "This file contains too many streams. Only playing first %d"
msgstr "Berkas ini berisi terlalu banyak arus. Hanya memutar %d pertama"
+msgid "Internal data flow error."
+msgstr "Galat aliran data internal."
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -103,9 +106,6 @@
"banyak transportasi atau kehilangan plugin ekstensi GStreamer RTSP yang "
"benar."
-msgid "Internal data flow error."
-msgstr "Galat aliran data internal."
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/it.gmo b/po/it.gmo
index ba828e2..0bf64ce 100644
--- a/po/it.gmo
+++ b/po/it.gmo
Binary files differ
diff --git a/po/it.po b/po/it.po
index ff2ef2f..e15857a 100644
--- a/po/it.po
+++ b/po/it.po
@@ -7,7 +7,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good 0.10.25.3\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2010-10-25 10:11+0200\n"
"Last-Translator: Luca Ferretti <elle.uca@infinito.it>\n"
"Language-Team: Italian <tp@lists.linux.it>\n"
@@ -86,6 +86,9 @@
msgid "This file contains too many streams. Only playing first %d"
msgstr "Questo file contiene troppi stream. Riprodotti solo i primi %d"
+msgid "Internal data flow error."
+msgstr "Errore interno nel flusso di dati."
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -103,9 +106,6 @@
"abilitare altri protocolli di trasporto oppure in alternativa potrebbe "
"mancare il corretto plugin GStreamer di tipo estensione RTSP."
-msgid "Internal data flow error."
-msgstr "Errore interno nel flusso di dati."
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/ja.gmo b/po/ja.gmo
index 68ce520..7f03f7a 100644
--- a/po/ja.gmo
+++ b/po/ja.gmo
Binary files differ
diff --git a/po/ja.po b/po/ja.po
index c3d0ad9..8c6b6e3 100644
--- a/po/ja.po
+++ b/po/ja.po
@@ -8,7 +8,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good 1.0.3\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2012-12-22 20:13+0900\n"
"Last-Translator: Takeshi Hamasaki <hmatrjp@users.sourceforge.jp>\n"
"Language-Team: Japanese <translation-team-ja@lists.sourceforge.net>\n"
@@ -89,6 +89,9 @@
"このファイルには多くのストリームが格納されているため、最初の %d 個のみ再生し"
"ます"
+msgid "Internal data flow error."
+msgstr "内部データフローエラー。"
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -105,9 +108,6 @@
"する必要があるかもしれません。または、正しい GStreamer RTSP 拡張プラグインが"
"ないのかもしれません。"
-msgid "Internal data flow error."
-msgstr "内部データフローエラー。"
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/lt.gmo b/po/lt.gmo
index b90ff43..9a854f9 100644
--- a/po/lt.gmo
+++ b/po/lt.gmo
Binary files differ
diff --git a/po/lt.po b/po/lt.po
index 1a34a12..8d37c9f 100644
--- a/po/lt.po
+++ b/po/lt.po
@@ -7,7 +7,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good 0.10.23.2\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2010-07-16 19:34+0300\n"
"Last-Translator: Žygimantas Beručka <uid0@akl.lt>\n"
"Language-Team: Lithuanian <komp_lt@konferencijos.lt>\n"
@@ -88,6 +88,9 @@
msgid "This file contains too many streams. Only playing first %d"
msgstr "Šiame faile per daug srautų. Grojami tik pirmieji %d"
+msgid "Internal data flow error."
+msgstr "Vidinė duomenų srauto klaida."
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -103,9 +106,6 @@
"Nerasta palaikomų srautų. Jums gali reikėti leisti daugiau transporto "
"protokolų, arba jūs neturite tinkamo GStreamer RTSP papildinio įskiepio."
-msgid "Internal data flow error."
-msgstr "Vidinė duomenų srauto klaida."
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/lv.gmo b/po/lv.gmo
index 3cec46d..ca7c1e1 100644
--- a/po/lv.gmo
+++ b/po/lv.gmo
Binary files differ
diff --git a/po/lv.po b/po/lv.po
index 32f0bf3..906ba1e 100644
--- a/po/lv.po
+++ b/po/lv.po
@@ -9,7 +9,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good 1.2.1\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2014-04-20 16:38+0300\n"
"Last-Translator: Rihards Prieditis <rprieditis@gmail.com>\n"
"Language-Team: Latvian <translation-team-lv@lists.sourceforge.net>\n"
@@ -88,6 +88,9 @@
msgid "This file contains too many streams. Only playing first %d"
msgstr "Šī datne satur pārāk daudz straumju. Tiek atskaņota tikai pirmā %d"
+msgid "Internal data flow error."
+msgstr "Iekšēja datu plūsmas kļūda."
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -103,9 +106,6 @@
"Netika atrasta atbalstīta straume. Iespējams, jums jāatļauj vairāk "
"transporta protokoli vai arī trūkst GStreamer RTSP paplašinājuma spraudnis. "
-msgid "Internal data flow error."
-msgstr "Iekšēja datu plūsmas kļūda."
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/mt.gmo b/po/mt.gmo
index 5efcf7e..9480ded 100644
--- a/po/mt.gmo
+++ b/po/mt.gmo
Binary files differ
diff --git a/po/mt.po b/po/mt.po
index 035814a..6470f8b 100644
--- a/po/mt.po
+++ b/po/mt.po
@@ -5,7 +5,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good-0.10.10.3\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2008-10-26 19:09+0100\n"
"Last-Translator: Michel Bugeja <michelbugeja@rabatmalta.com>\n"
"Language-Team: Maltese <translation-team-mt@lists.sourceforge.net>\n"
@@ -86,6 +86,9 @@
msgid "This file contains too many streams. Only playing first %d"
msgstr "Il-fajl fiħ aktar streams milli suppost. Ser indoqq l-ewwel %d"
+msgid "Internal data flow error."
+msgstr "Problema interna ta' data flow"
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -97,9 +100,6 @@
"plugin."
msgstr ""
-msgid "Internal data flow error."
-msgstr "Problema interna ta' data flow"
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/nb.gmo b/po/nb.gmo
index 256cecf..9de6d05 100644
--- a/po/nb.gmo
+++ b/po/nb.gmo
Binary files differ
diff --git a/po/nb.po b/po/nb.po
index 916ea99..15c8db3 100644
--- a/po/nb.po
+++ b/po/nb.po
@@ -8,7 +8,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good 1.2.1\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2014-01-22 17:49+0100\n"
"Last-Translator: Johnny A. Solbu <johnny@solbu.net>\n"
"Language-Team: Norwegian Bokmaal <i18n-nb@lister.ping.uio.no>\n"
@@ -85,6 +85,9 @@
msgid "This file contains too many streams. Only playing first %d"
msgstr "Filen inneholder for mange strømmer. Spiller bare første %d"
+msgid "Internal data flow error."
+msgstr "Intern dataflytfeil."
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -101,9 +104,6 @@
"transportprotokoller eller den kan mangle den rette GStreamer RTSP-"
"utvidelsestillegget."
-msgid "Internal data flow error."
-msgstr "Intern dataflytfeil."
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/nl.gmo b/po/nl.gmo
index aa3b7e3..631c8cd 100644
--- a/po/nl.gmo
+++ b/po/nl.gmo
Binary files differ
diff --git a/po/nl.po b/po/nl.po
index e32ef5e..26309ee 100644
--- a/po/nl.po
+++ b/po/nl.po
@@ -6,7 +6,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good 1.3.2\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2014-05-22 00:41+0200\n"
"Last-Translator: Freek de Kruijf <f.de.kruijf@gmail.com>\n"
"Language-Team: Dutch <vertaling@vrijschrift.org>\n"
@@ -87,6 +87,9 @@
msgstr ""
"Dit bestand bevat te veel stromen. Alleen de eerste %d worden afgespeeld."
+msgid "Internal data flow error."
+msgstr "Interne fout in de gegevensdoorvoer."
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -103,9 +106,6 @@
"overdrachtsprotocollen toestaan of anders ontbreekt de juiste GStreamer RTSP "
"extensie-plugin."
-msgid "Internal data flow error."
-msgstr "Interne fout in de gegevensdoorvoer."
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/or.gmo b/po/or.gmo
index ed3cadd..1431ad9 100644
--- a/po/or.gmo
+++ b/po/or.gmo
Binary files differ
diff --git a/po/or.po b/po/or.po
index 08c9017..74b2966 100644
--- a/po/or.po
+++ b/po/or.po
@@ -8,7 +8,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-0.8.3\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2004-09-27 13:32+0530\n"
"Last-Translator: Gora Mohanty <gora_mohanty@yahoo.co.in>\n"
"Language-Team: Oriya <gora_mohanty@yahoo.co.in>\n"
@@ -88,6 +88,9 @@
msgid "This file contains too many streams. Only playing first %d"
msgstr ""
+msgid "Internal data flow error."
+msgstr ""
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -99,9 +102,6 @@
"plugin."
msgstr ""
-msgid "Internal data flow error."
-msgstr ""
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/pl.gmo b/po/pl.gmo
index 55c13f6..cc31e65 100644
--- a/po/pl.gmo
+++ b/po/pl.gmo
Binary files differ
diff --git a/po/pl.po b/po/pl.po
index 035d370..b346433 100644
--- a/po/pl.po
+++ b/po/pl.po
@@ -6,7 +6,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good 1.3.90\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2014-06-29 10:45+0200\n"
"Last-Translator: Jakub Bogusz <qboosh@pld-linux.org>\n"
"Language-Team: Polish <translation-team-pl@lists.sourceforge.net>\n"
@@ -83,6 +83,9 @@
msgid "This file contains too many streams. Only playing first %d"
msgstr "Ten plik zawiera zbyt dużo strumieni. Odtwarzanie tylko %d pierwszych"
+msgid "Internal data flow error."
+msgstr "Błąd wewnętrzny przepływu danych."
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -99,9 +102,6 @@
"protokołów transportu, albo może brakować właściwej wtyczki rozszerzenia "
"RTSP GStreamera."
-msgid "Internal data flow error."
-msgstr "Błąd wewnętrzny przepływu danych."
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/pt_BR.gmo b/po/pt_BR.gmo
index 37bfb11..6055b68 100644
--- a/po/pt_BR.gmo
+++ b/po/pt_BR.gmo
Binary files differ
diff --git a/po/pt_BR.po b/po/pt_BR.po
index ece5c94..9925fa2 100644
--- a/po/pt_BR.po
+++ b/po/pt_BR.po
@@ -11,7 +11,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good-1.2.1\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2013-12-29 17:48-0200\n"
"Last-Translator: Fabrício Godoy <skarllot@gmail.com>\n"
"Language-Team: Brazilian Portuguese <ldpbr-translation@lists.sourceforge."
@@ -92,6 +92,9 @@
msgid "This file contains too many streams. Only playing first %d"
msgstr "Este arquivo contêm muitos fluxos. Apenas reproduzindo o primeiro %d"
+msgid "Internal data flow error."
+msgstr "Erro interno no fluxo de dados."
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -108,9 +111,6 @@
"protocolos de transporte sejam permitidos ou senão o plug-in de extensão "
"GStreamer RTSP pode estar incorreto."
-msgid "Internal data flow error."
-msgstr "Erro interno no fluxo de dados."
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/ro.gmo b/po/ro.gmo
index 7f1c6a3..8b6acd6 100644
--- a/po/ro.gmo
+++ b/po/ro.gmo
Binary files differ
diff --git a/po/ro.po b/po/ro.po
index 2077bd2..bad6971 100644
--- a/po/ro.po
+++ b/po/ro.po
@@ -5,7 +5,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good 0.10.23.2\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2010-08-16 03:22+0300\n"
"Last-Translator: Lucian Adrian Grijincu <lucian.grijincu@gmail.com>\n"
"Language-Team: Romanian <translation-team-ro@lists.sourceforge.net>\n"
@@ -88,6 +88,9 @@
msgid "This file contains too many streams. Only playing first %d"
msgstr "Fișierul conține prea multe fluxuri. Se redau doar primele %d"
+msgid "Internal data flow error."
+msgstr "Eroare internă a fluxului de date."
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -104,9 +107,6 @@
"multor protocoale de transport sau de altfel s-ar putea să lipsească modulul "
"corect de extensie GStreamer RTSP."
-msgid "Internal data flow error."
-msgstr "Eroare internă a fluxului de date."
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/ru.gmo b/po/ru.gmo
index 6f205a1..124c346 100644
--- a/po/ru.gmo
+++ b/po/ru.gmo
Binary files differ
diff --git a/po/ru.po b/po/ru.po
index 0227a3d..7f7f827 100644
--- a/po/ru.po
+++ b/po/ru.po
@@ -9,7 +9,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good 1.3.90\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2014-06-29 18:59+0400\n"
"Last-Translator: Yuri Kozlov <yuray@komyakino.ru>\n"
"Language-Team: Russian <gnu@mx.ru>\n"
@@ -93,6 +93,9 @@
msgstr ""
"Файл содержит слишком много потоков. Будет воспроизведёно только первых %d"
+msgid "Internal data flow error."
+msgstr "Внутренняя ошибка потока данных."
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -108,9 +111,6 @@
"Ни один поток не поддерживается. Возможно требуется разрешить дополнительные "
"протоколы передачи или же отсутствует правильный модуль GStreamer RTSP."
-msgid "Internal data flow error."
-msgstr "Внутренняя ошибка потока данных."
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/sk.gmo b/po/sk.gmo
index d4c2aa3..a16191d 100644
--- a/po/sk.gmo
+++ b/po/sk.gmo
Binary files differ
diff --git a/po/sk.po b/po/sk.po
index 3fc2be4..f3c7960 100644
--- a/po/sk.po
+++ b/po/sk.po
@@ -6,7 +6,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good 1.2.1\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2014-01-30 11:08+0100\n"
"Last-Translator: Peter Tuharsky <tuharsky@misbb.sk>\n"
"Language-Team: Slovak <sk-i18n@lists.linux.sk>\n"
@@ -86,6 +86,9 @@
msgid "This file contains too many streams. Only playing first %d"
msgstr "Tento súbor obsahuje príliš mnoho prúdov údajov. Prehrávam iba prvý %d"
+msgid "Internal data flow error."
+msgstr "Vnútorná chyba toku údajov."
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -102,9 +105,6 @@
"viacero prenosových protokolov, alebo vám možno chýba správny zásuvný modul "
"GStreamer RTSP."
-msgid "Internal data flow error."
-msgstr "Vnútorná chyba toku údajov."
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/sl.gmo b/po/sl.gmo
index 4c76498..d19bd16 100644
--- a/po/sl.gmo
+++ b/po/sl.gmo
Binary files differ
diff --git a/po/sl.po b/po/sl.po
index 3499af2..ad2aaf6 100644
--- a/po/sl.po
+++ b/po/sl.po
@@ -10,7 +10,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good-1.2.1\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2014-04-09 22:52+0100\n"
"Last-Translator: Klemen Košir <klemen913@gmail.com>\n"
"Language-Team: Slovenian <translation-team-sl@lists.sourceforge.net>\n"
@@ -89,6 +89,9 @@
msgid "This file contains too many streams. Only playing first %d"
msgstr "Datoteka vsebuje več pretokov. Predvajano bo le začetnih %d"
+msgid "Internal data flow error."
+msgstr "Notranja napaka pretoka podatkov."
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -105,9 +108,6 @@
"protokolov prenosa, oziroma manjka ustrezen vstavek GStreamer RTSP "
"razširitve."
-msgid "Internal data flow error."
-msgstr "Notranja napaka pretoka podatkov."
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/sq.gmo b/po/sq.gmo
index 4d36442..fe6afba 100644
--- a/po/sq.gmo
+++ b/po/sq.gmo
Binary files differ
diff --git a/po/sq.po b/po/sq.po
index ce00228..13ebeff 100644
--- a/po/sq.po
+++ b/po/sq.po
@@ -6,7 +6,7 @@
msgstr ""
"Project-Id-Version: gst-plugins 0.8.3\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2004-08-07 20:29+0200\n"
"Last-Translator: Laurent Dhima <laurenti@alblinux.net>\n"
"Language-Team: Albanian <begraj@hotmail.com>\n"
@@ -86,6 +86,9 @@
msgid "This file contains too many streams. Only playing first %d"
msgstr ""
+msgid "Internal data flow error."
+msgstr ""
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -97,9 +100,6 @@
"plugin."
msgstr ""
-msgid "Internal data flow error."
-msgstr ""
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/sr.gmo b/po/sr.gmo
index 4f0705a..0e7766d 100644
--- a/po/sr.gmo
+++ b/po/sr.gmo
Binary files differ
diff --git a/po/sr.po b/po/sr.po
index abc70c9..a9e2259 100644
--- a/po/sr.po
+++ b/po/sr.po
@@ -7,7 +7,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good-1.3.2\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2014-06-18 19:56+0200\n"
"Last-Translator: Мирослав Николић <miroslavnikolic@rocketmail.com>\n"
"Language-Team: Serbian <(nothing)>\n"
@@ -86,6 +86,9 @@
msgid "This file contains too many streams. Only playing first %d"
msgstr "Ова датотека садржи превише токова. Пуштам само први %d"
+msgid "Internal data flow error."
+msgstr "Унутрaшња грешка протока података."
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -102,9 +105,6 @@
"протокола преноса или можда на неки други начин недостаје прави прикључак "
"РТСП проширења ГСтримера."
-msgid "Internal data flow error."
-msgstr "Унутрaшња грешка протока података."
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/sv.gmo b/po/sv.gmo
index 4658a00..438f34e 100644
--- a/po/sv.gmo
+++ b/po/sv.gmo
Binary files differ
diff --git a/po/sv.po b/po/sv.po
index faa3379..603151e 100644
--- a/po/sv.po
+++ b/po/sv.po
@@ -9,7 +9,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good 1.3.90\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2014-07-11 02:52+0200\n"
"Last-Translator: Sebastian Rasmussen <sebras@gmail.com>\n"
"Language-Team: Swedish <tp-sv@listor.tp-sv.se>\n"
@@ -86,6 +86,9 @@
msgstr ""
"Den här filen innehåller för många strömmar. Spelar endast upp de första %d"
+msgid "Internal data flow error."
+msgstr "Internt fel i dataflöde."
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -102,9 +105,6 @@
"transportprotokoll eller så saknar du kanske det rätta GStreamer-tillägget "
"för RTSP."
-msgid "Internal data flow error."
-msgstr "Internt fel i dataflöde."
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/tr.gmo b/po/tr.gmo
index 42eb33f..8032a25 100644
--- a/po/tr.gmo
+++ b/po/tr.gmo
Binary files differ
diff --git a/po/tr.po b/po/tr.po
index cdecfa4..f0efdff 100644
--- a/po/tr.po
+++ b/po/tr.po
@@ -7,7 +7,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good 1.0.3\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2013-03-10 13:05+0200\n"
"Last-Translator: Server Acim <serveracim@gmail.com>\n"
"Language-Team: Turkish <gnu-tr-u12a@lists.sourceforge.net>\n"
@@ -85,6 +85,9 @@
msgid "This file contains too many streams. Only playing first %d"
msgstr "Bu dosya çok fazla akış içeriyor. Sadece ilki oynatılıyor %d"
+msgid "Internal data flow error."
+msgstr "İç veri akış hatası."
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -100,9 +103,6 @@
"Desteklenen akış bulunamadı. Daha fazla aktarım protokolüne izin vermeniz "
"veya doğru GStreamer RTSP uzantı eklentisine onay vermeniz gerekebilir."
-msgid "Internal data flow error."
-msgstr "İç veri akış hatası."
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/uk.gmo b/po/uk.gmo
index 78c1ab2..10dce38 100644
--- a/po/uk.gmo
+++ b/po/uk.gmo
Binary files differ
diff --git a/po/uk.po b/po/uk.po
index 211bf65..f42f82f 100644
--- a/po/uk.po
+++ b/po/uk.po
@@ -8,7 +8,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good 1.3.90\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2014-06-29 11:40+0300\n"
"Last-Translator: Yuri Chornoivan <yurchor@ukr.net>\n"
"Language-Team: Ukrainian <translation-team-uk@lists.sourceforge.net>\n"
@@ -93,6 +93,9 @@
msgid "This file contains too many streams. Only playing first %d"
msgstr "Файл містить надто багато потоків. Відтворюються лише перші %d"
+msgid "Internal data flow error."
+msgstr "Помилка внутрішнього перенесення даних."
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -108,9 +111,6 @@
"Не знайдено підтримуваних потоків. Ймовірно, вам слід увімкнути додаткові "
"протоколи передавання даних або встановити додатковий модуль RTSP GStreamer."
-msgid "Internal data flow error."
-msgstr "Помилка внутрішнього перенесення даних."
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/vi.gmo b/po/vi.gmo
index 8821c82..00073b3 100644
--- a/po/vi.gmo
+++ b/po/vi.gmo
Binary files differ
diff --git a/po/vi.po b/po/vi.po
index 3a3b179..1c91135 100644
--- a/po/vi.po
+++ b/po/vi.po
@@ -8,7 +8,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good 1.3.90\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2014-06-30 09:12+0700\n"
"Last-Translator: Trần Ngọc Quân <vnwildman@gmail.com>\n"
"Language-Team: Vietnamese <translation-team-vi@lists.sourceforge.net>\n"
@@ -92,6 +92,9 @@
msgid "This file contains too many streams. Only playing first %d"
msgstr "Tập tin này chứa quá nhiều luồng nên chỉ phát %d đầu tiên"
+msgid "Internal data flow error."
+msgstr "Lỗi luồng dữ liệu nội bộ."
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -107,9 +110,6 @@
"Không tìm thấy luồng dữ liệu được hỗ trợ. Người dùng có thể cần cho phép "
"thêm giao thức truyền, hoặc chưa cài đặt phần bổ sung RTSP Gstreamer đúng."
-msgid "Internal data flow error."
-msgstr "Lỗi luồng dữ liệu nội bộ."
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/zh_CN.gmo b/po/zh_CN.gmo
index 6893ccc..c26f323 100644
--- a/po/zh_CN.gmo
+++ b/po/zh_CN.gmo
Binary files differ
diff --git a/po/zh_CN.po b/po/zh_CN.po
index 7702952..a641a12 100644
--- a/po/zh_CN.po
+++ b/po/zh_CN.po
@@ -7,7 +7,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good 0.10.16.2\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2009-11-13 22:20+0800\n"
"Last-Translator: Ji ZhengYu <zhengyuji@gmail.com>\n"
"Language-Team: Chinese (simplified) <i18n-zh@googlegroups.com>\n"
@@ -86,6 +86,9 @@
msgid "This file contains too many streams. Only playing first %d"
msgstr "此文件包含了太多的流。只播放前面 %d 个"
+msgid "Internal data flow error."
+msgstr "内部数据流错误。"
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -101,9 +104,6 @@
"未找到支持的流媒体。您可能需要启用更多的传送协议，或者也有可能是缺少正确的 "
"GStreamer RTSP 扩展插件。"
-msgid "Internal data flow error."
-msgstr "内部数据流错误。"
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/zh_HK.gmo b/po/zh_HK.gmo
index e0a3a79..bde5918 100644
--- a/po/zh_HK.gmo
+++ b/po/zh_HK.gmo
Binary files differ
diff --git a/po/zh_HK.po b/po/zh_HK.po
index d2b2bd0..9822f9a 100644
--- a/po/zh_HK.po
+++ b/po/zh_HK.po
@@ -6,7 +6,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good-0.10.2 0.10.2\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2006-08-29 01:08+0800\n"
"Last-Translator: Abel Cheung <abelcheung@gmail.com>\n"
"Language-Team: Chinese (Hong Kong) <community@linuxhall.org>\n"
@@ -82,6 +82,10 @@
msgid "This file contains too many streams. Only playing first %d"
msgstr ""
+#, fuzzy
+msgid "Internal data flow error."
+msgstr "內部資料串流發生錯誤。"
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -93,10 +97,6 @@
"plugin."
msgstr ""
-#, fuzzy
-msgid "Internal data flow error."
-msgstr "內部資料串流發生錯誤。"
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/po/zh_TW.gmo b/po/zh_TW.gmo
index 1b793b8..68499a0 100644
--- a/po/zh_TW.gmo
+++ b/po/zh_TW.gmo
Binary files differ
diff --git a/po/zh_TW.po b/po/zh_TW.po
index e692846..42899f0 100644
--- a/po/zh_TW.po
+++ b/po/zh_TW.po
@@ -6,7 +6,7 @@
msgstr ""
"Project-Id-Version: gst-plugins-good-0.10.2 0.10.2\n"
"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
-"POT-Creation-Date: 2015-03-17 09:11+0100\n"
+"POT-Creation-Date: 2015-05-13 13:04+0300\n"
"PO-Revision-Date: 2006-08-29 01:08+0800\n"
"Last-Translator: Abel Cheung <abelcheung@gmail.com>\n"
"Language-Team: Chinese (traditional) <zh-l10n@linux.org.tw>\n"
@@ -82,6 +82,10 @@
msgid "This file contains too many streams. Only playing first %d"
msgstr ""
+#, fuzzy
+msgid "Internal data flow error."
+msgstr "內部資料串流發生錯誤。"
+
msgid ""
"No supported stream was found. You might need to install a GStreamer RTSP "
"extension plugin for Real media streams."
@@ -93,10 +97,6 @@
"plugin."
msgstr ""
-#, fuzzy
-msgid "Internal data flow error."
-msgstr "內部資料串流發生錯誤。"
-
msgid ""
"Could not open audio device for playback. Device is being used by another "
"application."
diff --git a/sys/osxaudio/gstosxcoreaudio.h b/sys/osxaudio/gstosxcoreaudio.h
index 8b509aa..b319ced 100644
--- a/sys/osxaudio/gstosxcoreaudio.h
+++ b/sys/osxaudio/gstosxcoreaudio.h
@@ -68,7 +68,7 @@
#define CORE_AUDIO_FORMAT_IS_SPDIF(f) ((f).mFormat.mFormatID == 'IAC3' || (f).mFormat.mFormatID == 'iac3' || (f).mFormat.mFormatID == kAudioFormat60958AC3 || (f).mFormat.mFormatID == kAudioFormatAC3)
#define CORE_AUDIO_FORMAT "FormatID: %" GST_FOURCC_FORMAT " rate: %f flags: 0x%x BytesPerPacket: %u FramesPerPacket: %u BytesPerFrame: %u ChannelsPerFrame: %u BitsPerChannel: %u"
-#define CORE_AUDIO_FORMAT_ARGS(f) GST_FOURCC_ARGS((f).mFormatID),(f).mSampleRate,(unsigned)(f).mFormatFlags,(unsigned)(f).mBytesPerPacket,(unsigned)(f).mFramesPerPacket,(unsigned)(f).mBytesPerFrame,(unsigned)(f).mChannelsPerFrame,(unsigned)(f).mBitsPerChannel
+#define CORE_AUDIO_FORMAT_ARGS(f) GST_FOURCC_ARGS((unsigned int)(f).mFormatID),(f).mSampleRate,(unsigned int)(f).mFormatFlags,(unsigned int)(f).mBytesPerPacket,(unsigned int)(f).mFramesPerPacket,(unsigned int)(f).mBytesPerFrame,(unsigned int)(f).mChannelsPerFrame,(unsigned int)(f).mBitsPerChannel
typedef struct _GstCoreAudio GstCoreAudio;
typedef struct _GstCoreAudioClass GstCoreAudioClass;
diff --git a/sys/osxaudio/gstosxcoreaudiohal.c b/sys/osxaudio/gstosxcoreaudiohal.c
index f1ebdf5..776d765 100644
--- a/sys/osxaudio/gstosxcoreaudiohal.c
+++ b/sys/osxaudio/gstosxcoreaudiohal.c
@@ -341,7 +341,7 @@
status = AudioObjectGetPropertyDataSize (device_id,
&channelLayoutAddress, 0, NULL, &propertySize);
if (status != noErr) {
- GST_ERROR ("failed to get prefered layout: %d", (int) status);
+ GST_ERROR ("failed to get preferred layout: %d", (int) status);
goto beach;
}
@@ -350,7 +350,7 @@
status = AudioObjectGetPropertyData (device_id,
&channelLayoutAddress, 0, NULL, &propertySize, layout);
if (status != noErr) {
- GST_ERROR ("failed to get prefered layout: %d", (int) status);
+ GST_ERROR ("failed to get preferred layout: %d", (int) status);
goto failed;
}
diff --git a/sys/osxvideo/cocoawindow.m b/sys/osxvideo/cocoawindow.m
index 57d19a9..71e1ca0 100644
--- a/sys/osxvideo/cocoawindow.m
+++ b/sys/osxvideo/cocoawindow.m
@@ -336,7 +336,9 @@
NSOpenGLPFAColorSize, 24,
NSOpenGLPFAAlphaSize, 8,
NSOpenGLPFADepthSize, 24,
+#if MAC_OS_X_VERSION_MAX_ALLOWED < 1090
NSOpenGLPFAWindow,
+#endif
0
};
@@ -573,7 +575,9 @@
NSOpenGLPFAColorSize, 24,
NSOpenGLPFAAlphaSize, 8,
NSOpenGLPFADepthSize, 24,
+#if MAC_OS_X_VERSION_MAX_ALLOWED < 1060
NSOpenGLPFAFullScreen,
+#endif
NSOpenGLPFAScreenMask,
CGDisplayIDToOpenGLDisplayMask (kCGDirectMainDisplay),
0
@@ -602,7 +606,9 @@
GST_WARNING ("CGCaptureAllDisplays() failed");
return;
}
+#if MAC_OS_X_VERSION_MAX_ALLOWED < 1070
[fullScreenContext setFullScreen];
+#endif
[fullScreenContext makeCurrentContext];
fullscreen = YES;
diff --git a/sys/v4l2/gstv4l2bufferpool.c b/sys/v4l2/gstv4l2bufferpool.c
index 4815336..7c963cc 100644
--- a/sys/v4l2/gstv4l2bufferpool.c
+++ b/sys/v4l2/gstv4l2bufferpool.c
@@ -549,6 +549,13 @@
GST_INFO_OBJECT (pool, "increasing minimum buffers to %u", min_buffers);
}
+ /* respect driver requirements */
+ if (min_buffers < obj->min_buffers) {
+ updated = TRUE;
+ min_buffers = obj->min_buffers;
+ GST_INFO_OBJECT (pool, "increasing minimum buffers to %u", min_buffers);
+ }
+
if (max_buffers > VIDEO_MAX_FRAME || max_buffers == 0) {
updated = TRUE;
max_buffers = VIDEO_MAX_FRAME;
@@ -1240,6 +1247,8 @@
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_CORRUPTED);
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
+ GST_BUFFER_OFFSET (outbuf) = group->buffer.sequence;
+ GST_BUFFER_OFFSET_END (outbuf) = group->buffer.sequence + 1;
done:
*buffer = outbuf;
diff --git a/sys/v4l2/gstv4l2object.c b/sys/v4l2/gstv4l2object.c
index 9c8dfb2..816b42c 100644
--- a/sys/v4l2/gstv4l2object.c
+++ b/sys/v4l2/gstv4l2object.c
@@ -721,6 +721,28 @@
}
static void
+gst_v4l2_get_driver_min_buffers (GstV4l2Object * v4l2object)
+{
+ int min;
+ gboolean ret = FALSE;
+
+ /* Certain driver may expose a minimum number of buffers through controls. */
+ /* If the ioctl is not supported by the driver, min_buffers remains zero. */
+ ret = gst_v4l2_get_attribute (v4l2object,
+ V4L2_TYPE_IS_OUTPUT (v4l2object->type)
+ ? V4L2_CID_MIN_BUFFERS_FOR_OUTPUT : V4L2_CID_MIN_BUFFERS_FOR_CAPTURE,
+ &min);
+
+ if (ret) {
+ GST_DEBUG_OBJECT (v4l2object->element,
+ "driver requires a minimum of %d buffers", min);
+ v4l2object->min_buffers = min;
+ } else {
+ v4l2object->min_buffers = 0;
+ }
+}
+
+static void
gst_v4l2_set_defaults (GstV4l2Object * v4l2object)
{
GstTunerNorm *norm = NULL;
@@ -1731,7 +1753,7 @@
{
struct v4l2_format fmt;
GValue interlace_formats = { 0, };
- GstVideoInterlaceMode interlace_mode;
+ GstVideoInterlaceMode interlace_mode, prev = -1;
const gchar *mode_strings[] = { "progressive",
"interleaved",
@@ -1762,6 +1784,7 @@
g_value_init (&interlace_enum, G_TYPE_STRING);
g_value_set_string (&interlace_enum, mode_strings[interlace_mode]);
gst_value_list_append_and_take_value (&interlace_formats, &interlace_enum);
+ prev = interlace_mode;
}
memset (&fmt, 0, sizeof (fmt));
@@ -1772,7 +1795,8 @@
fmt.fmt.pix.field = V4L2_FIELD_INTERLACED;
if (gst_v4l2_object_try_fmt (v4l2object, &fmt) == 0 &&
- gst_v4l2_object_get_interlace_mode (fmt.fmt.pix.field, &interlace_mode)) {
+ gst_v4l2_object_get_interlace_mode (fmt.fmt.pix.field, &interlace_mode) &&
+ prev != interlace_mode) {
GValue interlace_enum = { 0, };
g_value_init (&interlace_enum, G_TYPE_STRING);
g_value_set_string (&interlace_enum, mode_strings[interlace_mode]);
@@ -2407,6 +2431,11 @@
GST_INFO_OBJECT (v4l2object->element, "accessing buffers via mode %d", mode);
v4l2object->mode = mode;
+ /* If min_buffers is not set, the driver either does not support the control or
+ it has not been asked yet via propose_allocation/decide_allocation. */
+ if (!v4l2object->min_buffers)
+ gst_v4l2_get_driver_min_buffers (v4l2object);
+
/* Map the buffers */
GST_LOG_OBJECT (v4l2object->element, "initiating buffer pool");
@@ -3315,7 +3344,6 @@
gboolean update;
gboolean has_video_meta;
gboolean can_share_own_pool, pushing_from_our_pool = FALSE;
- struct v4l2_control ctl = { 0, };
GstAllocator *allocator = NULL;
GstAllocationParams params = { 0 };
@@ -3352,16 +3380,7 @@
can_share_own_pool = (has_video_meta || !obj->need_video_meta);
- /* Certain driver may expose a minimum through controls */
- ctl.id = V4L2_CID_MIN_BUFFERS_FOR_CAPTURE;
- if (v4l2_ioctl (obj->video_fd, VIDIOC_G_CTRL, &ctl) >= 0) {
- GST_DEBUG_OBJECT (obj->element, "driver require a minimum of %d buffers",
- ctl.value);
- obj->min_buffers = ctl.value;
- } else {
- obj->min_buffers = 0;
- }
-
+ gst_v4l2_get_driver_min_buffers (obj);
/* We can't share our own pool, if it exceed V4L2 capacity */
if (min + obj->min_buffers + 1 > VIDEO_MAX_FRAME)
can_share_own_pool = FALSE;
@@ -3597,7 +3616,6 @@
guint size, min, max;
GstCaps *caps;
gboolean need_pool;
- struct v4l2_control ctl = { 0, };
/* Set defaults allocation parameters */
size = obj->info.size;
@@ -3629,16 +3647,7 @@
}
gst_structure_free (config);
}
-
- /* Some devices may expose a minimum */
- ctl.id = V4L2_CID_MIN_BUFFERS_FOR_OUTPUT;
- if (v4l2_ioctl (obj->video_fd, VIDIOC_G_CTRL, &ctl) >= 0) {
- GST_DEBUG_OBJECT (obj->element, "driver require a miminum of %d buffers",
- ctl.value);
- obj->min_buffers = ctl.value;
- } else {
- obj->min_buffers = 0;
- }
+ gst_v4l2_get_driver_min_buffers (obj);
min = MAX (obj->min_buffers, GST_V4L2_MIN_BUFFERS);
diff --git a/sys/v4l2/gstv4l2src.c b/sys/v4l2/gstv4l2src.c
index 28a2d0f..633e46f 100644
--- a/sys/v4l2/gstv4l2src.c
+++ b/sys/v4l2/gstv4l2src.c
@@ -459,9 +459,10 @@
* Basesrc will do an allocation query that
* should indirectly reclaim buffers, after that we can
* set the format and then configure our pool */
- if (gst_v4l2_object_try_format (obj, caps))
+ if (gst_v4l2_object_try_format (obj, caps)) {
+ v4l2src->renegotiation_adjust = v4l2src->offset + 1;
v4l2src->pending_set_fmt = TRUE;
- else
+ } else
return FALSE;
} else {
/* make sure we stop capturing and dealloc buffers */
@@ -588,6 +589,7 @@
GstV4l2Src *v4l2src = GST_V4L2SRC (src);
v4l2src->offset = 0;
+ v4l2src->renegotiation_adjust = 0;
/* activate settings for first frame */
v4l2src->ctrl_time = 0;
@@ -675,6 +677,7 @@
GstClock *clock;
GstClockTime abs_time, base_time, timestamp, duration;
GstClockTime delay;
+ GstMessage *qos_msg;
do {
ret = GST_BASE_SRC_CLASS (parent_class)->alloc (GST_BASE_SRC (src), 0,
@@ -778,8 +781,6 @@
}
/* set buffer metadata */
- GST_BUFFER_OFFSET (*buf) = v4l2src->offset++;
- GST_BUFFER_OFFSET_END (*buf) = v4l2src->offset;
if (G_LIKELY (abs_time != GST_CLOCK_TIME_NONE)) {
/* the time now is the time of the clock minus the base time */
@@ -808,6 +809,32 @@
GST_INFO_OBJECT (src, "sync to %" GST_TIME_FORMAT " out ts %" GST_TIME_FORMAT,
GST_TIME_ARGS (v4l2src->ctrl_time), GST_TIME_ARGS (timestamp));
+ /* use generated offset values only if there are not already valid ones
+ * set by the v4l2 device */
+ if (!GST_BUFFER_OFFSET_IS_VALID (*buf) || !GST_BUFFER_OFFSET_END_IS_VALID (*buf)) {
+ GST_BUFFER_OFFSET (*buf) = v4l2src->offset++;
+ GST_BUFFER_OFFSET_END (*buf) = v4l2src->offset;
+ } else {
+ /* adjust raw v4l2 device sequence, will restart at null in case of renegotiation
+ * (streamoff/streamon) */
+ GST_BUFFER_OFFSET (*buf) += v4l2src->renegotiation_adjust;
+ GST_BUFFER_OFFSET_END (*buf) += v4l2src->renegotiation_adjust;
+ /* check for frame loss with given (from v4l2 device) buffer offset */
+ if ((v4l2src->offset != 0) && (GST_BUFFER_OFFSET (*buf) != (v4l2src->offset + 1))) {
+ guint64 lost_frame_count = GST_BUFFER_OFFSET (*buf) - v4l2src->offset - 1;
+ GST_WARNING_OBJECT (v4l2src,
+ "lost frames detected: count = %" G_GUINT64_FORMAT " - ts: %" GST_TIME_FORMAT,
+ lost_frame_count, GST_TIME_ARGS (timestamp));
+
+ qos_msg = gst_message_new_qos (GST_OBJECT_CAST (v4l2src), TRUE,
+ GST_CLOCK_TIME_NONE, GST_CLOCK_TIME_NONE, timestamp,
+ GST_CLOCK_TIME_IS_VALID (duration) ? lost_frame_count * duration : GST_CLOCK_TIME_NONE);
+ gst_element_post_message (GST_ELEMENT_CAST (v4l2src), qos_msg);
+
+ }
+ v4l2src->offset = GST_BUFFER_OFFSET (*buf);
+ }
+
GST_BUFFER_TIMESTAMP (*buf) = timestamp;
GST_BUFFER_DURATION (*buf) = duration;
diff --git a/sys/v4l2/gstv4l2src.h b/sys/v4l2/gstv4l2src.h
index 79dec17..90a0dd6 100644
--- a/sys/v4l2/gstv4l2src.h
+++ b/sys/v4l2/gstv4l2src.h
@@ -59,6 +59,9 @@
guint64 offset;
+ /* offset adjust after renegotiation */
+ guint64 renegotiation_adjust;
+
GstClockTime ctrl_time;
gboolean pending_set_fmt;
diff --git a/sys/v4l2/v4l2_calls.c b/sys/v4l2/v4l2_calls.c
index d1cc284..356d42f 100644
--- a/sys/v4l2/v4l2_calls.c
+++ b/sys/v4l2/v4l2_calls.c
@@ -842,7 +842,7 @@
freq.tuner = tunernum;
/* fill in type - ignore error */
- v4l2_ioctl (v4l2object->video_fd, VIDIOC_G_FREQUENCY, &freq);
+ (void) v4l2_ioctl (v4l2object->video_fd, VIDIOC_G_FREQUENCY, &freq);
freq.frequency = frequency / channel->freq_multiplicator;
if (v4l2_ioctl (v4l2object->video_fd, VIDIOC_S_FREQUENCY, &freq) < 0)
diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am
index 407e11f..a64f285 100644
--- a/tests/check/Makefile.am
+++ b/tests/check/Makefile.am
@@ -6,7 +6,7 @@
REGISTRY_ENVIRONMENT = \
GST_REGISTRY_1_0=$(CHECK_REGISTRY)
-TESTS_ENVIRONMENT = \
+AM_TESTS_ENVIRONMENT = \
$(REGISTRY_ENVIRONMENT) \
GST_PLUGIN_SYSTEM_PATH_1_0= \
GST_PLUGIN_PATH_1_0=$(top_builddir)/gst:$(top_builddir)/ext:$(top_builddir)/sys:$(GSTPB_PLUGINS_DIR):$(GST_PLUGINS_DIR) \
@@ -37,7 +37,9 @@
distclean-local: distclean-local-orc
if USE_PLUGIN_ALPHA
-check_alpha = elements/alphacolor
+check_alpha = \
+ elements/alphacolor \
+ elements/alpha
else
check_alpha =
endif
@@ -392,6 +394,7 @@
AM_CFLAGS = $(GST_OBJ_CFLAGS) $(GST_CHECK_CFLAGS) $(CHECK_CFLAGS) \
$(GST_OPTION_CFLAGS) $(GST_CFLAGS) -DGST_TEST_FILES_PATH="\"$(TEST_FILES_DIRECTORY)\"" \
+ -DGST_CHECK_TEST_ENVIRONMENT_BEACON="\"GST_PLUGIN_LOADING_WHITELIST\"" \
-UG_DISABLE_ASSERT -UG_DISABLE_CAST_CHECKS
LDADD = $(GST_OBJ_LIBS) $(GST_CHECK_LIBS) $(CHECK_LIBS)
@@ -472,6 +475,7 @@
elements_spectrum_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_API_VERSION) $(LDADD)
elements_alphacolor_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
+elements_alpha_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
elements_deinterlace_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
elements_deinterlace_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstvideo-$(GST_API_VERSION) $(LDADD)
diff --git a/tests/check/Makefile.in b/tests/check/Makefile.in
index edc6845..ea362b5 100644
--- a/tests/check/Makefile.in
+++ b/tests/check/Makefile.in
@@ -151,7 +151,8 @@
libparser_la_LINK = $(LIBTOOL) $(AM_V_lt) --tag=CC $(AM_LIBTOOLFLAGS) \
$(LIBTOOLFLAGS) --mode=link $(CCLD) $(libparser_la_CFLAGS) \
$(CFLAGS) $(AM_LDFLAGS) $(LDFLAGS) -o $@
-@USE_PLUGIN_ALPHA_TRUE@am__EXEEXT_1 = elements/alphacolor$(EXEEXT)
+@USE_PLUGIN_ALPHA_TRUE@am__EXEEXT_1 = elements/alphacolor$(EXEEXT) \
+@USE_PLUGIN_ALPHA_TRUE@ elements/alpha$(EXEEXT)
@USE_PLUGIN_AUDIOFX_TRUE@am__EXEEXT_2 = \
@USE_PLUGIN_AUDIOFX_TRUE@ elements/audioamplify$(EXEEXT) \
@USE_PLUGIN_AUDIOFX_TRUE@ elements/audiochebband$(EXEEXT) \
@@ -254,6 +255,14 @@
elements_ac3parse_SOURCES = elements/ac3parse.c
elements_ac3parse_OBJECTS = elements/ac3parse.$(OBJEXT)
elements_ac3parse_DEPENDENCIES = libparser.la $(am__DEPENDENCIES_2)
+elements_alpha_SOURCES = elements/alpha.c
+elements_alpha_OBJECTS = elements/elements_alpha-alpha.$(OBJEXT)
+elements_alpha_LDADD = $(LDADD)
+elements_alpha_DEPENDENCIES = $(am__DEPENDENCIES_1)
+elements_alpha_LINK = $(LIBTOOL) $(AM_V_lt) --tag=CC \
+ $(AM_LIBTOOLFLAGS) $(LIBTOOLFLAGS) --mode=link $(CCLD) \
+ $(elements_alpha_CFLAGS) $(CFLAGS) $(AM_LDFLAGS) $(LDFLAGS) -o \
+ $@
elements_alphacolor_SOURCES = elements/alphacolor.c
elements_alphacolor_OBJECTS = \
elements/elements_alphacolor-alphacolor.$(OBJEXT)
@@ -854,18 +863,19 @@
am__v_CCLD_0 = @echo " CCLD " $@;
am__v_CCLD_1 =
SOURCES = $(libparser_la_SOURCES) elements/aacparse.c \
- elements/ac3parse.c elements/alphacolor.c elements/amrparse.c \
- elements/apev2mux.c elements/aspectratiocrop.c \
- elements/audioamplify.c elements/audiochebband.c \
- elements/audiocheblimit.c elements/audiodynamic.c \
- elements/audioecho.c elements/audiofirfilter.c \
- elements/audioiirfilter.c elements/audioinvert.c \
- elements/audiopanorama.c elements/audiowsincband.c \
- elements/audiowsinclimit.c elements/autodetect.c \
- elements/avimux.c elements/avisubtitle.c elements/capssetter.c \
- elements/deinterlace.c elements/deinterleave.c elements/dtmf.c \
- elements/equalizer.c elements/flacparse.c elements/flvdemux.c \
- elements/flvmux.c elements/gdkpixbufsink.c elements/icydemux.c \
+ elements/ac3parse.c elements/alpha.c elements/alphacolor.c \
+ elements/amrparse.c elements/apev2mux.c \
+ elements/aspectratiocrop.c elements/audioamplify.c \
+ elements/audiochebband.c elements/audiocheblimit.c \
+ elements/audiodynamic.c elements/audioecho.c \
+ elements/audiofirfilter.c elements/audioiirfilter.c \
+ elements/audioinvert.c elements/audiopanorama.c \
+ elements/audiowsincband.c elements/audiowsinclimit.c \
+ elements/autodetect.c elements/avimux.c elements/avisubtitle.c \
+ elements/capssetter.c elements/deinterlace.c \
+ elements/deinterleave.c elements/dtmf.c elements/equalizer.c \
+ elements/flacparse.c elements/flvdemux.c elements/flvmux.c \
+ elements/gdkpixbufsink.c elements/icydemux.c \
elements/id3demux.c elements/id3v2mux.c elements/imagefreeze.c \
elements/interleave.c elements/jpegdec.c elements/jpegenc.c \
elements/level.c elements/matroskamux.c \
@@ -890,18 +900,19 @@
pipelines/tagschecking.c pipelines/wavenc.c \
pipelines/wavpack.c
DIST_SOURCES = $(libparser_la_SOURCES) elements/aacparse.c \
- elements/ac3parse.c elements/alphacolor.c elements/amrparse.c \
- elements/apev2mux.c elements/aspectratiocrop.c \
- elements/audioamplify.c elements/audiochebband.c \
- elements/audiocheblimit.c elements/audiodynamic.c \
- elements/audioecho.c elements/audiofirfilter.c \
- elements/audioiirfilter.c elements/audioinvert.c \
- elements/audiopanorama.c elements/audiowsincband.c \
- elements/audiowsinclimit.c elements/autodetect.c \
- elements/avimux.c elements/avisubtitle.c elements/capssetter.c \
- elements/deinterlace.c elements/deinterleave.c elements/dtmf.c \
- elements/equalizer.c elements/flacparse.c elements/flvdemux.c \
- elements/flvmux.c elements/gdkpixbufsink.c elements/icydemux.c \
+ elements/ac3parse.c elements/alpha.c elements/alphacolor.c \
+ elements/amrparse.c elements/apev2mux.c \
+ elements/aspectratiocrop.c elements/audioamplify.c \
+ elements/audiochebband.c elements/audiocheblimit.c \
+ elements/audiodynamic.c elements/audioecho.c \
+ elements/audiofirfilter.c elements/audioiirfilter.c \
+ elements/audioinvert.c elements/audiopanorama.c \
+ elements/audiowsincband.c elements/audiowsinclimit.c \
+ elements/autodetect.c elements/avimux.c elements/avisubtitle.c \
+ elements/capssetter.c elements/deinterlace.c \
+ elements/deinterleave.c elements/dtmf.c elements/equalizer.c \
+ elements/flacparse.c elements/flvdemux.c elements/flvmux.c \
+ elements/gdkpixbufsink.c elements/icydemux.c \
elements/id3demux.c elements/id3v2mux.c elements/imagefreeze.c \
elements/interleave.c elements/jpegdec.c elements/jpegenc.c \
elements/level.c elements/matroskamux.c \
@@ -1482,7 +1493,7 @@
REGISTRY_ENVIRONMENT = \
GST_REGISTRY_1_0=$(CHECK_REGISTRY)
-TESTS_ENVIRONMENT = \
+AM_TESTS_ENVIRONMENT = \
$(REGISTRY_ENVIRONMENT) \
GST_PLUGIN_SYSTEM_PATH_1_0= \
GST_PLUGIN_PATH_1_0=$(top_builddir)/gst:$(top_builddir)/ext:$(top_builddir)/sys:$(GSTPB_PLUGINS_DIR):$(GST_PLUGINS_DIR) \
@@ -1510,7 +1521,10 @@
# the core dumps of some machines have PIDs appended
CLEANFILES = core.* test-registry.*
@USE_PLUGIN_ALPHA_FALSE@check_alpha =
-@USE_PLUGIN_ALPHA_TRUE@check_alpha = elements/alphacolor
+@USE_PLUGIN_ALPHA_TRUE@check_alpha = \
+@USE_PLUGIN_ALPHA_TRUE@ elements/alphacolor \
+@USE_PLUGIN_ALPHA_TRUE@ elements/alpha
+
@USE_PLUGIN_AUDIOFX_FALSE@check_audiofx =
@USE_PLUGIN_AUDIOFX_TRUE@check_audiofx = \
@USE_PLUGIN_AUDIOFX_TRUE@ elements/audioamplify \
@@ -1662,6 +1676,7 @@
TESTS = $(check_PROGRAMS)
AM_CFLAGS = $(GST_OBJ_CFLAGS) $(GST_CHECK_CFLAGS) $(CHECK_CFLAGS) \
$(GST_OPTION_CFLAGS) $(GST_CFLAGS) -DGST_TEST_FILES_PATH="\"$(TEST_FILES_DIRECTORY)\"" \
+ -DGST_CHECK_TEST_ENVIRONMENT_BEACON="\"GST_PLUGIN_LOADING_WHITELIST\"" \
-UG_DISABLE_ASSERT -UG_DISABLE_CAST_CHECKS
LDADD = $(GST_OBJ_LIBS) $(GST_CHECK_LIBS) $(CHECK_LIBS)
@@ -1725,6 +1740,7 @@
elements_spectrum_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
elements_spectrum_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_API_VERSION) $(LDADD)
elements_alphacolor_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
+elements_alpha_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
elements_deinterlace_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
elements_deinterlace_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstvideo-$(GST_API_VERSION) $(LDADD)
elements_dtmf_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) \
@@ -1895,6 +1911,12 @@
elements/ac3parse$(EXEEXT): $(elements_ac3parse_OBJECTS) $(elements_ac3parse_DEPENDENCIES) $(EXTRA_elements_ac3parse_DEPENDENCIES) elements/$(am__dirstamp)
@rm -f elements/ac3parse$(EXEEXT)
$(AM_V_CCLD)$(LINK) $(elements_ac3parse_OBJECTS) $(elements_ac3parse_LDADD) $(LIBS)
+elements/elements_alpha-alpha.$(OBJEXT): elements/$(am__dirstamp) \
+ elements/$(DEPDIR)/$(am__dirstamp)
+
+elements/alpha$(EXEEXT): $(elements_alpha_OBJECTS) $(elements_alpha_DEPENDENCIES) $(EXTRA_elements_alpha_DEPENDENCIES) elements/$(am__dirstamp)
+ @rm -f elements/alpha$(EXEEXT)
+ $(AM_V_CCLD)$(elements_alpha_LINK) $(elements_alpha_OBJECTS) $(elements_alpha_LDADD) $(LIBS)
elements/elements_alphacolor-alphacolor.$(OBJEXT): \
elements/$(am__dirstamp) elements/$(DEPDIR)/$(am__dirstamp)
@@ -2426,6 +2448,7 @@
@AMDEP_TRUE@@am__include@ @am__quote@elements/$(DEPDIR)/avimux.Po@am__quote@
@AMDEP_TRUE@@am__include@ @am__quote@elements/$(DEPDIR)/avisubtitle.Po@am__quote@
@AMDEP_TRUE@@am__include@ @am__quote@elements/$(DEPDIR)/capssetter.Po@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@elements/$(DEPDIR)/elements_alpha-alpha.Po@am__quote@
@AMDEP_TRUE@@am__include@ @am__quote@elements/$(DEPDIR)/elements_alphacolor-alphacolor.Po@am__quote@
@AMDEP_TRUE@@am__include@ @am__quote@elements/$(DEPDIR)/elements_aspectratiocrop-aspectratiocrop.Po@am__quote@
@AMDEP_TRUE@@am__include@ @am__quote@elements/$(DEPDIR)/elements_audioamplify-audioamplify.Po@am__quote@
@@ -2534,6 +2557,20 @@
@AMDEP_TRUE@@am__fastdepCC_FALSE@ DEPDIR=$(DEPDIR) $(CCDEPMODE) $(depcomp) @AMDEPBACKSLASH@
@am__fastdepCC_FALSE@ $(AM_V_CC@am__nodep@)$(LIBTOOL) $(AM_V_lt) --tag=CC $(AM_LIBTOOLFLAGS) $(LIBTOOLFLAGS) --mode=compile $(CC) $(DEFS) $(DEFAULT_INCLUDES) $(INCLUDES) $(AM_CPPFLAGS) $(CPPFLAGS) $(libparser_la_CFLAGS) $(CFLAGS) -c -o elements/libparser_la-parser.lo `test -f 'elements/parser.c' || echo '$(srcdir)/'`elements/parser.c
+elements/elements_alpha-alpha.o: elements/alpha.c
+@am__fastdepCC_TRUE@ $(AM_V_CC)$(CC) $(DEFS) $(DEFAULT_INCLUDES) $(INCLUDES) $(AM_CPPFLAGS) $(CPPFLAGS) $(elements_alpha_CFLAGS) $(CFLAGS) -MT elements/elements_alpha-alpha.o -MD -MP -MF elements/$(DEPDIR)/elements_alpha-alpha.Tpo -c -o elements/elements_alpha-alpha.o `test -f 'elements/alpha.c' || echo '$(srcdir)/'`elements/alpha.c
+@am__fastdepCC_TRUE@ $(AM_V_at)$(am__mv) elements/$(DEPDIR)/elements_alpha-alpha.Tpo elements/$(DEPDIR)/elements_alpha-alpha.Po
+@AMDEP_TRUE@@am__fastdepCC_FALSE@ $(AM_V_CC)source='elements/alpha.c' object='elements/elements_alpha-alpha.o' libtool=no @AMDEPBACKSLASH@
+@AMDEP_TRUE@@am__fastdepCC_FALSE@ DEPDIR=$(DEPDIR) $(CCDEPMODE) $(depcomp) @AMDEPBACKSLASH@
+@am__fastdepCC_FALSE@ $(AM_V_CC@am__nodep@)$(CC) $(DEFS) $(DEFAULT_INCLUDES) $(INCLUDES) $(AM_CPPFLAGS) $(CPPFLAGS) $(elements_alpha_CFLAGS) $(CFLAGS) -c -o elements/elements_alpha-alpha.o `test -f 'elements/alpha.c' || echo '$(srcdir)/'`elements/alpha.c
+
+elements/elements_alpha-alpha.obj: elements/alpha.c
+@am__fastdepCC_TRUE@ $(AM_V_CC)$(CC) $(DEFS) $(DEFAULT_INCLUDES) $(INCLUDES) $(AM_CPPFLAGS) $(CPPFLAGS) $(elements_alpha_CFLAGS) $(CFLAGS) -MT elements/elements_alpha-alpha.obj -MD -MP -MF elements/$(DEPDIR)/elements_alpha-alpha.Tpo -c -o elements/elements_alpha-alpha.obj `if test -f 'elements/alpha.c'; then $(CYGPATH_W) 'elements/alpha.c'; else $(CYGPATH_W) '$(srcdir)/elements/alpha.c'; fi`
+@am__fastdepCC_TRUE@ $(AM_V_at)$(am__mv) elements/$(DEPDIR)/elements_alpha-alpha.Tpo elements/$(DEPDIR)/elements_alpha-alpha.Po
+@AMDEP_TRUE@@am__fastdepCC_FALSE@ $(AM_V_CC)source='elements/alpha.c' object='elements/elements_alpha-alpha.obj' libtool=no @AMDEPBACKSLASH@
+@AMDEP_TRUE@@am__fastdepCC_FALSE@ DEPDIR=$(DEPDIR) $(CCDEPMODE) $(depcomp) @AMDEPBACKSLASH@
+@am__fastdepCC_FALSE@ $(AM_V_CC@am__nodep@)$(CC) $(DEFS) $(DEFAULT_INCLUDES) $(INCLUDES) $(AM_CPPFLAGS) $(CPPFLAGS) $(elements_alpha_CFLAGS) $(CFLAGS) -c -o elements/elements_alpha-alpha.obj `if test -f 'elements/alpha.c'; then $(CYGPATH_W) 'elements/alpha.c'; else $(CYGPATH_W) '$(srcdir)/elements/alpha.c'; fi`
+
elements/elements_alphacolor-alphacolor.o: elements/alphacolor.c
@am__fastdepCC_TRUE@ $(AM_V_CC)$(CC) $(DEFS) $(DEFAULT_INCLUDES) $(INCLUDES) $(AM_CPPFLAGS) $(CPPFLAGS) $(elements_alphacolor_CFLAGS) $(CFLAGS) -MT elements/elements_alphacolor-alphacolor.o -MD -MP -MF elements/$(DEPDIR)/elements_alphacolor-alphacolor.Tpo -c -o elements/elements_alphacolor-alphacolor.o `test -f 'elements/alphacolor.c' || echo '$(srcdir)/'`elements/alphacolor.c
@am__fastdepCC_TRUE@ $(AM_V_at)$(am__mv) elements/$(DEPDIR)/elements_alphacolor-alphacolor.Tpo elements/$(DEPDIR)/elements_alphacolor-alphacolor.Po
@@ -3451,6 +3488,13 @@
--log-file $$b.log --trs-file $$b.trs \
$(am__common_driver_flags) $(AM_LOG_DRIVER_FLAGS) $(LOG_DRIVER_FLAGS) -- $(LOG_COMPILE) \
"$$tst" $(AM_TESTS_FD_REDIRECT)
+elements/alpha.log: elements/alpha$(EXEEXT)
+ @p='elements/alpha$(EXEEXT)'; \
+ b='elements/alpha'; \
+ $(am__check_pre) $(LOG_DRIVER) --test-name "$$f" \
+ --log-file $$b.log --trs-file $$b.trs \
+ $(am__common_driver_flags) $(AM_LOG_DRIVER_FLAGS) $(LOG_DRIVER_FLAGS) -- $(LOG_COMPILE) \
+ "$$tst" $(AM_TESTS_FD_REDIRECT)
elements/audioamplify.log: elements/audioamplify$(EXEEXT)
@p='elements/audioamplify$(EXEEXT)'; \
b='elements/audioamplify'; \
@@ -4208,38 +4252,38 @@
# run any given test by running make test.check
# if the test fails, run it again at at least debug level 2
%.check: %
- @$(TESTS_ENVIRONMENT) \
+ @$(AM_TESTS_ENVIRONMENT) \
CK_DEFAULT_TIMEOUT=20 \
$* || \
- $(TESTS_ENVIRONMENT) \
+ $(AM_TESTS_ENVIRONMENT) \
GST_DEBUG=$$GST_DEBUG,*:2 \
CK_DEFAULT_TIMEOUT=20 \
$*
# just like 'check', but don't run it again if it fails (useful for debugging)
%.check-norepeat: %
- @$(TESTS_ENVIRONMENT) \
+ @$(AM_TESTS_ENVIRONMENT) \
CK_DEFAULT_TIMEOUT=20 \
$*
# run any given test in a loop
%.torture: %
@for i in `seq 1 $(LOOPS)`; do \
- $(TESTS_ENVIRONMENT) \
+ $(AM_TESTS_ENVIRONMENT) \
CK_DEFAULT_TIMEOUT=20 \
$*; done
# run any given test in an infinite loop
%.forever: %
@while true; do \
- $(TESTS_ENVIRONMENT) \
+ $(AM_TESTS_ENVIRONMENT) \
CK_DEFAULT_TIMEOUT=20 \
$* || break; done
# valgrind any given test by running make test.valgrind
%.valgrind: %
@valgrind_log=$(subst /,-,$*-valgrind.log); \
- $(TESTS_ENVIRONMENT) \
+ $(AM_TESTS_ENVIRONMENT) \
CK_DEFAULT_TIMEOUT=360 \
G_SLICE=always-malloc \
$(LIBTOOL) --mode=execute \
@@ -4257,7 +4301,7 @@
# valgrind any given test and generate suppressions for it
%.valgrind.gen-suppressions: %
- @$(TESTS_ENVIRONMENT) \
+ @$(AM_TESTS_ENVIRONMENT) \
CK_DEFAULT_TIMEOUT=360 \
G_SLICE=always-malloc \
$(LIBTOOL) --mode=execute \
@@ -4287,7 +4331,7 @@
# gdb any given test by running make test.gdb
%.gdb: %
- @$(TESTS_ENVIRONMENT) \
+ @$(AM_TESTS_ENVIRONMENT) \
CK_FORK=no \
$(LIBTOOL) --mode=execute \
gdb $*
@@ -4391,7 +4435,7 @@
fi
inspect:
@echo "Inspecting features ..."
- @for e in `$(TESTS_ENVIRONMENT) $(GST_INSPECT) | head -n -2 \
+ @for e in `$(AM_TESTS_ENVIRONMENT) $(GST_INSPECT) | head -n -2 \
| cut -d: -f2`; \
do echo Inspecting $$e; \
$(GST_INSPECT) $$e > /dev/null 2>&1; done
diff --git a/tests/check/elements/alpha.c b/tests/check/elements/alpha.c
new file mode 100644
index 0000000..7c03bc1
--- /dev/null
+++ b/tests/check/elements/alpha.c
@@ -0,0 +1,265 @@
+/* GStreamer unit test for the alpha element
+ *
+ * Copyright (C) 2007 Ravi Kiran K N <ravi.kiran@samsung.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+#include <gst/video/video.h>
+
+
+GstPad *srcpad, *sinkpad;
+
+static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS (GST_VIDEO_CAPS_MAKE ("AYUV"))
+ );
+static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS (GST_VIDEO_CAPS_MAKE ("{ AYUV, "
+ "ARGB, BGRA, ABGR, RGBA, Y444, xRGB, BGRx, xBGR, "
+ "RGBx, RGB, BGR, Y42B, YUY2, YVYU, UYVY, I420, YV12, Y41B } "))
+ );
+
+
+typedef enum
+{
+ FILL_GREEN,
+ FILL_BLUE
+}
+FillColor;
+
+static GstElement *
+setup_alpha (void)
+{
+ GstElement *alpha;
+
+ alpha = gst_check_setup_element ("alpha");
+ srcpad = gst_check_setup_src_pad (alpha, &srctemplate);
+ sinkpad = gst_check_setup_sink_pad (alpha, &sinktemplate);
+
+ gst_pad_set_active (srcpad, TRUE);
+ gst_pad_set_active (sinkpad, TRUE);
+
+ return alpha;
+}
+
+static void
+cleanup_alpha (GstElement * alpha)
+{
+ gst_pad_set_active (srcpad, FALSE);
+ gst_pad_set_active (sinkpad, FALSE);
+ gst_check_teardown_src_pad (alpha);
+ gst_check_teardown_sink_pad (alpha);
+ gst_check_teardown_element (alpha);
+}
+
+#define WIDTH 3
+#define HEIGHT 4
+
+static GstCaps *
+create_caps_rgba32 (void)
+{
+ GstCaps *caps;
+
+ caps = gst_caps_new_simple ("video/x-raw",
+ "width", G_TYPE_INT, WIDTH,
+ "height", G_TYPE_INT, HEIGHT,
+ "framerate", GST_TYPE_FRACTION, 0, 1,
+ "format", G_TYPE_STRING, "RGBA", NULL);
+
+ return caps;
+}
+
+static GstBuffer *
+create_buffer_rgba32 (FillColor color)
+{
+ guint8 rgba32_img[HEIGHT * WIDTH * 4];
+ guint32 *rgba32 = (guint32 *) rgba32_img;
+
+ GstBuffer *buf;
+ GstMapInfo map;
+ guint32 rgba_col;
+ int i;
+
+ if (color == FILL_GREEN)
+ rgba_col = 0xff00ff00; /* GREEN */
+ else
+ rgba_col = 0xffff0000; /* BLUE */
+
+ for (i = 0; i < HEIGHT * WIDTH; i++)
+ rgba32[i] = rgba_col;
+
+ buf = gst_buffer_new_and_alloc (HEIGHT * WIDTH * 4);
+ gst_buffer_map (buf, &map, GST_MAP_READWRITE);
+ fail_unless_equals_int (map.size, sizeof (rgba32_img));
+ memcpy (map.data, rgba32_img, sizeof (rgba32_img));
+
+ gst_buffer_unmap (buf, &map);
+
+ return buf;
+}
+
+
+GST_START_TEST (test_chromakeying)
+{
+ GstElement *alpha;
+ GstBuffer *inbuffer;
+ GstBuffer *outbuffer;
+ GstCaps *incaps;
+ guint8 *ayuv;
+ guint outlength;
+ GstMapInfo map;
+ int i;
+
+ incaps = create_caps_rgba32 ();
+
+ alpha = setup_alpha ();
+
+ g_object_set (alpha, "method", 1, NULL); /* Chroma-keying GREEN */
+
+ fail_unless_equals_int (gst_element_set_state (alpha, GST_STATE_PLAYING),
+ GST_STATE_CHANGE_SUCCESS);
+
+ gst_check_setup_events (srcpad, alpha, incaps, GST_FORMAT_TIME);
+
+ inbuffer = create_buffer_rgba32 (FILL_GREEN);
+ GST_DEBUG ("Created buffer of %" G_GSIZE_FORMAT " bytes",
+ gst_buffer_get_size (inbuffer));
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ fail_unless_equals_int (gst_pad_push (srcpad, inbuffer), GST_FLOW_OK);
+
+ fail_unless (g_list_length (buffers) == 1);
+ outbuffer = (GstBuffer *) buffers->data;
+ fail_if (outbuffer == NULL);
+ fail_unless (GST_IS_BUFFER (outbuffer));
+
+ ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1);
+ outlength = WIDTH * HEIGHT * 4; /* output is AYUV */
+ gst_buffer_map (outbuffer, &map, GST_MAP_READ);
+ fail_unless_equals_int (map.size, outlength);
+
+ ayuv = map.data;
+
+ /* check chroma keying GREEN */
+ for (i = 0; i < HEIGHT * WIDTH; i += 4)
+ fail_unless_equals_int (ayuv[i], 0x00);
+
+ gst_buffer_unmap (outbuffer, &map);
+
+ buffers = g_list_remove (buffers, outbuffer);
+ gst_buffer_unref (outbuffer);
+
+ fail_unless_equals_int (gst_element_set_state (alpha, GST_STATE_NULL),
+ GST_STATE_CHANGE_SUCCESS);
+
+ /* cleanup */
+ cleanup_alpha (alpha);
+ ASSERT_CAPS_REFCOUNT (incaps, "incaps", 1);
+ gst_caps_unref (incaps);
+
+}
+
+GST_END_TEST;
+
+
+
+GST_START_TEST (test_alpha)
+{
+ GstElement *alpha;
+ GstBuffer *inbuffer;
+ GstBuffer *outbuffer;
+ GstCaps *incaps;
+ guint8 *ayuv;
+ guint outlength;
+ GstMapInfo map;
+ int i;
+
+ incaps = create_caps_rgba32 ();
+
+ alpha = setup_alpha ();
+
+ g_object_set (alpha, "alpha", 0.5, NULL); /* Alpha value 0.5 */
+
+ fail_unless_equals_int (gst_element_set_state (alpha, GST_STATE_PLAYING),
+ GST_STATE_CHANGE_SUCCESS);
+
+ gst_check_setup_events (srcpad, alpha, incaps, GST_FORMAT_TIME);
+
+ inbuffer = create_buffer_rgba32 (FILL_BLUE);
+ GST_DEBUG ("Created buffer of %" G_GSIZE_FORMAT " bytes",
+ gst_buffer_get_size (inbuffer));
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ /* pushing gives away reference */
+ GST_DEBUG ("push it");
+ fail_unless_equals_int (gst_pad_push (srcpad, inbuffer), GST_FLOW_OK);
+ GST_DEBUG ("pushed it");
+
+ /* ... and puts a new buffer on the global list */
+ fail_unless (g_list_length (buffers) == 1);
+ outbuffer = (GstBuffer *) buffers->data;
+ fail_if (outbuffer == NULL);
+ fail_unless (GST_IS_BUFFER (outbuffer));
+
+ ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1);
+ outlength = WIDTH * HEIGHT * 4; /* output is AYUV */
+ gst_buffer_map (outbuffer, &map, GST_MAP_READ);
+ fail_unless_equals_int (map.size, outlength);
+
+ ayuv = map.data;
+
+ for (i = 0; i < HEIGHT * WIDTH; i += 4)
+ fail_unless_equals_int (ayuv[i], 0x7F);
+
+ gst_buffer_unmap (outbuffer, &map);
+
+ buffers = g_list_remove (buffers, outbuffer);
+ gst_buffer_unref (outbuffer);
+
+ fail_unless_equals_int (gst_element_set_state (alpha, GST_STATE_NULL),
+ GST_STATE_CHANGE_SUCCESS);
+
+ /* cleanup */
+ GST_DEBUG ("cleanup alpha");
+ cleanup_alpha (alpha);
+ GST_DEBUG ("cleanup, unref incaps");
+ ASSERT_CAPS_REFCOUNT (incaps, "incaps", 1);
+ gst_caps_unref (incaps);
+
+}
+
+GST_END_TEST;
+
+
+static Suite *
+alpha_suite (void)
+{
+ Suite *s = suite_create ("alpha");
+ TCase *tc_chain = tcase_create ("general");
+
+ suite_add_tcase (s, tc_chain);
+ tcase_add_test (tc_chain, test_alpha);
+ tcase_add_test (tc_chain, test_chromakeying);
+
+ return s;
+}
+
+GST_CHECK_MAIN (alpha);
diff --git a/tests/check/elements/multifile.c b/tests/check/elements/multifile.c
index 20770bb..68717dc 100644
--- a/tests/check/elements/multifile.c
+++ b/tests/check/elements/multifile.c
@@ -30,14 +30,64 @@
#include <stdlib.h>
#include <unistd.h>
+static GList *mfs_messages = NULL;
+
+static void
+mfs_check_next_message (const gchar * filename)
+{
+ GstMessage *msg;
+ const gchar *msg_filename;
+ const GstStructure *structure;
+
+ fail_unless (mfs_messages != NULL);
+
+ msg = mfs_messages->data;
+ mfs_messages = g_list_delete_link (mfs_messages, mfs_messages);
+
+ structure = gst_message_get_structure (msg);
+
+ msg_filename = gst_structure_get_string (structure, "filename");
+
+ fail_unless (strcmp (filename, msg_filename) == 0);
+
+ gst_message_unref (msg);
+}
+
static void
run_pipeline (GstElement * pipeline)
{
+ GstMessage *msg;
+ GstBus *bus;
+
+ bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
+
gst_element_set_state (pipeline, GST_STATE_PAUSED);
gst_element_get_state (pipeline, NULL, NULL, -1);
gst_element_set_state (pipeline, GST_STATE_PLAYING);
- /* FIXME too lazy */
- g_usleep (1000000);
+
+ while (1) {
+ msg =
+ gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE,
+ GST_MESSAGE_EOS | GST_MESSAGE_ERROR | GST_MESSAGE_ELEMENT);
+
+ fail_unless (msg != NULL);
+ if (msg) {
+ if (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_ELEMENT) {
+ if (gst_message_has_name (msg, "GstMultiFileSink"))
+ mfs_messages = g_list_append (mfs_messages, msg);
+ else
+ gst_message_unref (msg);
+
+ continue;
+ }
+
+ fail_unless (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_EOS);
+ gst_message_unref (msg);
+ }
+ break;
+ }
+
+ gst_object_unref (bus);
gst_element_set_state (pipeline, GST_STATE_NULL);
}
@@ -64,7 +114,8 @@
mfs = gst_bin_get_by_name (GST_BIN (pipeline), "mfs");
fail_if (mfs == NULL);
mfs_pattern = g_build_filename (my_tmpdir, "%05d", NULL);
- g_object_set (G_OBJECT (mfs), "location", mfs_pattern, NULL);
+ g_object_set (G_OBJECT (mfs), "location", mfs_pattern, "post-messages", TRUE,
+ NULL);
g_object_unref (mfs);
run_pipeline (pipeline);
gst_object_unref (pipeline);
@@ -74,10 +125,14 @@
s = g_strdup_printf (mfs_pattern, i);
fail_if (g_remove (s) != 0);
+
+ mfs_check_next_message (s);
+
g_free (s);
}
fail_if (g_remove (my_tmpdir) != 0);
+ fail_unless (mfs_messages == NULL);
g_free (mfs_pattern);
g_free (my_tmpdir);
}
@@ -145,6 +200,7 @@
GstBuffer *buf;
GstPad *sink;
GstSegment segment;
+ GstBus *bus;
tmpdir = g_get_tmp_dir ();
template = g_build_filename (tmpdir, "multifile-test-XXXXXX", NULL);
@@ -154,7 +210,10 @@
mfs = gst_element_factory_make ("multifilesink", NULL);
fail_if (mfs == NULL);
mfs_pattern = g_build_filename (my_tmpdir, "%05d", NULL);
- g_object_set (G_OBJECT (mfs), "location", mfs_pattern, "next-file", 3, NULL);
+ g_object_set (G_OBJECT (mfs), "location", mfs_pattern, "next-file", 3,
+ "post-messages", TRUE, NULL);
+ bus = gst_bus_new ();
+ gst_element_set_bus (mfs, bus);
fail_if (gst_element_set_state (mfs,
GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
@@ -179,18 +238,32 @@
gst_buffer_fill (buf, 0, "baz", 4);
fail_if (gst_pad_chain (sink, buf) != GST_FLOW_OK);
+ gst_pad_send_event (sink, gst_event_new_eos ());
+
fail_if (gst_element_set_state (mfs,
GST_STATE_NULL) == GST_STATE_CHANGE_FAILURE);
+ gst_element_set_bus (mfs, NULL);
for (i = 0; i < 2; i++) {
char *s;
+ GstMessage *msg;
s = g_strdup_printf (mfs_pattern, i);
fail_if (g_remove (s) != 0);
+
+ msg = gst_bus_pop_filtered (bus, GST_MESSAGE_ELEMENT);
+ fail_unless (msg != NULL);
+ fail_unless (gst_message_has_name (msg, "GstMultiFileSink"));
+ fail_unless (strcmp (s,
+ gst_structure_get_string (gst_message_get_structure (msg),
+ "filename")) == 0);
+
+ gst_message_unref (msg);
g_free (s);
}
fail_if (g_remove (my_tmpdir) != 0);
+ gst_object_unref (bus);
g_free (mfs_pattern);
g_free (my_tmpdir);
gst_object_unref (sink);
diff --git a/tests/check/elements/qtmux.c b/tests/check/elements/qtmux.c
index 105b1e8..39c7c5e 100644
--- a/tests/check/elements/qtmux.c
+++ b/tests/check/elements/qtmux.c
@@ -38,6 +38,8 @@
* get_peer, and then remove references in every test function */
static GstPad *mysrcpad, *mysinkpad;
+#define VIDEO_RAW_CAPS_STRING "video/x-raw"
+
#define AUDIO_CAPS_STRING "audio/mpeg, " \
"mpegversion = (int) 1, " \
"layer = (int) 3, " \
@@ -78,6 +80,7 @@
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("video/quicktime"));
+
static GstStaticPadTemplate srcvideotemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
@@ -89,6 +92,12 @@
GST_PAD_ALWAYS,
GST_STATIC_CAPS (VIDEO_CAPS_H264_STRING));
+static GstStaticPadTemplate srcvideorawtemplate =
+GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS (VIDEO_RAW_CAPS_STRING));
+
static GstStaticPadTemplate srcaudiotemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
@@ -153,8 +162,24 @@
gst_object_unref (srcpad);
}
+gboolean downstream_is_seekable;
+static gboolean
+qtmux_sinkpad_query (GstPad * pad, GstObject * parent, GstQuery * query)
+{
+ gboolean ret = FALSE;
+
+ if (GST_QUERY_TYPE (query) == GST_QUERY_SEEKING) {
+ gst_query_set_seeking (query, GST_FORMAT_BYTES, downstream_is_seekable, 0,
+ -1);
+ ret = TRUE;
+ }
+
+ return ret;
+}
+
static GstElement *
-setup_qtmux (GstStaticPadTemplate * srctemplate, const gchar * sinkname)
+setup_qtmux (GstStaticPadTemplate * srctemplate, const gchar * sinkname,
+ gboolean seekable)
{
GstElement *qtmux;
@@ -162,6 +187,10 @@
qtmux = gst_check_setup_element ("qtmux");
mysrcpad = setup_src_pad (qtmux, srctemplate, sinkname);
mysinkpad = gst_check_setup_sink_pad (qtmux, &sinktemplate);
+
+ downstream_is_seekable = seekable;
+ gst_pad_set_query_function (mysinkpad, qtmux_sinkpad_query);
+
gst_pad_set_active (mysrcpad, TRUE);
gst_pad_set_active (mysinkpad, TRUE);
@@ -195,7 +224,7 @@
guint8 data2[4] = "moov";
GstSegment segment;
- qtmux = setup_qtmux (srctemplate, sinkname);
+ qtmux = setup_qtmux (srctemplate, sinkname, TRUE);
g_object_set (qtmux, "dts-method", dts_method, NULL);
fail_unless (gst_element_set_state (qtmux,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
@@ -285,7 +314,7 @@
guint8 data4[4] = "mfra";
GstSegment segment;
- qtmux = setup_qtmux (srctemplate, sinkname);
+ qtmux = setup_qtmux (srctemplate, sinkname, !streamable);
g_object_set (qtmux, "dts-method", dts_method, NULL);
g_object_set (qtmux, "fragment-duration", 2000, NULL);
g_object_set (qtmux, "streamable", streamable, NULL);
@@ -515,7 +544,7 @@
GST_START_TEST (test_reuse)
{
- GstElement *qtmux = setup_qtmux (&srcvideotemplate, "video_%u");
+ GstElement *qtmux = setup_qtmux (&srcvideotemplate, "video_%u", TRUE);
GstBuffer *inbuffer;
GstCaps *caps;
GstSegment segment;
@@ -888,6 +917,468 @@
GST_END_TEST;
+struct TestInputData
+{
+ GstPad *srcpad;
+ GstSegment segment;
+ GList *input;
+ GThread *thread;
+
+ /* When comparing ts, the input will be subtracted from this */
+ gint64 ts_offset;
+
+ GstPad *sinkpad;
+
+ GList *output_iter;
+};
+
+static void
+test_input_data_init (struct TestInputData *data)
+{
+ data->ts_offset = 0;
+ data->srcpad = NULL;
+ data->sinkpad = NULL;
+ data->input = NULL;
+ data->thread = NULL;
+}
+
+static void
+test_input_data_clean (struct TestInputData *data)
+{
+ g_list_free_full (data->input, (GDestroyNotify) gst_mini_object_unref);
+
+ if (data->sinkpad) {
+ gst_pad_set_active (data->sinkpad, FALSE);
+ gst_object_unref (data->sinkpad);
+ }
+
+ gst_pad_set_active (data->srcpad, FALSE);
+ teardown_src_pad (data->srcpad);
+}
+
+static gpointer
+test_input_push_data (gpointer user_data)
+{
+ struct TestInputData *data = user_data;
+ GList *iter;
+ GstFlowReturn flow;
+
+ for (iter = data->input; iter; iter = g_list_next (iter)) {
+ if (GST_IS_BUFFER (iter->data)) {
+ GST_INFO ("Pushing buffer %" GST_PTR_FORMAT " on pad: %s:%s", iter->data,
+ GST_DEBUG_PAD_NAME (data->srcpad));
+ flow =
+ gst_pad_push (data->srcpad,
+ gst_buffer_ref ((GstBuffer *) iter->data));
+ fail_unless (flow == GST_FLOW_OK);
+ } else {
+ GST_INFO_OBJECT (data->srcpad, "Pushing event: %"
+ GST_PTR_FORMAT, iter->data);
+ fail_unless (gst_pad_push_event (data->srcpad,
+ gst_event_ref ((GstEvent *) iter->data)) == TRUE);
+ }
+ }
+ return NULL;
+}
+
+static GstBuffer *
+create_buffer (GstClockTime pts, GstClockTime dts, GstClockTime duration,
+ guint bytes)
+{
+ GstBuffer *buf;
+ guint8 *data;
+
+ data = g_malloc0 (bytes);
+ buf = gst_buffer_new_wrapped (data, bytes);
+ GST_BUFFER_PTS (buf) = pts;
+ GST_BUFFER_DTS (buf) = dts;
+ GST_BUFFER_DURATION (buf) = duration;
+ return buf;
+}
+
+static GstFlowReturn
+_test_sink_pad_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
+{
+ struct TestInputData *test_data = g_object_get_qdata (G_OBJECT (pad),
+ g_quark_from_static_string ("test-mux-pad"));
+ GstBuffer *expected_buffer;
+
+ fail_unless (test_data->output_iter);
+ fail_unless (GST_IS_BUFFER (test_data->output_iter->data));
+ expected_buffer = test_data->output_iter->data;
+
+ fail_unless (GST_BUFFER_PTS (buffer) ==
+ (GST_BUFFER_PTS_IS_VALID (expected_buffer) ?
+ GST_BUFFER_PTS (expected_buffer) -
+ test_data->ts_offset : GST_BUFFER_PTS (expected_buffer)));
+ fail_unless (GST_BUFFER_DTS (buffer) ==
+ (GST_BUFFER_DTS_IS_VALID (expected_buffer) ?
+ GST_BUFFER_DTS (expected_buffer) -
+ test_data->ts_offset : GST_BUFFER_DTS (buffer)));
+ fail_unless (GST_BUFFER_DURATION (buffer) ==
+ GST_BUFFER_DURATION (expected_buffer));
+
+ test_data->output_iter = g_list_next (test_data->output_iter);
+
+ gst_buffer_unref (buffer);
+ return GST_FLOW_OK;
+}
+
+static void
+compare_event (GstEvent * event, GstEvent * expected)
+{
+ fail_unless (GST_EVENT_TYPE (event) == GST_EVENT_TYPE (expected));
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_CAPS:{
+ GstCaps *caps, *expected_caps;
+
+ gst_event_parse_caps (event, &caps);
+ gst_event_parse_caps (expected, &expected_caps);
+ fail_unless (gst_caps_can_intersect (caps, expected_caps));
+ }
+ break;
+ default:
+ break;
+ }
+}
+
+static gboolean
+_test_sink_pad_event (GstPad * pad, GstObject * parent, GstEvent * event)
+{
+ struct TestInputData *test_data = g_object_get_qdata (G_OBJECT (pad),
+ g_quark_from_static_string ("test-mux-pad"));
+
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_STREAM_START:
+ case GST_EVENT_SEGMENT:
+ case GST_EVENT_CAPS:
+ case GST_EVENT_EOS:
+ fail_unless (test_data->output_iter);
+ fail_unless (GST_IS_EVENT (test_data->output_iter->data));
+ compare_event (event, test_data->output_iter->data);
+ test_data->output_iter = g_list_next (test_data->output_iter);
+ break;
+ case GST_EVENT_TAG:
+ /* ignore this event */
+ break;
+ default:
+ GST_ERROR_OBJECT (pad, "Unexpected event: %" GST_PTR_FORMAT, event);
+ fail ("Unexpected event received %s", GST_EVENT_TYPE_NAME (event));
+ break;
+ }
+
+ gst_event_unref (event);
+ return TRUE;
+}
+
+static void
+_test_pad_added_cb (GstElement * element, GstPad * pad, gpointer udata)
+{
+ GstCaps *caps;
+ struct TestInputData **inputs = udata;
+ gint i = -1;
+ const gchar *name;
+ const gchar *strname;
+
+ caps = gst_pad_get_current_caps (pad);
+ strname = gst_structure_get_name (gst_caps_get_structure (caps, 0));
+ if (g_str_has_prefix (strname, "video/")) {
+ i = 0; /* video is 0, audio is 1 */
+ name = "videosink";
+ } else {
+ i = 1;
+ name = "audiosink";
+ }
+ gst_caps_unref (caps);
+
+ fail_unless (i != -1);
+ fail_unless (inputs[i]->sinkpad == NULL);
+ inputs[i]->sinkpad = gst_pad_new (name, GST_PAD_SINK);
+ inputs[i]->output_iter = inputs[i]->input;
+ g_object_set_qdata (G_OBJECT (inputs[i]->sinkpad),
+ g_quark_from_static_string ("test-mux-pad"), inputs[i]);
+ gst_pad_set_chain_function (inputs[i]->sinkpad, _test_sink_pad_chain);
+ gst_pad_set_event_function (inputs[i]->sinkpad, _test_sink_pad_event);
+ gst_pad_set_active (inputs[i]->sinkpad, TRUE);
+ fail_unless (gst_pad_link (pad, inputs[i]->sinkpad) == GST_PAD_LINK_OK);
+}
+
+static void
+check_output (const gchar * location, struct TestInputData *input1,
+ struct TestInputData *input2)
+{
+ GstElement *filesrc;
+ GstElement *demux;
+ struct TestInputData *inputs[2] = { input1, input2 };
+
+ filesrc = gst_element_factory_make ("filesrc", NULL);
+ demux = gst_element_factory_make ("qtdemux", NULL);
+
+ fail_unless (gst_element_link (filesrc, demux));
+
+ g_object_set (filesrc, "location", location, NULL);
+ g_signal_connect (demux, "pad-added", (GCallback) _test_pad_added_cb, inputs);
+
+ fail_unless (gst_element_set_state (demux,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
+ fail_unless (gst_element_set_state (filesrc,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
+
+ /* FIXME use a main loop */
+ g_usleep (2 * G_USEC_PER_SEC);
+
+ fail_unless (gst_element_set_state (demux,
+ GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS);
+ fail_unless (gst_element_set_state (filesrc,
+ GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS);
+ gst_object_unref (filesrc);
+ gst_object_unref (demux);
+}
+
+/* Muxes a file with qtmux using the inputs provided and
+ * then verifies that the generated file corresponds to the
+ * data in the inputs */
+static void
+run_muxing_test (struct TestInputData *input1, struct TestInputData *input2)
+{
+ gchar *location;
+ GstElement *qtmux;
+ GstElement *filesink;
+
+ location = g_strdup_printf ("%s/%s-%d", g_get_tmp_dir (), "qtmuxtest",
+ g_random_int ());
+ qtmux = gst_check_setup_element ("qtmux");
+ filesink = gst_element_factory_make ("filesink", NULL);
+ g_object_set (filesink, "location", location, NULL);
+ gst_element_link (qtmux, filesink);
+
+ input1->srcpad = setup_src_pad (qtmux, &srcvideorawtemplate, "video_%u");
+ fail_unless (input1->srcpad != NULL);
+ gst_pad_set_active (input1->srcpad, TRUE);
+
+ input2->srcpad = setup_src_pad (qtmux, &srcaudioaactemplate, "audio_%u");
+ fail_unless (input2->srcpad != NULL);
+ gst_pad_set_active (input2->srcpad, TRUE);
+
+ fail_unless (gst_element_set_state (filesink,
+ GST_STATE_PLAYING) != GST_STATE_CHANGE_FAILURE,
+ "could not set filesink to playing");
+ fail_unless (gst_element_set_state (qtmux,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ input1->thread =
+ g_thread_new ("test-push-data-1", test_input_push_data, input1);
+ input2->thread =
+ g_thread_new ("test-push-data-2", test_input_push_data, input2);
+
+ /* FIXME set a mainloop and wait for EOS */
+
+ g_thread_join (input1->thread);
+ g_thread_join (input2->thread);
+ input1->thread = NULL;
+ input2->thread = NULL;
+
+ gst_element_set_state (qtmux, GST_STATE_NULL);
+ gst_element_set_state (filesink, GST_STATE_NULL);
+
+ check_output (location, input1, input2);
+
+ gst_object_unref (filesink);
+ test_input_data_clean (input1);
+ test_input_data_clean (input2);
+ gst_check_teardown_element (qtmux);
+
+ /* delete file */
+ g_unlink (location);
+ g_free (location);
+}
+
+GST_START_TEST (test_muxing)
+{
+ struct TestInputData input1, input2;
+ GstCaps *caps;
+
+ test_input_data_init (&input1);
+ test_input_data_init (&input2);
+
+ /* Create the inputs, after calling the run below, all this data is
+ * transfered to it and we have no need to clean up */
+ input1.input = NULL;
+ input1.input =
+ g_list_append (input1.input, gst_event_new_stream_start ("test-1"));
+ caps = gst_caps_from_string
+ ("video/x-raw, width=(int)800, height=(int)600, "
+ "framerate=(fraction)1/1, format=(string)RGB");
+ input1.input = g_list_append (input1.input, gst_event_new_caps (caps));
+ gst_caps_unref (caps);
+ gst_segment_init (&input1.segment, GST_FORMAT_TIME);
+ input1.input =
+ g_list_append (input1.input, gst_event_new_segment (&input1.segment));
+ input1.input =
+ g_list_append (input1.input, create_buffer (0, GST_CLOCK_TIME_NONE,
+ GST_SECOND, 800 * 600 * 3));
+ input1.input =
+ g_list_append (input1.input, create_buffer (1 * GST_SECOND,
+ GST_CLOCK_TIME_NONE, GST_SECOND, 800 * 600 * 3));
+ input1.input =
+ g_list_append (input1.input, create_buffer (2 * GST_SECOND,
+ GST_CLOCK_TIME_NONE, GST_SECOND, 800 * 600 * 3));
+ input1.input = g_list_append (input1.input, gst_event_new_eos ());
+
+ input2.input = NULL;
+ input2.input =
+ g_list_append (input2.input, gst_event_new_stream_start ("test-2"));
+ caps = gst_caps_from_string
+ ("audio/mpeg, rate=(int)44100, channels=(int)1, mpegversion=(int)4, "
+ "stream-format=(string)raw, framed=(boolean)true");
+ input2.input = g_list_append (input2.input, gst_event_new_caps (caps));
+ gst_caps_unref (caps);
+ gst_segment_init (&input2.segment, GST_FORMAT_TIME);
+ input2.input =
+ g_list_append (input2.input, gst_event_new_segment (&input2.segment));
+ input2.input =
+ g_list_append (input2.input, create_buffer (0, 0, GST_SECOND, 4096));
+ input2.input =
+ g_list_append (input2.input, create_buffer (1 * GST_SECOND,
+ 1 * GST_SECOND, GST_SECOND, 4096));
+ input2.input =
+ g_list_append (input2.input, create_buffer (2 * GST_SECOND,
+ 2 * GST_SECOND, GST_SECOND, 4096));
+ input2.input = g_list_append (input2.input, gst_event_new_eos ());
+
+ run_muxing_test (&input1, &input2);
+}
+
+GST_END_TEST;
+
+
+GST_START_TEST (test_muxing_non_zero_segment)
+{
+ struct TestInputData input1, input2;
+ GstCaps *caps;
+
+ test_input_data_init (&input1);
+ test_input_data_init (&input2);
+
+ /* Create the inputs, after calling the run below, all this data is
+ * transfered to it and we have no need to clean up */
+ input1.input = NULL;
+ input1.input =
+ g_list_append (input1.input, gst_event_new_stream_start ("test-1"));
+ caps = gst_caps_from_string
+ ("video/x-raw, width=(int)800, height=(int)600, "
+ "framerate=(fraction)1/1, format=(string)RGB");
+ input1.input = g_list_append (input1.input, gst_event_new_caps (caps));
+ gst_caps_unref (caps);
+ gst_segment_init (&input1.segment, GST_FORMAT_TIME);
+ input1.segment.start = 10 * GST_SECOND;
+ input1.input =
+ g_list_append (input1.input, gst_event_new_segment (&input1.segment));
+ input1.input =
+ g_list_append (input1.input, create_buffer (10 * GST_SECOND,
+ GST_CLOCK_TIME_NONE, GST_SECOND, 800 * 600 * 3));
+ input1.input =
+ g_list_append (input1.input, create_buffer (11 * GST_SECOND,
+ GST_CLOCK_TIME_NONE, GST_SECOND, 800 * 600 * 3));
+ input1.input =
+ g_list_append (input1.input, create_buffer (12 * GST_SECOND,
+ GST_CLOCK_TIME_NONE, GST_SECOND, 800 * 600 * 3));
+ input1.input = g_list_append (input1.input, gst_event_new_eos ());
+ input1.ts_offset = GST_SECOND * 10;
+
+ input2.input = NULL;
+ input2.input =
+ g_list_append (input2.input, gst_event_new_stream_start ("test-2"));
+ caps = gst_caps_from_string
+ ("audio/mpeg, rate=(int)44100, channels=(int)1, mpegversion=(int)4, "
+ "stream-format=(string)raw, framed=(boolean)true");
+ input2.input = g_list_append (input2.input, gst_event_new_caps (caps));
+ gst_caps_unref (caps);
+ gst_segment_init (&input2.segment, GST_FORMAT_TIME);
+ input2.segment.start = 10 * GST_SECOND;
+ input2.input =
+ g_list_append (input2.input, gst_event_new_segment (&input2.segment));
+ input2.input =
+ g_list_append (input2.input, create_buffer (10 * GST_SECOND,
+ 10 * GST_SECOND, GST_SECOND, 4096));
+ input2.input =
+ g_list_append (input2.input, create_buffer (11 * GST_SECOND,
+ 11 * GST_SECOND, GST_SECOND, 4096));
+ input2.input =
+ g_list_append (input2.input, create_buffer (12 * GST_SECOND,
+ 12 * GST_SECOND, GST_SECOND, 4096));
+ input2.input = g_list_append (input2.input, gst_event_new_eos ());
+ input2.ts_offset = GST_SECOND * 10;
+
+ run_muxing_test (&input1, &input2);
+}
+
+GST_END_TEST;
+
+
+GST_START_TEST (test_muxing_non_zero_segment_different)
+{
+ struct TestInputData input1, input2;
+ GstCaps *caps;
+
+ test_input_data_init (&input1);
+ test_input_data_init (&input2);
+
+ /* Create the inputs, after calling the run below, all this data is
+ * transfered to it and we have no need to clean up */
+ input1.input = NULL;
+ input1.input =
+ g_list_append (input1.input, gst_event_new_stream_start ("test-1"));
+ caps = gst_caps_from_string
+ ("video/x-raw, width=(int)800, height=(int)600, "
+ "framerate=(fraction)1/1, format=(string)RGB");
+ input1.input = g_list_append (input1.input, gst_event_new_caps (caps));
+ gst_caps_unref (caps);
+ gst_segment_init (&input1.segment, GST_FORMAT_TIME);
+ input1.segment.start = 5 * GST_SECOND;
+ input1.input =
+ g_list_append (input1.input, gst_event_new_segment (&input1.segment));
+ input1.input =
+ g_list_append (input1.input, create_buffer (5 * GST_SECOND,
+ GST_CLOCK_TIME_NONE, GST_SECOND, 800 * 600 * 3));
+ input1.input =
+ g_list_append (input1.input, create_buffer (6 * GST_SECOND,
+ GST_CLOCK_TIME_NONE, GST_SECOND, 800 * 600 * 3));
+ input1.input =
+ g_list_append (input1.input, create_buffer (7 * GST_SECOND,
+ GST_CLOCK_TIME_NONE, GST_SECOND, 800 * 600 * 3));
+ input1.input = g_list_append (input1.input, gst_event_new_eos ());
+ input1.ts_offset = GST_SECOND * 5;
+
+ input2.input = NULL;
+ input2.input =
+ g_list_append (input2.input, gst_event_new_stream_start ("test-2"));
+ caps = gst_caps_from_string
+ ("audio/mpeg, rate=(int)44100, channels=(int)1, mpegversion=(int)4, "
+ "stream-format=(string)raw, framed=(boolean)true");
+ input2.input = g_list_append (input2.input, gst_event_new_caps (caps));
+ gst_caps_unref (caps);
+ gst_segment_init (&input2.segment, GST_FORMAT_TIME);
+ input2.segment.start = 10 * GST_SECOND;
+ input2.input =
+ g_list_append (input2.input, gst_event_new_segment (&input2.segment));
+ input2.input =
+ g_list_append (input2.input, create_buffer (10 * GST_SECOND,
+ 10 * GST_SECOND, GST_SECOND, 4096));
+ input2.input =
+ g_list_append (input2.input, create_buffer (11 * GST_SECOND,
+ 11 * GST_SECOND, GST_SECOND, 4096));
+ input2.input =
+ g_list_append (input2.input, create_buffer (12 * GST_SECOND,
+ 12 * GST_SECOND, GST_SECOND, 4096));
+ input2.input = g_list_append (input2.input, gst_event_new_eos ());
+ input2.ts_offset = GST_SECOND * 10;
+
+ run_muxing_test (&input1, &input2);
+}
+
+GST_END_TEST;
static Suite *
qtmux_suite (void)
@@ -926,6 +1417,10 @@
tcase_add_test (tc_chain, test_encodebin_qtmux);
tcase_add_test (tc_chain, test_encodebin_mp4mux);
+ tcase_add_test (tc_chain, test_muxing);
+ tcase_add_test (tc_chain, test_muxing_non_zero_segment);
+ tcase_add_test (tc_chain, test_muxing_non_zero_segment_different);
+
return s;
}
diff --git a/tests/check/elements/rtp-payloading.c b/tests/check/elements/rtp-payloading.c
index a5db021..7dc9ce0 100644
--- a/tests/check/elements/rtp-payloading.c
+++ b/tests/check/elements/rtp-payloading.c
@@ -476,7 +476,12 @@
GST_START_TEST (rtp_h263)
{
rtp_pipeline_test (rtp_h263_frame_data, rtp_h263_frame_data_size,
- rtp_h263_frame_count, "video/x-h263,variant=(string)itu,h263version=h263",
+ rtp_h263_frame_count,
+ "video/x-h263,variant=(string)itu,h263version=h263",
+ "rtph263pay", "rtph263depay", 0, 0, FALSE);
+ rtp_pipeline_test (rtp_h263_frame_data, rtp_h263_frame_data_size,
+ rtp_h263_frame_count,
+ "video/x-h263,variant=(string)itu,h263version=h263,width=10,height=20",
"rtph263pay", "rtph263depay", 0, 0, FALSE);
}
diff --git a/tests/check/elements/rtpjitterbuffer.c b/tests/check/elements/rtpjitterbuffer.c
index 3f71131..cc1cd02 100644
--- a/tests/check/elements/rtpjitterbuffer.c
+++ b/tests/check/elements/rtpjitterbuffer.c
@@ -1019,12 +1019,13 @@
gst_clock_id_unref (id);
/* put second buffer, the jitterbuffer should now know that the packet spacing
- * is 20ms and should ask for retransmission of seqnum 2 in 20ms */
+ * is 20ms and should ask for retransmission of seqnum 2 in 20ms+10ms because
+ * 2*jitter==0 and 0.5*packet_spacing==10ms */
in_buf = generate_test_buffer (20 * GST_MSECOND, TRUE, 1, 160);
g_assert_cmpint (gst_pad_push (data.test_src_pad, in_buf), ==, GST_FLOW_OK);
gst_test_clock_wait_for_next_pending_id (GST_TEST_CLOCK (data.clock), &id);
- gst_test_clock_set_time (GST_TEST_CLOCK (data.clock), 60 * GST_MSECOND);
+ gst_test_clock_set_time (GST_TEST_CLOCK (data.clock), 50 * GST_MSECOND);
tid = gst_test_clock_process_next_clock_id (GST_TEST_CLOCK (data.clock));
g_assert (tid == id);
gst_clock_id_unref (tid);
@@ -1032,11 +1033,12 @@
out_event = g_async_queue_pop (data.src_event_queue);
g_assert (out_event != NULL);
- verify_rtx_event (out_event, 2, 40 * GST_MSECOND, 20, 20 * GST_MSECOND);
+ verify_rtx_event (out_event, 2, 40 * GST_MSECOND, 10, 20 * GST_MSECOND);
- /* now we wait for the next timeout */
+ /* now we wait for the next timeout, all following timeouts 40ms in the
+ * future because this is rtx-retry-timeout */
gst_test_clock_wait_for_next_pending_id (GST_TEST_CLOCK (data.clock), &id);
- gst_test_clock_set_time (GST_TEST_CLOCK (data.clock), 100 * GST_MSECOND);
+ gst_test_clock_set_time (GST_TEST_CLOCK (data.clock), 90 * GST_MSECOND);
tid = gst_test_clock_process_next_clock_id (GST_TEST_CLOCK (data.clock));
g_assert (id == tid);
gst_clock_id_unref (tid);
@@ -1044,10 +1046,10 @@
out_event = g_async_queue_pop (data.src_event_queue);
g_assert (out_event != NULL);
- verify_rtx_event (out_event, 2, 40 * GST_MSECOND, 60, 20 * GST_MSECOND);
+ verify_rtx_event (out_event, 2, 40 * GST_MSECOND, 50, 20 * GST_MSECOND);
gst_test_clock_wait_for_next_pending_id (GST_TEST_CLOCK (data.clock), &id);
- gst_test_clock_set_time (GST_TEST_CLOCK (data.clock), 140 * GST_MSECOND);
+ gst_test_clock_set_time (GST_TEST_CLOCK (data.clock), 130 * GST_MSECOND);
tid = gst_test_clock_process_next_clock_id (GST_TEST_CLOCK (data.clock));
g_assert (id == tid);
gst_clock_id_unref (tid);
@@ -1055,7 +1057,7 @@
out_event = g_async_queue_pop (data.src_event_queue);
g_assert (out_event != NULL);
- verify_rtx_event (out_event, 2, 40 * GST_MSECOND, 100, 20 * GST_MSECOND);
+ verify_rtx_event (out_event, 2, 40 * GST_MSECOND, 90, 20 * GST_MSECOND);
gst_test_clock_wait_for_next_pending_id (GST_TEST_CLOCK (data.clock), &id);
gst_test_clock_set_time (GST_TEST_CLOCK (data.clock), 200 * GST_MSECOND);
@@ -1070,7 +1072,7 @@
gst_test_clock_wait_for_next_pending_id (GST_TEST_CLOCK (data.clock), &id);
- gst_test_clock_set_time (GST_TEST_CLOCK (data.clock), 260 * GST_MSECOND);
+ gst_test_clock_set_time (GST_TEST_CLOCK (data.clock), 240 * GST_MSECOND);
tid = gst_test_clock_process_next_clock_id (GST_TEST_CLOCK (data.clock));
g_assert (tid == id);
gst_clock_id_unref (tid);
@@ -1112,7 +1114,8 @@
gst_test_clock_set_time (GST_TEST_CLOCK (data.clock), 20 * GST_MSECOND);
/* put second buffer, the jitterbuffer should now know that the packet spacing
- * is 20ms and should ask for retransmission of seqnum 2 at 60ms */
+ * is 20ms and should ask for retransmission of seqnum 2 in 20ms+10ms because
+ * 2*jitter==0 and 0.5*packet_spacing==10ms */
in_buf = generate_test_buffer (20 * GST_MSECOND, TRUE, 1, 160);
g_assert_cmpint (gst_pad_push (data.test_src_pad, in_buf), ==, GST_FLOW_OK);
@@ -1122,23 +1125,45 @@
g_assert_cmpint (gst_pad_push (data.test_src_pad, in_buf), ==, GST_FLOW_OK);
/* wait for first retransmission request */
- gst_test_clock_set_time (GST_TEST_CLOCK (data.clock), 60 * GST_MSECOND);
- do {
- gst_test_clock_wait_for_next_pending_id (GST_TEST_CLOCK (data.clock), &id);
- tid = gst_test_clock_process_next_clock_id (GST_TEST_CLOCK (data.clock));
- gst_clock_id_unref (id);
- gst_clock_id_unref (tid);
- } while (id != tid);
+ gst_test_clock_set_time (GST_TEST_CLOCK (data.clock), 50 * GST_MSECOND);
+ gst_test_clock_wait_for_next_pending_id (GST_TEST_CLOCK (data.clock), &id);
+ tid = gst_test_clock_process_next_clock_id (GST_TEST_CLOCK (data.clock));
+ g_assert (id == tid);
+ gst_clock_id_unref (id);
+ gst_clock_id_unref (tid);
- /* we should have 2 events now, one for 2 and another for 3 */
+ /* First event for 2 */
out_event = g_async_queue_pop (data.src_event_queue);
g_assert (out_event != NULL);
- verify_rtx_event (out_event, 2, 40 * GST_MSECOND, 20, 20 * GST_MSECOND);
+ verify_rtx_event (out_event, 2, 40 * GST_MSECOND, 10, 20 * GST_MSECOND);
+
+ /* wait for second retransmission request */
+ gst_test_clock_set_time (GST_TEST_CLOCK (data.clock), 60 * GST_MSECOND);
+ gst_test_clock_wait_for_next_pending_id (GST_TEST_CLOCK (data.clock), &id);
+ tid = gst_test_clock_process_next_clock_id (GST_TEST_CLOCK (data.clock));
+ g_assert (id == tid);
+ gst_clock_id_unref (id);
+ gst_clock_id_unref (tid);
+
+ /* Second event for 3 */
out_event = g_async_queue_pop (data.src_event_queue);
g_assert (out_event != NULL);
verify_rtx_event (out_event, 3, 60 * GST_MSECOND, 0, 20 * GST_MSECOND);
- /* now we wait for the next timeout */
+ /* now we wait for the next timeout for 2 */
+ gst_test_clock_wait_for_next_pending_id (GST_TEST_CLOCK (data.clock), &id);
+ gst_test_clock_set_time (GST_TEST_CLOCK (data.clock), 90 * GST_MSECOND);
+ tid = gst_test_clock_process_next_clock_id (GST_TEST_CLOCK (data.clock));
+ g_assert (id == tid);
+ gst_clock_id_unref (id);
+ gst_clock_id_unref (tid);
+
+ /* First event for 2 */
+ out_event = g_async_queue_pop (data.src_event_queue);
+ g_assert (out_event != NULL);
+ verify_rtx_event (out_event, 2, 40 * GST_MSECOND, 50, 20 * GST_MSECOND);
+
+ /* now we wait for the next timeout for 3 */
gst_test_clock_wait_for_next_pending_id (GST_TEST_CLOCK (data.clock), &id);
gst_test_clock_set_time (GST_TEST_CLOCK (data.clock), 100 * GST_MSECOND);
tid = gst_test_clock_process_next_clock_id (GST_TEST_CLOCK (data.clock));
@@ -1146,10 +1171,7 @@
gst_clock_id_unref (id);
gst_clock_id_unref (tid);
- /* we should have 2 events now, one for 2 and another for 3 */
- out_event = g_async_queue_pop (data.src_event_queue);
- g_assert (out_event != NULL);
- verify_rtx_event (out_event, 2, 40 * GST_MSECOND, 60, 20 * GST_MSECOND);
+ /* Second event for 3 */
out_event = g_async_queue_pop (data.src_event_queue);
g_assert (out_event != NULL);
verify_rtx_event (out_event, 3, 60 * GST_MSECOND, 40, 20 * GST_MSECOND);
@@ -1165,7 +1187,7 @@
}
gst_test_clock_wait_for_next_pending_id (GST_TEST_CLOCK (data.clock), &id);
- gst_test_clock_set_time (GST_TEST_CLOCK (data.clock), 140 * GST_MSECOND);
+ gst_test_clock_set_time (GST_TEST_CLOCK (data.clock), 130 * GST_MSECOND);
tid = gst_test_clock_process_next_clock_id (GST_TEST_CLOCK (data.clock));
g_assert (id == tid);
gst_clock_id_unref (id);
@@ -1174,7 +1196,7 @@
/* now we only get requests for 2 */
out_event = g_async_queue_pop (data.src_event_queue);
g_assert (out_event != NULL);
- verify_rtx_event (out_event, 2, 40 * GST_MSECOND, 100, 20 * GST_MSECOND);
+ verify_rtx_event (out_event, 2, 40 * GST_MSECOND, 90, 20 * GST_MSECOND);
/* this is when buffer 0 deadline expires */
gst_test_clock_wait_for_next_pending_id (GST_TEST_CLOCK (data.clock), &id);
@@ -1260,7 +1282,8 @@
gst_test_clock_set_time (GST_TEST_CLOCK (data.clock), 20 * GST_MSECOND);
/* put second buffer, the jitterbuffer should now know that the packet spacing
- * is 20ms and should ask for retransmission of seqnum 2 at 60ms */
+ * is 20ms and should ask for retransmission of seqnum 2 in 20ms+10ms because
+ * 2*jitter==0 and 0.5*packet_spacing==10ms */
in_buf = generate_test_buffer (20 * GST_MSECOND, TRUE, 1, 160);
g_assert_cmpint (gst_pad_push (data.test_src_pad, in_buf), ==, GST_FLOW_OK);
@@ -1275,7 +1298,7 @@
/* we should now receive retransmission requests for 2 -> 5 */
out_event = g_async_queue_pop (data.src_event_queue);
g_assert (out_event != NULL);
- verify_rtx_event (out_event, 2, 20 * GST_MSECOND, 40, 20 * GST_MSECOND);
+ verify_rtx_event (out_event, 2, 20 * GST_MSECOND, 30, 20 * GST_MSECOND);
for (i = 3; i < 5; i++) {
GST_DEBUG ("popping %d", i);
diff --git a/tests/check/elements/rtprtx.c b/tests/check/elements/rtprtx.c
index 706f291..841f426 100644
--- a/tests/check/elements/rtprtx.c
+++ b/tests/check/elements/rtprtx.c
@@ -304,11 +304,62 @@
} RTXReceiveData;
static GstPadProbeReturn
+do_buffer_list_as_buffers_probe (GstPad * pad, GstPadProbeInfo * info,
+ gpointer user_data, GstPadProbeCallback callback)
+{
+ /* Iterate the buffer list, removing any items that we're
+ * told to drop and creating a new bufferlist. If all buffers
+ * are dropped, return DROP.
+ */
+ guint i, len;
+ GstBufferList *list;
+ GstBufferList *outlist;
+ GstPadProbeInfo buf_info = *info;
+
+ GST_INFO_OBJECT (pad, "probing each buffer in list individually");
+
+ list = gst_pad_probe_info_get_buffer_list (info);
+
+ g_return_val_if_fail (list != NULL, GST_PAD_PROBE_REMOVE);
+
+ len = gst_buffer_list_length (list);
+ outlist = gst_buffer_list_new_sized (len);
+
+ buf_info.type = GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_PUSH;
+ for (i = 0; i < len; i++) {
+ GstBuffer *buffer = gst_buffer_list_get (list, i);
+ GstPadProbeReturn ret;
+ buf_info.data = buffer;
+ ret = callback (pad, &buf_info, user_data);
+ /* If the buffer wasn't dropped, add it to the output list */
+ if (ret != GST_PAD_PROBE_DROP)
+ gst_buffer_list_insert (outlist, -1, gst_buffer_ref (buffer));
+ }
+
+ len = gst_buffer_list_length (outlist);
+ if (len == 0) {
+ /* Everything was discarded, drop our outlist */
+ gst_buffer_list_unref (outlist);
+ return GST_PAD_PROBE_DROP;
+ }
+
+ /* Replace the original buffer list with the modified one */
+ gst_buffer_list_unref (list);
+ info->data = outlist;
+ return GST_PAD_PROBE_OK;
+}
+
+static GstPadProbeReturn
rtprtxsend_srcpad_probe (GstPad * pad, GstPadProbeInfo * info,
gpointer user_data)
{
GstPadProbeReturn ret = GST_PAD_PROBE_OK;
+ GST_LOG_OBJECT (pad, "here");
+ if (info->type == (GST_PAD_PROBE_TYPE_BUFFER_LIST | GST_PAD_PROBE_TYPE_PUSH))
+ return do_buffer_list_as_buffers_probe (pad, info, user_data,
+ rtprtxsend_srcpad_probe);
+
if (info->type == (GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_PUSH)) {
GstBuffer *buffer = GST_BUFFER (info->data);
RTXSendData *rtxdata = (RTXSendData *) user_data;
@@ -531,7 +582,8 @@
srcpad = gst_element_get_static_pad (rtprtxsend, "src");
gst_pad_add_probe (srcpad,
- (GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_PUSH),
+ (GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_BUFFER_LIST |
+ GST_PAD_PROBE_TYPE_PUSH),
(GstPadProbeCallback) rtprtxsend_srcpad_probe, &send_rtxdata, NULL);
sinkpad = gst_pad_get_peer (srcpad);
fail_if (sinkpad == NULL);
diff --git a/tests/check/elements/wavpackparse.c b/tests/check/elements/wavpackparse.c
index ec40909..f123afb 100644
--- a/tests/check/elements/wavpackparse.c
+++ b/tests/check/elements/wavpackparse.c
@@ -116,6 +116,9 @@
/* should decode the buffer without problems */
fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK);
+ /* inform of no further data */
+ fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
+
num_buffers = g_list_length (buffers);
/* should get 2 buffers, each one complete wavpack frame */
fail_unless_equals_int (num_buffers, 2);
@@ -177,6 +180,9 @@
/* should decode the buffer without problems */
fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK);
+ /* inform of no further data */
+ fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
+
num_buffers = g_list_length (buffers);
/* should get 1 buffers, the second non-broken one */
diff --git a/tests/check/gst-plugins-good.supp b/tests/check/gst-plugins-good.supp
index 3e987e6..e86794d 100644
--- a/tests/check/gst-plugins-good.supp
+++ b/tests/check/gst-plugins-good.supp
@@ -125,3 +125,21 @@
fun:jinit_compress_master
fun:jpeg_start_compress
}
+
+{
+ <well known zlib bug>
+ Memcheck:Cond
+ fun:inflateReset2
+ fun:inflateInit2_
+ fun:png_create_read_struct_2
+ fun:png_create_read_struct
+}
+
+{
+ <apparent strtod bug when trying to parse "infinity" from "interleaved">
+ Memcheck:Addr8
+ fun:__GI___strncasecmp_l
+ fun:____strtod_l_internal
+ fun:gst_value_deserialize_double
+}
+
diff --git a/tests/check/pipelines/wavenc.c b/tests/check/pipelines/wavenc.c
index b79fbf6..ee30c18 100644
--- a/tests/check/pipelines/wavenc.c
+++ b/tests/check/pipelines/wavenc.c
@@ -58,7 +58,7 @@
}
/*
- * gst-launch \
+ * gst-launch-1.0 \
* audiotestsrc freq=440 num-buffers=100 ! interleave name=i ! audioconvert ! wavenc ! filesink location=/tmp/mc.wav \
* audiotestsrc freq=880 num-buffers=100 ! i.
* ...
diff --git a/tests/examples/equalizer/demo.c b/tests/examples/equalizer/demo.c
index bfa4870..387544d 100644
--- a/tests/examples/equalizer/demo.c
+++ b/tests/examples/equalizer/demo.c
@@ -1,3 +1,22 @@
+/* GStreamer
+ * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
#include <stdlib.h>
#include <string.h>
#include <math.h>
@@ -100,6 +119,15 @@
gdk_window_end_paint (gtk_widget_get_window (drawingarea));
}
+static void
+dynamic_link (GstPadTemplate * templ, GstPad * newpad, gpointer user_data)
+{
+ GstPad *target = GST_PAD (user_data);
+
+ gst_pad_link (newpad, target);
+ gst_object_unref (target);
+}
+
/* receive spectral data from element message */
static gboolean
message_handler (GstBus * bus, GstMessage * message, gpointer data)
@@ -130,22 +158,48 @@
main (int argc, char *argv[])
{
GstElement *bin;
- GstElement *src, *capsfilter, *equalizer, *spectrum, *audioconvert, *sink;
+ GstElement *decodebin, *decconvert;
+ GstElement *capsfilter, *equalizer, *spectrum, *sinkconvert, *sink;
GstCaps *caps;
GstBus *bus;
- GtkWidget *appwindow, *vbox, *hbox, *widget;
- int i;
+ GtkWidget *appwindow, *vbox, *hbox, *scale;
+ int i, num_bands = NBANDS;
+
+ GOptionEntry options[] = {
+ {"bands", 'b', 0, G_OPTION_ARG_INT, &num_bands,
+ "Number of bands", NULL},
+ {NULL}
+ };
+ GOptionContext *ctx;
+ GError *err = NULL;
+
+ ctx = g_option_context_new ("- demo of audio equalizer");
+ g_option_context_add_main_entries (ctx, options, NULL);
+ g_option_context_add_group (ctx, gst_init_get_option_group ());
+ g_option_context_add_group (ctx, gtk_get_option_group (TRUE));
+
+ if (!g_option_context_parse (ctx, &argc, &argv, &err)) {
+ g_print ("Error initializing: %s\n", err->message);
+ exit (1);
+ }
+
+ if (argc < 2) {
+ g_print ("Usage: %s <uri to play>\n", argv[0]);
+ g_print (" For optional arguments: --help\n");
+ exit (-1);
+ }
gst_init (&argc, &argv);
gtk_init (&argc, &argv);
bin = gst_pipeline_new ("bin");
- /* White noise */
- src = gst_element_factory_make ("audiotestsrc", "src");
- g_object_set (G_OBJECT (src), "wave", 5, "volume", 0.8, NULL);
+ /* Uri decoding */
+ decodebin = gst_element_factory_make ("uridecodebin", "decoder");
+ g_object_set (G_OBJECT (decodebin), "uri", argv[1], NULL);
/* Force float32 samples */
+ decconvert = gst_element_factory_make ("audioconvert", "decconvert");
capsfilter = gst_element_factory_make ("capsfilter", "capsfilter");
caps =
gst_caps_new_simple ("audio/x-raw", "format", G_TYPE_STRING, "F32LE",
@@ -153,29 +207,35 @@
g_object_set (capsfilter, "caps", caps, NULL);
equalizer = gst_element_factory_make ("equalizer-nbands", "equalizer");
- g_object_set (G_OBJECT (equalizer), "num-bands", NBANDS, NULL);
+ g_object_set (G_OBJECT (equalizer), "num-bands", num_bands, NULL);
spectrum = gst_element_factory_make ("spectrum", "spectrum");
g_object_set (G_OBJECT (spectrum), "bands", spect_bands, "threshold", -80,
"post-messages", TRUE, "interval", 500 * GST_MSECOND, NULL);
- audioconvert = gst_element_factory_make ("audioconvert", "audioconvert");
+ sinkconvert = gst_element_factory_make ("audioconvert", "sinkconvert");
sink = gst_element_factory_make ("autoaudiosink", "sink");
- gst_bin_add_many (GST_BIN (bin), src, capsfilter, equalizer, spectrum,
- audioconvert, sink, NULL);
- if (!gst_element_link_many (src, capsfilter, equalizer, spectrum,
- audioconvert, sink, NULL)) {
+ gst_bin_add_many (GST_BIN (bin), decodebin, decconvert, capsfilter, equalizer,
+ spectrum, sinkconvert, sink, NULL);
+ if (!gst_element_link_many (decconvert, capsfilter, equalizer, spectrum,
+ sinkconvert, sink, NULL)) {
fprintf (stderr, "can't link elements\n");
exit (1);
}
+ /* Handle dynamic pads */
+ g_signal_connect (G_OBJECT (decodebin), "pad-added",
+ G_CALLBACK (dynamic_link), gst_element_get_static_pad (decconvert,
+ "sink"));
+
bus = gst_element_get_bus (bin);
gst_bus_add_watch (bus, message_handler, NULL);
gst_object_unref (bus);
appwindow = gtk_window_new (GTK_WINDOW_TOPLEVEL);
+ gtk_window_set_title (GTK_WINDOW (appwindow), "Equalizer Demo");
g_signal_connect (G_OBJECT (appwindow), "destroy",
G_CALLBACK (on_window_destroy), NULL);
vbox = gtk_box_new (GTK_ORIENTATION_VERTICAL, 6);
@@ -188,7 +248,7 @@
hbox = gtk_box_new (GTK_ORIENTATION_HORIZONTAL, 20);
- for (i = 0; i < NBANDS; i++) {
+ for (i = 0; i < num_bands; i++) {
GObject *band;
gdouble freq;
gdouble bw;
@@ -208,35 +268,38 @@
scales_hbox = gtk_box_new (GTK_ORIENTATION_HORIZONTAL, 6);
- widget = gtk_scale_new_with_range (GTK_ORIENTATION_VERTICAL,
+ /* Create gain scale */
+ scale = gtk_scale_new_with_range (GTK_ORIENTATION_VERTICAL,
-24.0, 12.0, 0.5);
- gtk_scale_set_draw_value (GTK_SCALE (widget), TRUE);
- gtk_scale_set_value_pos (GTK_SCALE (widget), GTK_POS_TOP);
- gtk_range_set_value (GTK_RANGE (widget), gain);
- gtk_widget_set_size_request (widget, 25, 150);
- g_signal_connect (G_OBJECT (widget), "value-changed",
+ gtk_scale_set_draw_value (GTK_SCALE (scale), TRUE);
+ gtk_scale_set_value_pos (GTK_SCALE (scale), GTK_POS_TOP);
+ gtk_range_set_value (GTK_RANGE (scale), gain);
+ gtk_widget_set_size_request (scale, 35, 150);
+ g_signal_connect (G_OBJECT (scale), "value-changed",
G_CALLBACK (on_gain_changed), (gpointer) band);
- gtk_box_pack_start (GTK_BOX (scales_hbox), widget, FALSE, FALSE, 0);
+ gtk_box_pack_start (GTK_BOX (scales_hbox), scale, FALSE, FALSE, 0);
- widget = gtk_scale_new_with_range (GTK_ORIENTATION_VERTICAL,
+ /* Create bandwidth scale */
+ scale = gtk_scale_new_with_range (GTK_ORIENTATION_VERTICAL,
0.0, 20000.0, 5.0);
- gtk_scale_set_draw_value (GTK_SCALE (widget), TRUE);
- gtk_scale_set_value_pos (GTK_SCALE (widget), GTK_POS_TOP);
- gtk_range_set_value (GTK_RANGE (widget), bw);
- gtk_widget_set_size_request (widget, 25, 150);
- g_signal_connect (G_OBJECT (widget), "value-changed",
+ gtk_scale_set_draw_value (GTK_SCALE (scale), TRUE);
+ gtk_scale_set_value_pos (GTK_SCALE (scale), GTK_POS_TOP);
+ gtk_range_set_value (GTK_RANGE (scale), bw);
+ gtk_widget_set_size_request (scale, 45, 150);
+ g_signal_connect (G_OBJECT (scale), "value-changed",
G_CALLBACK (on_bandwidth_changed), (gpointer) band);
- gtk_box_pack_start (GTK_BOX (scales_hbox), widget, TRUE, TRUE, 0);
+ gtk_box_pack_start (GTK_BOX (scales_hbox), scale, TRUE, TRUE, 0);
- widget = gtk_scale_new_with_range (GTK_ORIENTATION_VERTICAL,
+ /* Create frequency scale */
+ scale = gtk_scale_new_with_range (GTK_ORIENTATION_VERTICAL,
20.0, 20000.0, 5.0);
- gtk_scale_set_draw_value (GTK_SCALE (widget), TRUE);
- gtk_scale_set_value_pos (GTK_SCALE (widget), GTK_POS_TOP);
- gtk_range_set_value (GTK_RANGE (widget), freq);
- gtk_widget_set_size_request (widget, 25, 150);
- g_signal_connect (G_OBJECT (widget), "value-changed",
+ gtk_scale_set_draw_value (GTK_SCALE (scale), TRUE);
+ gtk_scale_set_value_pos (GTK_SCALE (scale), GTK_POS_TOP);
+ gtk_range_set_value (GTK_RANGE (scale), freq);
+ gtk_widget_set_size_request (scale, 45, 150);
+ g_signal_connect (G_OBJECT (scale), "value-changed",
G_CALLBACK (on_freq_changed), (gpointer) band);
- gtk_box_pack_start (GTK_BOX (scales_hbox), widget, TRUE, TRUE, 0);
+ gtk_box_pack_start (GTK_BOX (scales_hbox), scale, TRUE, TRUE, 0);
gtk_container_add (GTK_CONTAINER (frame), scales_hbox);
diff --git a/tests/examples/rtp/client-PCMA.c b/tests/examples/rtp/client-PCMA.c
index ee8366f..664b336 100644
--- a/tests/examples/rtp/client-PCMA.c
+++ b/tests/examples/rtp/client-PCMA.c
@@ -116,9 +116,9 @@
/* build a pipeline equivalent to:
*
- * gst-launch -v rtpbin name=rtpbin \
+ * gst-launch-1.0 -v rtpbin name=rtpbin \
* udpsrc caps=$AUDIO_CAPS port=5002 ! rtpbin.recv_rtp_sink_0 \
- * rtpbin. ! rtppcmadepay ! alawdec ! audioconvert ! audioresample ! alsasink \
+ * rtpbin. ! rtppcmadepay ! alawdec ! audioconvert ! audioresample ! autoaudiosink \
* udpsrc port=5003 ! rtpbin.recv_rtcp_sink_0 \
* rtpbin.send_rtcp_src_0 ! udpsink port=5007 host=$DEST sync=false async=false
*/
diff --git a/tests/examples/rtp/server-alsasrc-PCMA.c b/tests/examples/rtp/server-alsasrc-PCMA.c
index 4690eed..af8f7de 100644
--- a/tests/examples/rtp/server-alsasrc-PCMA.c
+++ b/tests/examples/rtp/server-alsasrc-PCMA.c
@@ -108,7 +108,7 @@
/* build a pipeline equivalent to:
*
- * gst-launch -v rtpbin name=rtpbin \
+ * gst-launch-1.0 -v rtpbin name=rtpbin \
* $AUDIO_SRC ! audioconvert ! audioresample ! $AUDIO_ENC ! $AUDIO_PAY ! rtpbin.send_rtp_sink_0 \
* rtpbin.send_rtp_src_0 ! udpsink port=5002 host=$DEST \
* rtpbin.send_rtcp_src_0 ! udpsink port=5003 host=$DEST sync=false async=false \
diff --git a/tests/icles/test-segment-seeks.c b/tests/icles/test-segment-seeks.c
index ebd5db4..4c9b670 100644
--- a/tests/icles/test-segment-seeks.c
+++ b/tests/icles/test-segment-seeks.c
@@ -37,6 +37,7 @@
GstMessage *msg;
gchar *uri;
gint64 dur, start, stop;
+ gboolean prerolled = FALSE;
if (argc < 2) {
g_printerr ("Usage: %s FILENAME\n", argv[0]);
@@ -74,6 +75,7 @@
GST_CLOCK_TIME_NONE, GST_MESSAGE_ASYNC_DONE | GST_MESSAGE_ERROR);
g_assert (GST_MESSAGE_TYPE (msg) != GST_MESSAGE_ERROR);
+ prerolled = TRUE;
gst_message_unref (msg);
@@ -86,11 +88,14 @@
do {
GstSeekFlags seek_flags;
gboolean ret;
+ gboolean segment_done = FALSE;
seek_flags = GST_SEEK_FLAG_ACCURATE | GST_SEEK_FLAG_SEGMENT;
- if (start == 0)
+ if (start == 0) {
+ prerolled = FALSE;
seek_flags |= GST_SEEK_FLAG_FLUSH;
+ }
stop = start + SEGMENT_DURATION;
@@ -102,24 +107,34 @@
g_assert (ret);
- if (start == 0) {
- /* wait for preroll again */
- msg = gst_bus_timed_pop_filtered (GST_ELEMENT_BUS (playbin),
- GST_CLOCK_TIME_NONE, GST_MESSAGE_ASYNC_DONE | GST_MESSAGE_ERROR);
+ if (!prerolled) {
+ while (!prerolled) {
+ /* wait for preroll again */
+ msg = gst_bus_timed_pop_filtered (GST_ELEMENT_BUS (playbin),
+ GST_CLOCK_TIME_NONE, GST_MESSAGE_SEGMENT_DONE |
+ GST_MESSAGE_ASYNC_DONE | GST_MESSAGE_ERROR);
- g_assert (GST_MESSAGE_TYPE (msg) != GST_MESSAGE_ERROR);
- gst_message_unref (msg);
+ g_assert (GST_MESSAGE_TYPE (msg) != GST_MESSAGE_ERROR);
+ if (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_SEGMENT_DONE) {
+ segment_done = TRUE;
+ } else {
+ prerolled = TRUE;
+ }
+ gst_message_unref (msg);
+ }
gst_element_set_state (playbin, GST_STATE_PLAYING);
}
- /* wait for end of segment */
- msg = gst_bus_timed_pop_filtered (GST_ELEMENT_BUS (playbin),
- GST_CLOCK_TIME_NONE, GST_MESSAGE_SEGMENT_DONE | GST_MESSAGE_ERROR);
+ /* wait for end of segment if we didn't get it above already */
+ if (!segment_done) {
+ msg = gst_bus_timed_pop_filtered (GST_ELEMENT_BUS (playbin),
+ GST_CLOCK_TIME_NONE, GST_MESSAGE_SEGMENT_DONE | GST_MESSAGE_ERROR);
- g_assert (GST_MESSAGE_TYPE (msg) != GST_MESSAGE_ERROR);
+ g_assert (GST_MESSAGE_TYPE (msg) != GST_MESSAGE_ERROR);
- gst_message_unref (msg);
+ gst_message_unref (msg);
+ }
start = stop;
}
