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Initial Tomahawk import.

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This commit is contained in:
Christian Muehlhaeuser
2010-10-17 05:32:01 +02:00
commit 1f592fbbd9
413 changed files with 48521 additions and 0 deletions
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11
.gitignore vendored Normal file
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qtcreator-build/*
build/*
.directory
*.a
*.o
._*
*.user
Makefile*
moc_*
*~
/playdar

31
CMakeLists.txt Normal file
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PROJECT( tomahawk )
CMAKE_MINIMUM_REQUIRED( VERSION 2.8 )
SET( CMAKE_MODULE_PATH "${CMAKE_MODULE_PATH}" "${CMAKE_CURRENT_SOURCE_DIR}/CMakeModules" )
# Check if we need qtgui:
IF( "${gui}" STREQUAL "no" )
ADD_DEFINITIONS( -DENABLE_HEADLESS )
MESSAGE( STATUS "Building in HEADLESS mode ***" )
FIND_PACKAGE( Qt4 4.6.0 COMPONENTS QtCore QtXml QtNetwork REQUIRED )
ELSE()
MESSAGE( STATUS "Building full GUI version ***" )
FIND_PACKAGE( Qt4 4.6.0 COMPONENTS QtGui QtCore QtXml QtNetwork REQUIRED )
ENDIF()
FIND_PACKAGE( Taglib 1.6.0 REQUIRED )
FIND_PACKAGE( LibLastFm REQUIRED )
IF( "${gui}" STREQUAL "no" )
ELSE()
IF( UNIX AND NOT APPLE )
ADD_SUBDIRECTORY( alsa-playback )
ELSE()
ADD_SUBDIRECTORY( rtaudio )
ENDIF( UNIX AND NOT APPLE )
ENDIF()
ADD_SUBDIRECTORY( libportfwd )
ADD_SUBDIRECTORY( qxtweb-standalone )
ADD_SUBDIRECTORY( src )

42
CMakeModules/FindLibLastFm.cmake Normal file
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# - Find LibLastFM
# Find the liblastfm includes and the liblastfm libraries
# This module defines
# LIBLASTFM_INCLUDE_DIR, root lastfm include dir
# LIBLASTFM_LIBRARY, the path to liblastfm
# LIBLASTFM_FOUND, whether liblastfm was found
find_path(LIBLASTFM_INCLUDE_DIR NAMES Audioscrobbler
HINTS
~/usr/include
/opt/local/include
/usr/include
/usr/local/include
/opt/kde4/include
${KDE4_INCLUDE_DIR}
PATH_SUFFIXES lastfm
)
find_library( LIBLASTFM_LIBRARY NAMES lastfm
PATHS
~/usr/lib
/opt/local/lib
/usr/lib
/usr/lib64
/usr/local/lib
/opt/kde4/lib
${KDE4_LIB_DIR}
)
if(LIBLASTFM_INCLUDE_DIR AND LIBLASTFM_LIBRARY)
set(LIBLASTFM_FOUND TRUE)
message(STATUS "Found liblastfm: ${LIBLASTFM_INCLUDE_DIR}, ${LIBLASTFM_LIBRARY}")
else(LIBLASTFM_INCLUDE_DIR AND LIBLASTFM_LIBRARY)
set(LIBLASTFM_FOUND FALSE)
if (LIBLASTFM_FIND_REQUIRED)
message(FATAL_ERROR "Could NOT find required package LibLastFm")
endif(LIBLASTFM_FIND_REQUIRED)
endif(LIBLASTFM_INCLUDE_DIR AND LIBLASTFM_LIBRARY)
mark_as_advanced(LIBLASTFM_INCLUDE_DIR LIBLASTFM_LIBRARY)

113
CMakeModules/FindLibraryWithDebug.cmake Normal file
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#
# FIND_LIBRARY_WITH_DEBUG
# -> enhanced FIND_LIBRARY to allow the search for an
# optional debug library with a WIN32_DEBUG_POSTFIX similar
# to CMAKE_DEBUG_POSTFIX when creating a shared lib
# it has to be the second and third argument
# Copyright (c) 2007, Christian Ehrlicher, <ch.ehrlicher@gmx.de>
# Redistribution and use is allowed according to the terms of the BSD license.
# For details see the accompanying COPYING-CMAKE-SCRIPTS file.
MACRO(FIND_LIBRARY_WITH_DEBUG var_name win32_dbg_postfix_name dgb_postfix libname)
IF(NOT "${win32_dbg_postfix_name}" STREQUAL "WIN32_DEBUG_POSTFIX")
# no WIN32_DEBUG_POSTFIX -> simply pass all arguments to FIND_LIBRARY
FIND_LIBRARY(${var_name}
${win32_dbg_postfix_name}
${dgb_postfix}
${libname}
${ARGN}
)
ELSE(NOT "${win32_dbg_postfix_name}" STREQUAL "WIN32_DEBUG_POSTFIX")
IF(NOT WIN32)
# on non-win32 we don't need to take care about WIN32_DEBUG_POSTFIX
FIND_LIBRARY(${var_name} ${libname} ${ARGN})
ELSE(NOT WIN32)
# 1. get all possible libnames
SET(args ${ARGN})
SET(newargs "")
SET(libnames_release "")
SET(libnames_debug "")
LIST(LENGTH args listCount)
IF("${libname}" STREQUAL "NAMES")
SET(append_rest 0)
LIST(APPEND args " ")
FOREACH(i RANGE ${listCount})
LIST(GET args ${i} val)
IF(append_rest)
LIST(APPEND newargs ${val})
ELSE(append_rest)
IF("${val}" STREQUAL "PATHS")
LIST(APPEND newargs ${val})
SET(append_rest 1)
ELSE("${val}" STREQUAL "PATHS")
LIST(APPEND libnames_release "${val}")
LIST(APPEND libnames_debug "${val}${dgb_postfix}")
ENDIF("${val}" STREQUAL "PATHS")
ENDIF(append_rest)
ENDFOREACH(i)
ELSE("${libname}" STREQUAL "NAMES")
# just one name
LIST(APPEND libnames_release "${libname}")
LIST(APPEND libnames_debug "${libname}${dgb_postfix}")
SET(newargs ${args})
ENDIF("${libname}" STREQUAL "NAMES")
# search the release lib
FIND_LIBRARY(${var_name}_RELEASE
NAMES ${libnames_release}
${newargs}
)
# search the debug lib
FIND_LIBRARY(${var_name}_DEBUG
NAMES ${libnames_debug}
${newargs}
)
IF(${var_name}_RELEASE AND ${var_name}_DEBUG)
# both libs found
SET(${var_name} optimized ${${var_name}_RELEASE}
debug ${${var_name}_DEBUG})
ELSE(${var_name}_RELEASE AND ${var_name}_DEBUG)
IF(${var_name}_RELEASE)
# only release found
SET(${var_name} ${${var_name}_RELEASE})
ELSE(${var_name}_RELEASE)
# only debug (or nothing) found
SET(${var_name} ${${var_name}_DEBUG})
ENDIF(${var_name}_RELEASE)
ENDIF(${var_name}_RELEASE AND ${var_name}_DEBUG)
MARK_AS_ADVANCED(${var_name}_RELEASE)
MARK_AS_ADVANCED(${var_name}_DEBUG)
ENDIF(NOT WIN32)
ENDIF(NOT "${win32_dbg_postfix_name}" STREQUAL "WIN32_DEBUG_POSTFIX")
ENDMACRO(FIND_LIBRARY_WITH_DEBUG)

89
CMakeModules/FindOggVorbis.cmake Normal file
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# - Try to find the OggVorbis libraries
# Once done this will define
#
# OGGVORBIS_FOUND - system has OggVorbis
# OGGVORBIS_VERSION - set either to 1 or 2
# OGGVORBIS_INCLUDE_DIR - the OggVorbis include directory
# OGGVORBIS_LIBRARIES - The libraries needed to use OggVorbis
# OGG_LIBRARY - The Ogg library
# VORBIS_LIBRARY - The Vorbis library
# VORBISFILE_LIBRARY - The VorbisFile library
# VORBISENC_LIBRARY - The VorbisEnc library
# Copyright (c) 2006, Richard Laerkaeng, <richard@goteborg.utfors.se>
#
# Redistribution and use is allowed according to the terms of the BSD license.
# For details see the accompanying COPYING-CMAKE-SCRIPTS file.
include (CheckLibraryExists)
find_path(VORBIS_INCLUDE_DIR vorbis/vorbisfile.h)
find_path(OGG_INCLUDE_DIR ogg/ogg.h)
find_library(OGG_LIBRARY NAMES ogg)
find_library(VORBIS_LIBRARY NAMES vorbis)
find_library(VORBISFILE_LIBRARY NAMES vorbisfile)
find_library(VORBISENC_LIBRARY NAMES vorbisenc)
mark_as_advanced(VORBIS_INCLUDE_DIR OGG_INCLUDE_DIR
OGG_LIBRARY VORBIS_LIBRARY VORBISFILE_LIBRARY VORBISENC_LIBRARY)
if (VORBIS_INCLUDE_DIR AND VORBIS_LIBRARY AND VORBISFILE_LIBRARY AND VORBISENC_LIBRARY)
set(OGGVORBIS_FOUND TRUE)
set(OGGVORBIS_LIBRARIES ${OGG_LIBRARY} ${VORBIS_LIBRARY} ${VORBISFILE_LIBRARY} ${VORBISENC_LIBRARY})
set(_CMAKE_REQUIRED_LIBRARIES_TMP ${CMAKE_REQUIRED_LIBRARIES})
set(CMAKE_REQUIRED_LIBRARIES ${CMAKE_REQUIRED_LIBRARIES} ${OGGVORBIS_LIBRARIES})
check_library_exists(vorbis vorbis_bitrate_addblock "" HAVE_LIBVORBISENC2)
set(CMAKE_REQUIRED_LIBRARIES ${_CMAKE_REQUIRED_LIBRARIES_TMP})
if (HAVE_LIBVORBISENC2)
set (OGGVORBIS_VERSION 2)
else (HAVE_LIBVORBISENC2)
set (OGGVORBIS_VERSION 1)
endif (HAVE_LIBVORBISENC2)
else (VORBIS_INCLUDE_DIR AND VORBIS_LIBRARY AND VORBISFILE_LIBRARY AND VORBISENC_LIBRARY)
set (OGGVORBIS_VERSION)
set(OGGVORBIS_FOUND FALSE)
endif (VORBIS_INCLUDE_DIR AND VORBIS_LIBRARY AND VORBISFILE_LIBRARY AND VORBISENC_LIBRARY)
if (OGGVORBIS_FOUND)
if (NOT OggVorbis_FIND_QUIETLY)
message(STATUS "Found OggVorbis: ${OGGVORBIS_LIBRARIES}")
endif (NOT OggVorbis_FIND_QUIETLY)
else (OGGVORBIS_FOUND)
if (OggVorbis_FIND_REQUIRED)
message(FATAL_ERROR "Could NOT find OggVorbis libraries")
endif (OggVorbis_FIND_REQUIRED)
if (NOT OggVorbis_FIND_QUITELY)
message(STATUS "Could NOT find OggVorbis libraries")
endif (NOT OggVorbis_FIND_QUITELY)
endif (OGGVORBIS_FOUND)
#check_include_files(vorbis/vorbisfile.h HAVE_VORBISFILE_H)
#check_library_exists(ogg ogg_page_version "" HAVE_LIBOGG)
#check_library_exists(vorbis vorbis_info_init "" HAVE_LIBVORBIS)
#check_library_exists(vorbisfile ov_open "" HAVE_LIBVORBISFILE)
#check_library_exists(vorbisenc vorbis_info_clear "" HAVE_LIBVORBISENC)
#check_library_exists(vorbis vorbis_bitrate_addblock "" HAVE_LIBVORBISENC2)
#if (HAVE_LIBOGG AND HAVE_VORBISFILE_H AND HAVE_LIBVORBIS AND HAVE_LIBVORBISFILE AND HAVE_LIBVORBISENC)
# message(STATUS "Ogg/Vorbis found")
# set (VORBIS_LIBS "-lvorbis -logg")
# set (VORBISFILE_LIBS "-lvorbisfile")
# set (VORBISENC_LIBS "-lvorbisenc")
# set (OGGVORBIS_FOUND TRUE)
# if (HAVE_LIBVORBISENC2)
# set (HAVE_VORBIS 2)
# else (HAVE_LIBVORBISENC2)
# set (HAVE_VORBIS 1)
# endif (HAVE_LIBVORBISENC2)
#else (HAVE_LIBOGG AND HAVE_VORBISFILE_H AND HAVE_LIBVORBIS AND HAVE_LIBVORBISFILE AND HAVE_LIBVORBISENC)
# message(STATUS "Ogg/Vorbis not found")
#endif (HAVE_LIBOGG AND HAVE_VORBISFILE_H AND HAVE_LIBVORBIS AND HAVE_LIBVORBISFILE AND HAVE_LIBVORBISENC)

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CMakeModules/FindTaglib.cmake Normal file
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# - Try to find the Taglib library
# Once done this will define
#
# TAGLIB_FOUND - system has the taglib library
# TAGLIB_CFLAGS - the taglib cflags
# TAGLIB_LIBRARIES - The libraries needed to use taglib
# Copyright (c) 2006, Laurent Montel, <montel@kde.org>
#
# Redistribution and use is allowed according to the terms of the BSD license.
# For details see the accompanying COPYING-CMAKE-SCRIPTS file.
IF(TAGLIB_FOUND)
MESSAGE(STATUS "Using manually specified taglib locations")
ELSE()
if(NOT TAGLIB_MIN_VERSION)
set(TAGLIB_MIN_VERSION "1.6")
endif(NOT TAGLIB_MIN_VERSION)
if(NOT WIN32)
find_program(TAGLIBCONFIG_EXECUTABLE NAMES taglib-config PATHS
${BIN_INSTALL_DIR}
)
endif(NOT WIN32)
#reset vars
set(TAGLIB_LIBRARIES)
set(TAGLIB_CFLAGS)
# MESSAGE( STATUS "PATHS: ${PATHS}")
# if taglib-config has been found
if(TAGLIBCONFIG_EXECUTABLE)
exec_program(${TAGLIBCONFIG_EXECUTABLE} ARGS --version RETURN_VALUE _return_VALUE OUTPUT_VARIABLE TAGLIB_VERSION)
if(TAGLIB_VERSION STRLESS "${TAGLIB_MIN_VERSION}")
message(STATUS "TagLib version not found: version searched :${TAGLIB_MIN_VERSION}, found ${TAGLIB_VERSION}")
set(TAGLIB_FOUND FALSE)
else(TAGLIB_VERSION STRLESS "${TAGLIB_MIN_VERSION}")
exec_program(${TAGLIBCONFIG_EXECUTABLE} ARGS --libs RETURN_VALUE _return_VALUE OUTPUT_VARIABLE TAGLIB_LIBRARIES)
exec_program(${TAGLIBCONFIG_EXECUTABLE} ARGS --cflags RETURN_VALUE _return_VALUE OUTPUT_VARIABLE TAGLIB_CFLAGS)
if(TAGLIB_LIBRARIES AND TAGLIB_CFLAGS)
set(TAGLIB_FOUND TRUE)
# message(STATUS "Found taglib: ${TAGLIB_LIBRARIES}")
endif(TAGLIB_LIBRARIES AND TAGLIB_CFLAGS)
string(REGEX REPLACE " *-I" ";" TAGLIB_INCLUDES "${TAGLIB_CFLAGS}")
endif(TAGLIB_VERSION STRLESS "${TAGLIB_MIN_VERSION}")
mark_as_advanced(TAGLIB_CFLAGS TAGLIB_LIBRARIES TAGLIB_INCLUDES)
else(TAGLIBCONFIG_EXECUTABLE)
include(FindLibraryWithDebug)
include(FindPackageHandleStandardArgs)
find_path(TAGLIB_CFLAGS
NAMES
tag.h
PATH_SUFFIXES taglib
PATHS
${KDE4_INCLUDE_DIR}
${INCLUDE_INSTALL_DIR}
)
find_library_with_debug(TAGLIB_LIBRARIES
WIN32_DEBUG_POSTFIX d
NAMES tag
PATHS
${KDE4_LIB_DIR}
${LIB_INSTALL_DIR}
)
find_package_handle_standard_args(Taglib DEFAULT_MSG
TAGLIB_INCLUDES TAGLIB_LIBRARIES)
endif(TAGLIBCONFIG_EXECUTABLE)
ENDIF()
if(TAGLIB_FOUND)
if(NOT Taglib_FIND_QUIETLY AND TAGLIBCONFIG_EXECUTABLE)
message(STATUS "Found TagLib: ${TAGLIB_LIBRARIES}")
endif(NOT Taglib_FIND_QUIETLY AND TAGLIBCONFIG_EXECUTABLE)
else(TAGLIB_FOUND)
if(Taglib_FIND_REQUIRED)
message(FATAL_ERROR "Could not find Taglib")
endif(Taglib_FIND_REQUIRED)
endif(TAGLIB_FOUND)

1
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FindTaglib.cmake taken from KDE4 kdelibs/cmake/Modules

621
LICENSE.txt Normal file
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GNU GENERAL PUBLIC LICENSE
Version 3, 29 June 2007
Copyright (C) 2007 Free Software Foundation, Inc. <http://fsf.org/>
Everyone is permitted to copy and distribute verbatim copies
of this license document, but changing it is not allowed.
Preamble
The GNU General Public License is a free, copyleft license for
software and other kinds of works.
The licenses for most software and other practical works are designed
to take away your freedom to share and change the works. By contrast,
the GNU General Public License is intended to guarantee your freedom to
share and change all versions of a program--to make sure it remains free
software for all its users. We, the Free Software Foundation, use the
GNU General Public License for most of our software; it applies also to
any other work released this way by its authors. You can apply it to
your programs, too.
When we speak of free software, we are referring to freedom, not
price. Our General Public Licenses are designed to make sure that you
have the freedom to distribute copies of free software (and charge for
them if you wish), that you receive source code or can get it if you
want it, that you can change the software or use pieces of it in new
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To protect your rights, we need to prevent others from denying you
these rights or asking you to surrender the rights. Therefore, you have
certain responsibilities if you distribute copies of the software, or if
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For example, if you distribute copies of such a program, whether
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GENERAL, SPECIAL, INCIDENTAL OR CONSEQUENTIAL DAMAGES ARISING OUT OF THE
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END OF TERMS AND CONDITIONS

87
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Quickstart on Ubuntu
--------------------
sudo apt-get install build-essential cmake libtag1c2a libtag1-dev liblastfm-dev \
libqt4-dev libqt4-sql-sqlite libvorbis-dev libmad0-dev \
libasound2-dev libboost-dev zlib1g-dev libgnutls-dev pkg-config
Gloox 1.0 (XMPP library)
------------------------
See: http://camaya.net/glooxdownload
You need to build gloox 1.0 from source, Ubuntu 10.04 only packages v0.9.
$ # Download and unpack tarball
$ CXXFLAGS=-fPIC ./configure --without-openssl --with-gnutls --without-libidn --with-zlib --without-examples --without-tests
$ CXXFLAGS=-fPIC make
$ sudo make install
QJson (Qt JSON library)
-----------------------
On Ubuntu 10.04:
$ sudo apt-get install libqjson-dev
Otherwise:
See: http://sourceforge.net/projects/qjson/files/ (developed using 0.7.1)
$ # Download and unpack tarball
$ ./configure && make
$ sudo make install
Now compile Tomahawk
-------------------
$ sudo ldconfig -v | grep -Ei 'qjson|gloox'
$ mkdir build
$ cd build
$ cmake ..
$ make
$ cd ..
$ ./tomahawk
Dependencies
------------
CMake 2.8.0 http://www.cmake.org/
Qt 4.6.2 http://qt.nokia.com/
QJson 0.7.1 http://qjson.sourceforge.net/
Gloox 1.0 (0.9.x will fail) http://camaya.net/gloox/
SQLite 3.6.22 http://www.sqlite.org/
TagLib 1.6.2 http://developer.kde.org/~wheeler/taglib.html
Boost 1.3x http://www.boost.org/
Unless you enable the headless mode (no GUI), we also require the following libraries:
libmad 0.15.1b http://www.underbit.com/products/mad/
libvorbis 1.2.3 http://xiph.org/vorbis/
libogg 1.1.4 http://xiph.org/ogg/
liblastfm 0.3.0 http://github.com/mxcl/liblastfm/
Third party libraries that we ship with our source:
RtAudio 4.0.7 http://www.music.mcgill.ca/~gary/rtaudio/
MiniUPnP http://miniupnp.free.fr/
To build the app:
-----------------
$ mkdir build
$ cd build
(Pick one of the following two choices. If unsure pick the second one, you probably want a GUI)
$ cmake -Dgui=no .. # enables headless mode, build without GUI
$ cmake .. # normal build including GUI
$ make
To run the app:
---------------
$ cd .. # return to the top-level tomahawk dir
(Only run the next two commands if you installed any of the dependencies from source on Linux)
$ export LD_LIBRARY_PATH=/usr/local/lib:$LD_LIBRARY_PATH
$ sudo ldconfig -v
$ ./tomahawk
Enjoy!

28
admin/mac/Info.plist Normal file
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<?xml version="1.0" encoding="UTF-8"?>
<!DOCTYPE plist PUBLIC "-//Apple Computer//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
<plist version="1.0">
<dict>
<key>CFBundleDevelopmentRegion</key>
<string>English</string>
<key>CFBundleExecutable</key>
<string>tomahawk</string>
<key>CFBundleIdentifier</key>
<string>org.tomahawk.Tomahawk</string>
<key>CFBundleInfoDictionaryVersion</key>
<string>6.0</string>
<key>CFBundlePackageType</key>
<string>APPL</string>
<key>CFBundleVersion</key>
<string>0.0.1.0</string>
<key>CFBundleShortVersionString</key>
<string>0.0.1</string>
<key>CFBundleSignature</key>
<string>tomahawk</string>
<key>CFBundleIconFile</key>
<string>tomahawk.icns</string>
<key>CFBundleName</key>
<string>Tomahawk</string>
<key>LSMinimumSystemVersion</key>
<string>10.5.0</string>
</dict>
</plist>

64
admin/mac/add-Qt-to-bundle.sh Executable file
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#!/bin/sh
# author: max@last.fm
# usage: Run from inside the bundle root directory, eg. Last.fm.app
# The first parameter should be the QtFrameworks to copy.
# Remaining parameters are plugins to copy, directories and files are
# valid.
# eg: add-Qt-to-bundle.sh 'QtCore QtGui QtXml' \
# imageformats \
# sqldrivers/libsqlite.dylib
################################################################################
if [[ ! -d "$QTDIR/lib/QtCore.framework" ]]
then
# this dir is the location of install for the official Trolltech dmg
if [[ -d /Library/Frameworks/QtCore.framework ]]
then
QT_FRAMEWORKS_DIR=/Library/Frameworks
QT_PLUGINS_DIR=/Developer/Applications/Qt/plugins
fi
elif [[ $QTDIR ]]
then
QT_FRAMEWORKS_DIR="$QTDIR/lib"
QT_PLUGINS_DIR="$QTDIR/plugins"
fi
if [ -z $QTDIR ]
then
echo QTDIR must be set, or install the official Qt dmg
exit 1
fi
################################################################################
#first frameworks
mkdir -p Contents/Frameworks
for x in $1
do
echo "C $x"
cp -R $QT_FRAMEWORKS_DIR/$x.framework Contents/Frameworks/
done
#plugins
shift
mkdir -p Contents/MacOS
while (( "$#" ))
do
echo "C $1"
if [[ -d $QT_PLUGINS_DIR/$1 ]]
then
cp -R $QT_PLUGINS_DIR/$1 Contents/MacOS
else
dir=$(basename $(dirname $1))
mkdir Contents/MacOS/$dir
cp $QT_PLUGINS_DIR/$1 Contents/MacOS/$dir
fi
shift
done
#cleanup
find Contents/Frameworks -name Headers -o -name \*.prl -o -name \*_debug | xargs rm -rf
find Contents -name \*_debug -o -name \*_debug.dylib | xargs rm

48
admin/mac/build-release-osx.sh Executable file
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#!/bin/bash
#
# Usage: dist/build-relese-osx.sh [-j] [--no-clean]
#
# Adding the -j parameter results in building a japanese version.
################################################################################
function header {
echo -e "\033[0;34m==>\033[0;0;1m $1 \033[0;0m"
}
function die {
exit_code=$?
echo $1
exit $exit_code
}
################################################################################
ROOT=`pwd`
QTDIR=`which qmake`
QTDIR=`dirname $QTDIR`
QTDIR=`dirname $QTDIR`
test -L "$QTDIR" && QTDIR=`readlink $QTDIR`
export QMAKESPEC='macx-g++'
export QTDIR
export VERSION
################################################################################
CLEAN='1'
BUILD='1'
NOTQUICK='1'
CREATEDMG='1'
header addQt
cd tomahawk.app
# $ROOT/admin/mac/add-Qt-to-bundle.sh \
# 'QtCore QtGui QtXml QtNetwork QtSql'
header deposx
$ROOT/admin/mac/deposx.sh
header Done!

73
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#!/bin/sh
# author: max@last.fm, chris@last.fm
################################################################################
if [ -z $QTDIR ]
then
echo QTDIR must be set
exit 1
fi
cd Contents
QTLIBS=`ls Frameworks | cut -d. -f1`
LIBS=`cd MacOS && ls -fR1 | grep dylib`
################################################################################
function deposx_change
{
echo "D \`$1'"
echo $QTDIR
for y in $QTLIBS
do
install_name_tool -change $QTDIR/lib/$y.framework/Versions/4/$y \
@executable_path/../Frameworks/$y.framework/Versions/4/$y \
"$1"
install_name_tool -change $QTDIR/Cellar/qt/4.6.2/lib/$y.framework/Versions/4/$y \
@executable_path/../Frameworks/$y.framework/Versions/4/$y \
"$1"
done
for y in $LIBS
do
install_name_tool -change $y \
@executable_path/$y \
"$1"
done
}
################################################################################
# first all libraries and executables
find MacOS -type f -a -perm -100 | while read x
do
echo $x
y=$(file "$x" | grep 'Mach-O')
test -n "$y" && deposx_change "$x"
install_name_tool -change liblastfm.0.dylib @executable_path/liblastfm.0.dylib $x
install_name_tool -change /usr/local/Cellar/gloox/1.0/lib/libgloox.8.dylib @executable_path/libgloox.8.dylib $x
install_name_tool -change /usr/local/lib/libgloox.8.dylib @executable_path/libgloox.8.dylib $x
install_name_tool -change /usr/local/Cellar/taglib/1.6/lib/libtag.1.dylib @executable_path/libtag.1.dylib $x
install_name_tool -change /usr/local/Cellar/libogg/1.2.0/lib/libogg.0.dylib @executable_path/libogg.0.dylib $x
install_name_tool -change /usr/local/Cellar/libvorbis/1.3.1/lib/libvorbisfile.3.dylib @executable_path/libvorbisfile.3.dylib $x
install_name_tool -change /usr/local/Cellar/mad/0.15.1b/lib/libmad.0.dylib @executable_path/libmad.0.dylib $x
done
deposx_change MacOS/libqjson.0.7.1.dylib
deposx_change MacOS/liblastfm.0.dylib
# now Qt
for x in $QTLIBS
do
echo `pwd`
# ls -l Frameworks/$x.framework/Versions/4/$x
deposx_change Frameworks/$x.framework/Versions/4/$x
install_name_tool -id @executable_path/../Frameworks/$x.framework/Versions/4/$x \
Frameworks/$x.framework/Versions/4/$x
done

3
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# windres.exe tomahawk.rx -O coff -o tomahawk.res
# SEE: http://stackoverflow.com/questions/708238/how-do-i-add-an-icon-to-a-mingw-gcc-compiled-executable

124
admin/win/tomahawk.nsi Executable file
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; assuming the script is in ROOT/admin/win/
!define ROOTDIR "../.."
!include "MUI2.nsh"
Name "Tomahawk"
!define MUI_NAME "Tomahawk"
!define MUI_PRODUCT "Tomahawk"
!define MUI_FILE "Tomahawk"
!define MUI_VERSION "Alpha"
!define MUI_BRANDINGTEXT "Tomahawk-Player Alpha Test"
CRCCheck On
OutFile "tomahawk-setup-alpha.exe"
;ShowInstDetails "nevershow"
ShowUninstDetails "nevershow"
;SetCompressor "bzip2"
!define MUI_ICON "..\..\data\icons\tomahawk.ico"
!define MUI_UNICON "..\..\data\icons\tomahawk.ico"
;!define MUI_SPECIALBITMAP "Bitmap.bmp"
InstallDir "$PROGRAMFILES\${MUI_PRODUCT}"
;--------------------------------
;Modern UI Configuration
!define MUI_WELCOMEPAGE_TEXT "This is an Alpha release, and is still buggy.$\n$\nPlease join #tomahawk-player on irc.freenode.net"
!insertmacro MUI_PAGE_WELCOME
!insertmacro MUI_PAGE_LICENSE "${ROOTDIR}\LICENSE.txt"
;;!insertmacro MUI_PAGE_COMPONENTS
!insertmacro MUI_PAGE_DIRECTORY
!insertmacro MUI_PAGE_INSTFILES
!define MUI_FINISHPAGE_RUN "$INSTDIR\tomahawk.exe"
!insertmacro MUI_PAGE_FINISH
!insertmacro MUI_LANGUAGE "English"
;Modern UI System
;!insertmacro MUI_SYSTEM
LicenseData "${ROOTDIR}\LICENSE.txt"
Section "install"
;Add files
SetOutPath "$INSTDIR"
;Path to our DLL cache
!define DLLS "${ROOTDIR}\admin\win\dlls"
File "${ROOTDIR}\build\tomahawk.exe"
File "${ROOTDIR}\LICENSE.txt"
; QT stuff:
File "${DLLS}\QtCore4.dll"
File "${DLLS}\QtGui4.dll"
File "${DLLS}\QtNetwork4.dll"
File "${DLLS}\QtSql4.dll"
File "${DLLS}\QtXml4.dll"
SetOutPath "$INSTDIR\sqldrivers"
File "${DLLS}\sqldrivers\qsqlite4.dll"
SetOutPath "$INSTDIR"
; Cygwin/c++ stuff
File "${DLLS}\cygmad-0.dll"
File "${DLLS}\libgcc_s_dw2-1.dll"
File "${DLLS}\mingwm10.dll"
; Audio stuff
File "${DLLS}\libmad.dll"
File "${DLLS}\librtaudio.dll"
; Other
File "${DLLS}\libqjson.dll"
File "${DLLS}\libqxtweb-standalone.dll"
File "${DLLS}\libtag.dll"
;create desktop shortcut
CreateShortCut "$DESKTOP\${MUI_PRODUCT}.lnk" "$INSTDIR\${MUI_FILE}.exe" ""
;create start-menu items
CreateDirectory "$SMPROGRAMS\${MUI_PRODUCT}"
CreateShortCut "$SMPROGRAMS\${MUI_PRODUCT}\Uninstall.lnk" "$INSTDIR\Uninstall.exe" "" "$INSTDIR\Uninstall.exe" 0
CreateShortCut "$SMPROGRAMS\${MUI_PRODUCT}\${MUI_PRODUCT}.lnk" "$INSTDIR\${MUI_FILE}.exe" "" "$INSTDIR\${MUI_FILE}.exe" 0
;write uninstall information to the registry
WriteRegStr HKLM "Software\Microsoft\Windows\CurrentVersion\Uninstall\${MUI_PRODUCT}" "DisplayName" "${MUI_PRODUCT} (remove only)"
WriteRegStr HKLM "Software\Microsoft\Windows\CurrentVersion\Uninstall\${MUI_PRODUCT}" "UninstallString" "$INSTDIR\Uninstall.exe"
WriteUninstaller "$INSTDIR\Uninstall.exe"
SectionEnd
;--------------------------------
;Uninstaller Section
Section "Uninstall"
;Delete Files
RMDir /r "$INSTDIR\*.*"
;Remove the installation directory
RMDir "$INSTDIR"
;Delete Start Menu Shortcuts
Delete "$DESKTOP\${MUI_PRODUCT}.lnk"
Delete "$SMPROGRAMS\${MUI_PRODUCT}\*.*"
RmDir "$SMPROGRAMS\${MUI_PRODUCT}"
;Delete Uninstaller And Unistall Registry Entries
DeleteRegKey HKEY_LOCAL_MACHINE "SOFTWARE\${MUI_PRODUCT}"
DeleteRegKey HKEY_LOCAL_MACHINE "SOFTWARE\Microsoft\Windows\CurrentVersion\Uninstall\${MUI_PRODUCT}"
SectionEnd
Function un.onUninstSuccess
MessageBox MB_OK "You have successfully uninstalled ${MUI_PRODUCT}."
FunctionEnd

1
admin/win/tomahawk.rc Normal file
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@@ -0,0 +1 @@
ID ICON "data/tomahawk_logo_32x32.ico"

44
alsa-playback/CMakeLists.txt Normal file
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PROJECT(alsaplayback)
find_package( Qt4 REQUIRED )
include( ${QT_USE_FILE} )
CMAKE_MINIMUM_REQUIRED(VERSION 2.6 FATAL_ERROR)
SET(CMAKE_VERBOSE_MAKEFILE ON)
#SET(CMAKE_INSTALL_PREFIX ".")
SET(CMAKE_RUNTIME_OUTPUT_DIRECTORY "${CMAKE_CURRENT_SOURCE_DIR}")
SET(CMAKE_LIBRARY_OUTPUT_DIRECTORY "${CMAKE_CURRENT_SOURCE_DIR}")
SET(CMAKE_ARCHIVE_OUTPUT_DIRECTORY "${CMAKE_CURRENT_SOURCE_DIR}")
#ADD_DEFINITIONS(-Wall -O2 -DNDEBUG)
#ADD_DEFINITIONS(-fPIC)
SET(AUDIO_LIBS "")
if(UNIX AND NOT APPLE)
SET(AUDIO_LIBS "asound")
endif(UNIX AND NOT APPLE)
set( alsaplaybackSources
alsaplayback.cpp
alsaaudio.cpp
xconvert.c
)
set( alsaplaybackHeaders
alsaplayback.h
)
qt4_wrap_cpp( alsaplaybackMoc ${alsaplaybackHeaders} )
SET(final_src ${alsaplaybackMoc} ${alsaplaybackSources} ${alsaplaybackHeaders})
ADD_LIBRARY(alsaplayback STATIC ${final_src})
target_link_libraries( alsaplayback
${QT_LIBRARIES}
${AUDIO_LIBS}
)
#INSTALL(TARGETS alsaplayback ARCHIVE DESTINATION lib)

920
alsa-playback/alsaaudio.cpp Normal file
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/***************************************************************************
* Copyright (C) 2007 by John Stamp, <jstamp@users.sourceforge.net> *
* Copyright (C) 2007 by Max Howell, Last.fm Ltd. *
* Copyright (C) 2010 by Christian Muehlhaeuser <muesli@gmail.com> *
* *
* Large portions of this code are shamelessly copied from audio.c: *
* The XMMS ALSA output plugin *
* Copyright (C) 2001-2003 Matthieu Sozeau <mattam@altern.org> *
* Copyright (C) 1998-2003 Peter Alm, Mikael Alm, Olle Hallnas, *
* Thomas Nilsson and 4Front Technologies *
* Copyright (C) 1999-2007 Haavard Kvaalen *
* Copyright (C) 2005 Takashi Iwai *
* *
* This program is free software; you can redistribute it and/or modify *
* it under the terms of the GNU General Public License as published by *
* the Free Software Foundation; either version 2 of the License, or *
* (at your option) any later version. *
* *
* This program 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 General Public License for more details. *
* *
* You should have received a copy of the GNU General Public License *
* along with this program; if not, write to the *
* Free Software Foundation, Inc., *
* 59 Temple Place - Suite 330, Boston, MA 02110-1301, USA. *
***************************************************************************/
#include "alsaaudio.h"
#include <qendian.h>
#include <QDebug>
//no debug
#define snd_pcm_hw_params_dump( hwparams, logs )
#define snd_pcm_sw_params_dump( x, y )
#define snd_pcm_dump( x, y )
pthread_t AlsaAudio::audio_thread;
char* AlsaAudio::thread_buffer = NULL;
int AlsaAudio::thread_buffer_size = 0;
int AlsaAudio::rd_index = 0;
int AlsaAudio::wr_index = 0;
unsigned int AlsaAudio::pcmCounter = 0;
snd_output_t* AlsaAudio::logs = NULL;
bool AlsaAudio::going = false;
snd_pcm_t *AlsaAudio::alsa_pcm = NULL;
ssize_t AlsaAudio::hw_period_size_in = 0;
snd_format* AlsaAudio::inputf = NULL;
snd_format* AlsaAudio::outputf = NULL;
float AlsaAudio::volume = 1.0;
bool AlsaAudio::paused = false;
convert_func_t AlsaAudio::alsa_convert_func = NULL;
convert_channel_func_t AlsaAudio::alsa_stereo_convert_func = NULL;
convert_freq_func_t AlsaAudio::alsa_frequency_convert_func = NULL;
xmms_convert_buffers* AlsaAudio::convertb = NULL;
AlsaAudio::AlsaAudio()
{
}
AlsaAudio::~AlsaAudio()
{
// Close here just to be sure
// These are safe to call more than once
stopPlayback();
alsaClose();
}
/******************************************************************************
* Device Detection
******************************************************************************/
int
AlsaAudio::getCards( void )
{
int card = -1;
int err = 0;
m_devices.clear();
// First add the default PCM device
AlsaDeviceInfo dev;
dev.name = "Default PCM device (default)";
dev.device = "default";
m_devices.push_back( dev );
if ( (err = snd_card_next( &card )) != 0 )
goto getCardsFailed;
while ( card > -1 )
{
getDevicesForCard( card );
if ( (err = snd_card_next( &card )) != 0 )
goto getCardsFailed;
}
return m_devices.size();
getCardsFailed:
qDebug() << __PRETTY_FUNCTION__ << "failed:" << snd_strerror( -err );
return -1;
}
void
AlsaAudio::getDevicesForCard( int card )
{
int pcm_device = -1, err;
snd_pcm_info_t *pcm_info;
snd_ctl_t *ctl;
char *alsa_name;
QString cardName = "Unknown soundcard";
QString device_name = QString( "hw:%1" ).arg( card );
if ((err = snd_ctl_open( &ctl, device_name.toAscii(), 0 )) < 0) {
qDebug() << "Failed:" << snd_strerror( -err );
return;
}
if ((err = snd_card_get_name( card, &alsa_name )) != 0)
{
qDebug() << "Failed:" << snd_strerror( -err );
}
else
cardName = alsa_name;
snd_pcm_info_alloca( &pcm_info );
for (;;)
{
if ((err = snd_ctl_pcm_next_device( ctl, &pcm_device )) < 0)
{
qDebug() << "Failed:" << snd_strerror( -err );
pcm_device = -1;
}
if (pcm_device < 0)
break;
snd_pcm_info_set_device( pcm_info, pcm_device );
snd_pcm_info_set_subdevice( pcm_info, 0 );
snd_pcm_info_set_stream( pcm_info, SND_PCM_STREAM_PLAYBACK );
if ((err = snd_ctl_pcm_info( ctl, pcm_info )) < 0)
{
if ( err != -ENOENT )
qDebug() << "Failed: snd_ctl_pcm_info() failed"
"(" << card << ":" << pcm_device << "): "
<< snd_strerror( -err );
continue;
}
AlsaDeviceInfo dev;
dev.device = QString( "hw:%1,%2" )
.arg( card )
.arg( pcm_device );
dev.name = QString( "%1: %2 (%3)" )
.arg( cardName )
.arg( snd_pcm_info_get_name( pcm_info ) )
.arg( dev.device );
m_devices.push_back( dev );
}
snd_ctl_close( ctl );
}
AlsaDeviceInfo
AlsaAudio::getDeviceInfo( int device )
{
return m_devices[device];
}
/******************************************************************************
Device Setup
******************************************************************************/
bool
AlsaAudio::alsaOpen( QString device, AFormat format, unsigned int rate,
unsigned int channels, snd_pcm_uframes_t periodSize,
unsigned int periodCount, int minBufferCapacity )
{
int err, hw_buffer_size;
ssize_t hw_period_size;
snd_pcm_hw_params_t *hwparams;
snd_pcm_sw_params_t *swparams;
snd_pcm_uframes_t alsa_buffer_size, alsa_period_size;
inputf = snd_format_from_xmms( format, rate, channels );
convertb = xmms_convert_buffers_new();
snd_output_stdio_attach( &logs, stderr, 0 );
alsa_convert_func = NULL;
alsa_stereo_convert_func = NULL;
alsa_frequency_convert_func = NULL;
free( outputf );
outputf = snd_format_from_xmms( inputf->xmms_format, inputf->rate, inputf->channels );
qDebug() << "Opening device:" << device;
// FIXME: Can snd_pcm_open() return EAGAIN?
if ((err = snd_pcm_open( &alsa_pcm,
device.toAscii(),
SND_PCM_STREAM_PLAYBACK,
SND_PCM_NONBLOCK )) < 0)
{
qDebug() << "Failed to open pcm device (" << device << "):" << snd_strerror( -err );
alsa_pcm = NULL;
free( outputf );
outputf = NULL;
return false;
}
snd_pcm_info_t *info;
int alsa_card, alsa_device, alsa_subdevice;
snd_pcm_info_alloca( &info );
snd_pcm_info( alsa_pcm, info );
alsa_card = snd_pcm_info_get_card( info );
alsa_device = snd_pcm_info_get_device( info );
alsa_subdevice = snd_pcm_info_get_subdevice( info );
// qDebug() << "Card:" << alsa_card;
// qDebug() << "Device:" << alsa_device;
// qDebug() << "Subdevice:" << alsa_subdevice;
snd_pcm_hw_params_alloca( &hwparams );
if ( (err = snd_pcm_hw_params_any( alsa_pcm, hwparams ) ) < 0 )
{
qDebug() << "No configuration available for playback:"
<< snd_strerror( -err );
alsaClose();
return false;
}
if ( ( err = snd_pcm_hw_params_set_access( alsa_pcm, hwparams,
SND_PCM_ACCESS_RW_INTERLEAVED ) ) < 0 )
{
qDebug() << "Cannot set normal write mode:" << snd_strerror( -err );
alsaClose();
return false;
}
if ( ( err = snd_pcm_hw_params_set_format( alsa_pcm, hwparams, outputf->format ) ) < 0 )
{
// Try if one of these format work (one of them should work
// on almost all soundcards)
snd_pcm_format_t formats[] = { SND_PCM_FORMAT_S16_LE,
SND_PCM_FORMAT_S16_BE,
SND_PCM_FORMAT_U8 };
uint i;
for ( i = 0; i < sizeof( formats ) / sizeof( formats[0] ); i++ )
{
if ( snd_pcm_hw_params_set_format( alsa_pcm, hwparams, formats[i] ) == 0 )
{
outputf->format = formats[i];
break;
}
}
if ( outputf->format != inputf->format )
{
outputf->xmms_format = (AFormat)format_from_alsa( outputf->format );
qDebug() << "Converting format from" << inputf->xmms_format << "to" << outputf->xmms_format;
if ( outputf->xmms_format < 0 )
return -1;
alsa_convert_func = xmms_convert_get_func( outputf->xmms_format, inputf->xmms_format );
if ( alsa_convert_func == NULL )
{
qDebug() << "Format translation needed, but not available. Input:" << inputf->xmms_format << "; Output:" << outputf->xmms_format ;
alsaClose();
return false;
}
}
else
{
qDebug() << "Sample format not available for playback:" << snd_strerror( -err );
alsaClose();
return false;
}
}
snd_pcm_hw_params_set_channels_near( alsa_pcm, hwparams, &outputf->channels );
if ( outputf->channels != inputf->channels )
{
qDebug() << "Converting channels from" << inputf->channels << "to" << outputf->channels;
alsa_stereo_convert_func =
xmms_convert_get_channel_func( outputf->xmms_format,
outputf->channels,
inputf->channels );
if ( alsa_stereo_convert_func == NULL )
{
qDebug() << "No stereo conversion available. Format:" << outputf->xmms_format << "; Input Channels:" << inputf->channels << "; Output Channels:" << outputf->channels ;
alsaClose();
return false;
}
}
snd_pcm_hw_params_set_rate_near( alsa_pcm, hwparams, &outputf->rate, 0 );
if ( outputf->rate == 0 )
{
qDebug() << "No usable samplerate available.";
alsaClose();
return false;
}
if ( outputf->rate != inputf->rate )
{
qDebug() << "Converting samplerate from" << inputf->rate << "to" << outputf->rate ;
if ( outputf->channels < 1 || outputf->channels > 2 )
{
qDebug() << "Unsupported number of channels:" << outputf->channels << "- Resample function not available" ;
alsa_frequency_convert_func = NULL;
alsaClose();
return false;
}
alsa_frequency_convert_func =
xmms_convert_get_frequency_func( outputf->xmms_format,
outputf->channels );
if ( alsa_frequency_convert_func == NULL )
{
qDebug() << "Resample function not available. Format" << outputf->xmms_format ;
alsaClose();
return false;
}
}
outputf->sample_bits = snd_pcm_format_physical_width( outputf->format );
outputf->bps = ( outputf->rate * outputf->sample_bits * outputf->channels ) >> 3;
if ( ( err = snd_pcm_hw_params_set_period_size_near( alsa_pcm, hwparams,
&periodSize, NULL ) ) < 0 )
{
qDebug() << "Set period size failed:" << snd_strerror( -err );
alsaClose();
return false;
}
if ( ( err = snd_pcm_hw_params_set_periods_near( alsa_pcm, hwparams,
&periodCount, 0 ) ) < 0 )
{
qDebug() << "Set period count failed:" << snd_strerror( -err );
alsaClose();
return false;
}
if ( snd_pcm_hw_params( alsa_pcm, hwparams ) < 0 )
{
snd_pcm_hw_params_dump( hwparams, logs );
qDebug() << "Unable to install hw params";
alsaClose();
return false;
}
if ( ( err = snd_pcm_hw_params_get_buffer_size( hwparams, &alsa_buffer_size ) ) < 0 )
{
qDebug() << "snd_pcm_hw_params_get_buffer_size() failed:" << snd_strerror( -err );
alsaClose();
return false;
}
if ( ( err = snd_pcm_hw_params_get_period_size( hwparams, &alsa_period_size, 0 ) ) < 0 )
{
qDebug() << "snd_pcm_hw_params_get_period_size() failed:" << snd_strerror( -err );
alsaClose();
return false;
}
snd_pcm_sw_params_alloca( &swparams );
snd_pcm_sw_params_current( alsa_pcm, swparams );
if ( ( err = snd_pcm_sw_params_set_start_threshold( alsa_pcm,
swparams, alsa_buffer_size - alsa_period_size ) < 0 ) )
qDebug() << "Setting start threshold failed:" << snd_strerror( -err );
if ( snd_pcm_sw_params( alsa_pcm, swparams ) < 0 )
{
qDebug() << "Unable to install sw params";
alsaClose();
return false;
}
#ifndef QT_NO_DEBUG
snd_pcm_sw_params_dump( swparams, logs );
snd_pcm_dump( alsa_pcm, logs );
#endif
hw_period_size = snd_pcm_frames_to_bytes( alsa_pcm, alsa_period_size );
if ( inputf->bps != outputf->bps )
{
int align = ( inputf->sample_bits * inputf->channels ) / 8;
hw_period_size_in = ( (quint64)hw_period_size * inputf->bps +
outputf->bps/2 ) / outputf->bps;
hw_period_size_in -= hw_period_size_in % align;
}
else
{
hw_period_size_in = hw_period_size;
}
hw_buffer_size = snd_pcm_frames_to_bytes( alsa_pcm, alsa_buffer_size );
thread_buffer_size = minBufferCapacity * 4;
if ( thread_buffer_size < hw_buffer_size )
thread_buffer_size = hw_buffer_size * 2;
if ( thread_buffer_size < 8192 )
thread_buffer_size = 8192;
thread_buffer_size += hw_buffer_size;
thread_buffer_size -= thread_buffer_size % hw_period_size;
thread_buffer = (char*)calloc(thread_buffer_size, sizeof(char));
// qDebug() << "Device setup: period size:" << hw_period_size;
// qDebug() << "Device setup: hw_period_size_in:" << hw_period_size_in;
// qDebug() << "Device setup: hw_buffer_size:" << hw_buffer_size;
// qDebug() << "Device setup: thread_buffer_size:" << thread_buffer_size;
// qDebug() << "bits per sample:" << snd_pcm_format_physical_width( outputf->format )
// << "frame size:" << snd_pcm_frames_to_bytes( alsa_pcm, 1 )
// << "Bps:" << outputf->bps;
return true;
}
int
AlsaAudio::startPlayback()
{
int pthreadError = 0;
// We should double check this here. AlsaPlayback::initAudio
// isn't having its emitted error caught.
// So double check here to avoid a potential assert.
if ( !alsa_pcm )
return 1;
going = true;
// qDebug() << "Starting thread";
AlsaAudio* aaThread = new AlsaAudio();
pthreadError = pthread_create( &audio_thread, NULL, &alsa_loop, (void*)aaThread );
return pthreadError;
}
void
AlsaAudio::clearBuffer( void )
{
wr_index = rd_index = pcmCounter = 0;
if ( thread_buffer )
memset( thread_buffer, 0, thread_buffer_size );
}
/******************************************************************************
Play Interface
******************************************************************************/
void
AlsaAudio::alsaWrite( const QByteArray& input )
{
int cnt;
const char *src = input.data();
int length = input.size();
//qDebug() << "alsaWrite length:" << length;
while ( length > 0 )
{
int wr;
cnt = qMin(length, thread_buffer_size - wr_index);
memcpy(thread_buffer + wr_index, src, cnt);
wr = (wr_index + cnt) % thread_buffer_size;
wr_index = wr;
length -= cnt;
src += cnt;
}
}
int
AlsaAudio::get_thread_buffer_filled() const
{
if ( wr_index >= rd_index )
{
return wr_index - rd_index;
}
return ( thread_buffer_size - ( rd_index - wr_index ) );
}
// HACK: the buffer may have data, but not enough to send to the card. In that
// case we tell alsaplayback that we don't have any. This may chop off some
// data, but only at the natural end of a track. On my machine, this is at
// most 3759 bytes. That's less than 0.022 sec. It beats padding the buffer
// with 0's if the stream fails mid track. No stutter this way.
int
AlsaAudio::hasData()
{
int tempSize = get_thread_buffer_filled();
if ( tempSize < hw_period_size_in )
return 0;
else
return tempSize;
}
int
AlsaAudio::alsa_free() const
{
//qDebug() << "alsa_free:" << thread_buffer_size - get_thread_buffer_filled() - 1;
return thread_buffer_size - get_thread_buffer_filled() - 1;
}
void
AlsaAudio::setVolume ( float v )
{
volume = v;
}
void
AlsaAudio::stopPlayback()
{
if (going)
{
// Q_DEBUG_BLOCK;
going = false;
pthread_join( audio_thread, NULL );
}
}
void
AlsaAudio::alsaClose()
{
// Q_DEBUG_BLOCK;
alsa_close_pcm();
xmms_convert_buffers_destroy( convertb );
convertb = NULL;
if ( thread_buffer )
{
free(thread_buffer);
thread_buffer = NULL;
}
if ( inputf )
{
free( inputf );
inputf = NULL;
}
if (outputf )
{
free( outputf );
outputf = NULL;
}
if ( logs )
{
snd_output_close( logs );
logs = NULL;
}
}
/******************************************************************************
Play Thread
******************************************************************************/
void*
AlsaAudio::alsa_loop( void* pthis )
{
AlsaAudio* aaThread = (AlsaAudio*)pthis;
aaThread->run();
return NULL;
}
void
AlsaAudio::run()
{
int npfds = snd_pcm_poll_descriptors_count( alsa_pcm );
int wr = 0;
int err;
if ( npfds <= 0 )
goto _error;
err = snd_pcm_prepare( alsa_pcm );
if ( err < 0 )
qDebug() << "snd_pcm_prepare error:" << snd_strerror( err );
while ( going && alsa_pcm )
{
if ( !paused && get_thread_buffer_filled() >= hw_period_size_in )
{
wr = snd_pcm_wait( alsa_pcm, 10 );
if ( wr > 0 )
{
alsa_write_out_thread_data();
}
else if ( wr < 0 )
{
alsa_handle_error( wr );
}
}
else
{
struct timespec req;
req.tv_sec = 0;
req.tv_nsec = 10000000; //0.1 seconds
nanosleep( &req, NULL );
}
}
_error:
err = snd_pcm_drop( alsa_pcm );
if ( err < 0 )
qDebug() << "snd_pcm_drop error:" << snd_strerror( err );
wr_index = rd_index = 0;
memset( thread_buffer, 0, thread_buffer_size );
// qDebug() << "Exiting thread";
pthread_exit( NULL );
}
/* transfer audio data from thread buffer to h/w */
void
AlsaAudio::alsa_write_out_thread_data( void )
{
ssize_t length;
int cnt;
length = qMin( hw_period_size_in, ssize_t(get_thread_buffer_filled()) );
length = qMin( length, snd_pcm_frames_to_bytes( alsa_pcm, alsa_get_avail() ) );
while (length > 0)
{
int rd;
cnt = qMin(int(length), thread_buffer_size - rd_index);
alsa_do_write( thread_buffer + rd_index, cnt);
rd = (rd_index + cnt) % thread_buffer_size;
rd_index = rd;
length -= cnt;
}
}
/* update and get the available space on h/w buffer (in frames) */
snd_pcm_sframes_t
AlsaAudio::alsa_get_avail( void )
{
snd_pcm_sframes_t ret;
if ( alsa_pcm == NULL )
return 0;
while ( ( ret = snd_pcm_avail_update( alsa_pcm ) ) < 0 )
{
ret = alsa_handle_error( ret );
if ( ret < 0 )
{
qDebug() << "alsa_get_avail(): snd_pcm_avail_update() failed:" << snd_strerror( -ret );
return 0;
}
}
return ret;
}
/* transfer data to audio h/w; length is given in bytes
*
* data can be modified via rate conversion or
* software volume before passed to audio h/w
*/
void
AlsaAudio::alsa_do_write( void* data, ssize_t length )
{
if ( alsa_convert_func != NULL )
length = alsa_convert_func( convertb, &data, length );
if ( alsa_stereo_convert_func != NULL )
length = alsa_stereo_convert_func( convertb, &data, length );
if ( alsa_frequency_convert_func != NULL )
{
length = alsa_frequency_convert_func( convertb, &data, length,
inputf->rate,
outputf->rate );
}
volume_adjust( data, length, outputf->xmms_format );
alsa_write_audio( (char*)data, length );
}
#define VOLUME_ADJUST( type, endian ) \
do { \
type *ptr = (type*)data; \
for ( i = 0; i < length; i += 2 ) \
{ \
*ptr = qTo##endian( (type)( qFrom##endian( *ptr ) * volume ) ); \
ptr++; \
} \
} while ( 0 )
#define VOLUME_ADJUST8( type ) \
do { \
type *ptr = (type*)data; \
for ( i = 0; i < length; i++ ) \
{ \
*ptr = (type)( *ptr * volume ); \
ptr++; \
} \
} while ( 0 )
void
AlsaAudio::volume_adjust( void* data, ssize_t length, AFormat fmt )
{
ssize_t i;
if ( volume == 1.0 )
return;
switch ( fmt )
{
case FMT_S16_LE:
VOLUME_ADJUST( qint16, LittleEndian );
break;
case FMT_U16_LE:
VOLUME_ADJUST( quint16, LittleEndian );
break;
case FMT_S16_BE:
VOLUME_ADJUST( qint16, BigEndian );
break;
case FMT_U16_BE:
VOLUME_ADJUST( quint16, BigEndian );
break;
case FMT_S8:
VOLUME_ADJUST8( qint8 );
break;
case FMT_U8:
VOLUME_ADJUST8( quint8 );
break;
default:
qDebug() << __PRETTY_FUNCTION__ << "unhandled format:" << fmt ;
break;
}
}
/* transfer data to audio h/w via normal write */
void
AlsaAudio::alsa_write_audio( char *data, ssize_t length )
{
snd_pcm_sframes_t written_frames;
while ( length > 0 )
{
snd_pcm_sframes_t frames = snd_pcm_bytes_to_frames( alsa_pcm, length );
written_frames = snd_pcm_writei( alsa_pcm, data, frames );
if ( written_frames > 0 )
{
ssize_t written = snd_pcm_frames_to_bytes( alsa_pcm, written_frames );
pcmCounter += written;
length -= written;
data += written;
}
else
{
int err = alsa_handle_error( (int)written_frames );
if ( err < 0 )
{
qDebug() << __PRETTY_FUNCTION__ << "write error:" << snd_strerror( -err );
break;
}
}
}
}
/* handle generic errors */
int
AlsaAudio::alsa_handle_error( int err )
{
switch ( err )
{
case -EPIPE:
return xrun_recover();
case -ESTRPIPE:
return suspend_recover();
}
return err;
}
/* close PCM and release associated resources */
void
AlsaAudio::alsa_close_pcm( void )
{
if ( alsa_pcm )
{
int err;
snd_pcm_drop( alsa_pcm );
if ( ( err = snd_pcm_close( alsa_pcm ) ) < 0 )
qDebug() << "alsa_close_pcm() failed:" << snd_strerror( -err );
alsa_pcm = NULL;
}
}
int
AlsaAudio::format_from_alsa( snd_pcm_format_t fmt )
{
uint i;
for ( i = 0; i < sizeof( format_table ) / sizeof( format_table[0] ); i++ )
if ( format_table[i].alsa == fmt )
return format_table[i].xmms;
qDebug() << "Unsupported format:" << snd_pcm_format_name( fmt );
return -1;
}
struct snd_format*
AlsaAudio::snd_format_from_xmms( AFormat fmt, unsigned int rate, unsigned int channels )
{
struct snd_format *f = (struct snd_format*)malloc( sizeof( struct snd_format ) );
uint i;
f->xmms_format = fmt;
f->format = SND_PCM_FORMAT_UNKNOWN;
for ( i = 0; i < sizeof( format_table ) / sizeof( format_table[0] ); i++ )
{
if ( format_table[i].xmms == fmt )
{
f->format = format_table[i].alsa;
break;
}
}
/* Get rid of _NE */
for ( i = 0; i < sizeof( format_table ) / sizeof( format_table[0] ); i++ )
{
if ( format_table[i].alsa == f->format )
{
f->xmms_format = format_table[i].xmms;
break;
}
}
f->rate = rate;
f->channels = channels;
f->sample_bits = snd_pcm_format_physical_width( f->format );
f->bps = ( rate * f->sample_bits * channels ) >> 3;
return f;
}
int
AlsaAudio::xrun_recover( void )
{
#ifndef QT_NO_DEBUG
snd_pcm_status_t *alsa_status;
snd_pcm_status_alloca( &alsa_status );
if ( snd_pcm_status( alsa_pcm, alsa_status ) < 0 )
{
qDebug() << "AlsaAudio::xrun_recover(): snd_pcm_status() failed";
}
else
{
snd_pcm_status_dump( alsa_status, logs );
qDebug() << "Status:\n" << logs;
}
#endif
return snd_pcm_prepare( alsa_pcm );
}
int
AlsaAudio::suspend_recover( void )
{
int err;
while ( ( err = snd_pcm_resume( alsa_pcm ) ) == -EAGAIN )
/* wait until suspend flag is released */
sleep( 1 );
if ( err < 0 )
{
qDebug() << "alsa_handle_error(): snd_pcm_resume() failed." ;
return snd_pcm_prepare( alsa_pcm );
}
return err;
}
unsigned int
AlsaAudio::timeElapsed()
{
return pcmCounter / outputf->bps;
}

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/***************************************************************************
* Copyright (C) 2007 by John Stamp, <jstamp@users.sourceforge.net> *
* Copyright (C) 2007 by Max Howell, Last.fm Ltd. *
* Copyright (C) 2010 by Christian Muehlhaeuser <muesli@gmail.com> *
* *
* This program is free software; you can redistribute it and/or modify *
* it under the terms of the GNU General Public License as published by *
* the Free Software Foundation; either version 2 of the License, or *
* (at your option) any later version. *
* *
* This program 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 General Public License for more details. *
* *
* You should have received a copy of the GNU General Public License *
* along with this program; if not, write to the *
* Free Software Foundation, Inc., *
* 59 Temple Place - Suite 330, Boston, MA 02110-1301, USA. *
***************************************************************************/
#ifndef ALSA_AUDIO_H
#define ALSA_AUDIO_H
#include <QByteArray>
#include <QList>
#include <QString>
#include <alsa/asoundlib.h>
#include "xconvert.h"
struct AlsaDeviceInfo
{
QString name;
QString device;
};
struct snd_format
{
unsigned int rate;
unsigned int channels;
snd_pcm_format_t format;
AFormat xmms_format;
int sample_bits;
int bps;
};
static const struct
{
AFormat xmms;
snd_pcm_format_t alsa;
}
format_table[] = { { FMT_S16_LE, SND_PCM_FORMAT_S16_LE },
{ FMT_S16_BE, SND_PCM_FORMAT_S16_BE },
{ FMT_S16_NE, SND_PCM_FORMAT_S16 },
{ FMT_U16_LE, SND_PCM_FORMAT_U16_LE },
{ FMT_U16_BE, SND_PCM_FORMAT_U16_BE },
{ FMT_U16_NE, SND_PCM_FORMAT_U16 },
{ FMT_U8, SND_PCM_FORMAT_U8 },
{ FMT_S8, SND_PCM_FORMAT_S8 }, };
class AlsaAudio
{
public:
AlsaAudio();
~AlsaAudio();
int getCards();
AlsaDeviceInfo getDeviceInfo( int device );
bool alsaOpen( QString device, AFormat format, unsigned int rate,
unsigned int channels, snd_pcm_uframes_t periodSize,
unsigned int periodCount, int minBufferCapacity );
int startPlayback();
void stopPlayback();
void alsaWrite( const QByteArray& inputData );
void alsaClose();
void setVolume( float vol );
void setPaused( bool enabled ) { paused = enabled; }
unsigned int timeElapsed();
int hasData();
int get_thread_buffer_filled() const;
int alsa_free() const;
void clearBuffer();
private:
QList<AlsaDeviceInfo> m_devices;
// The following static variables are configured in either
// alsaOpen or alsaSetup and used later in the audio thread
static ssize_t hw_period_size_in;
static snd_output_t *logs;
static bool going;
static snd_pcm_t *alsa_pcm;
static snd_format* inputf;
static snd_format* outputf;
static float volume;
static bool paused;
static convert_func_t alsa_convert_func;
static convert_channel_func_t alsa_stereo_convert_func;
static convert_freq_func_t alsa_frequency_convert_func;
static xmms_convert_buffers *convertb;
static pthread_t audio_thread;
static unsigned int pcmCounter;
void getDevicesForCard( int card );
static void* alsa_loop( void* );
void run();
void alsa_write_out_thread_data();
void alsa_do_write( void* data, ssize_t length );
void volume_adjust( void* data, ssize_t length, AFormat fmt );
void alsa_write_audio( char *data, ssize_t length );
//int get_thread_buffer_filled() const;
static char* thread_buffer;
static int thread_buffer_size;
static int rd_index, wr_index;
snd_pcm_sframes_t alsa_get_avail( void );
int alsa_handle_error( int err );
int xrun_recover();
int suspend_recover();
int format_from_alsa( snd_pcm_format_t fmt );
snd_format* snd_format_from_xmms( AFormat fmt, unsigned int rate, unsigned int channels );
void alsa_close_pcm( void );
};
#endif

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/***************************************************************************
* Copyright (C) 2005 - 2010 by *
* Christian Muehlhaeuser <muesli@gmail.com> *
* Erik Jaelevik, Last.fm Ltd <erik@last.fm> *
* Max Howell, Last.fm Ltd <max@last.fm> *
* *
* This program is free software; you can redistribute it and/or modify *
* it under the terms of the GNU General Public License as published by *
* the Free Software Foundation; either version 2 of the License, or *
* (at your option) any later version. *
* *
* This program 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 General Public License for more details. *
* *
* You should have received a copy of the GNU General Public License *
* along with this program; if not, write to the *
* Free Software Foundation, Inc., *
* 51 Franklin Steet, Fifth Floor, Boston, MA 02110-1301, USA. *
***************************************************************************/
#include "alsaaudio.h"
#include "alsaplayback.h"
#include <QDebug>
#include <QStringList>
AlsaPlayback::AlsaPlayback()
: m_audio( 0 )
, m_paused( false )
, m_playing( false )
, m_volume( 0.75 )
, m_deviceNum( 0 )
{
setBufferCapacity( 32768 * 4 ); //FIXME: const value
}
AlsaPlayback::~AlsaPlayback()
{
delete m_audio;
}
bool
AlsaPlayback::haveData()
{
return ( m_audio->hasData() > 0 );
}
bool
AlsaPlayback::needData()
{
return ( m_audio->get_thread_buffer_filled() < m_bufferCapacity );
}
void
AlsaPlayback::setBufferCapacity( int size )
{
m_bufferCapacity = size;
}
int
AlsaPlayback::bufferSize()
{
return m_audio->get_thread_buffer_filled();
}
float
AlsaPlayback::volume()
{
return m_volume;
}
void
AlsaPlayback::setVolume( int volume )
{
m_volume = (float)volume / 100.0;
m_audio->setVolume( m_volume );
}
void
AlsaPlayback::triggerTimers()
{
if ( m_audio )
emit timeElapsed( m_audio->timeElapsed() );
}
QStringList
AlsaPlayback::soundSystems()
{
return QStringList() << "Alsa";
}
QStringList
AlsaPlayback::devices()
{
// Q_DEBUG_BLOCK << "Querying audio devices";
QStringList devices;
for (int i = 0, n = m_audio->getCards(); i < n; i++)
devices << m_audio->getDeviceInfo( i ).name;
return devices;
}
bool
AlsaPlayback::startPlayback()
{
if ( !m_audio )
{
goto _error;
}
if ( m_audio->startPlayback() )
{
goto _error;
}
m_playing = true;
return true;
_error:
return false;
}
void
AlsaPlayback::stopPlayback()
{
m_audio->stopPlayback();
m_paused = false;
m_playing = false;
}
void
AlsaPlayback::initAudio( long sampleRate, int channels )
{
int periodSize = 1024; // According to mplayer, these two are good defaults.
int periodCount = 16; // They create a buffer size of 16384 frames.
QString cardDevice;
delete m_audio;
m_audio = new AlsaAudio;
m_audio->clearBuffer();
cardDevice = internalSoundCardID( m_deviceNum );
// We assume host byte order
#ifdef WORDS_BIGENDIAN
if ( !m_audio->alsaOpen( cardDevice, FMT_S16_BE, sampleRate, channels, periodSize, periodCount, m_bufferCapacity ) )
#else
if ( !m_audio->alsaOpen( cardDevice, FMT_S16_LE, sampleRate, channels, periodSize, periodCount, m_bufferCapacity ) )
#endif
{
}
}
void
AlsaPlayback::processData( const QByteArray &buffer )
{
m_audio->alsaWrite( buffer );
}
void
AlsaPlayback::clearBuffers()
{
m_audio->clearBuffer();
}
QString
AlsaPlayback::internalSoundCardID( int settingsID )
{
int cards = m_audio->getCards();
if ( settingsID < cards )
return m_audio->getDeviceInfo( settingsID ).device;
else
return "default";
}
void
AlsaPlayback::pause()
{
m_paused = true;
if ( m_audio )
{
m_audio->setPaused( true );
}
}
void
AlsaPlayback::resume()
{
m_paused = false;
if ( m_audio )
m_audio->setPaused( false );
}

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/***************************************************************************
* Copyright (C) 2005 - 2010 by *
* Christian Muehlhaeuser <muesli@gmail.com> *
* Erik Jaelevik, Last.fm Ltd <erik@last.fm> *
* Max Howell, Last.fm Ltd <max@last.fm> *
* *
* This program is free software; you can redistribute it and/or modify *
* it under the terms of the GNU General Public License as published by *
* the Free Software Foundation; either version 2 of the License, or *
* (at your option) any later version. *
* *
* This program 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 General Public License for more details. *
* *
* You should have received a copy of the GNU General Public License *
* along with this program; if not, write to the *
* Free Software Foundation, Inc., *
* 51 Franklin Steet, Fifth Floor, Boston, MA 02110-1301, USA. *
***************************************************************************/
#ifndef ALSAPLAYBACK_H
#define ALSAPLAYBACK_H
#include <QObject>
class AlsaPlayback : public QObject
{
Q_OBJECT
public:
AlsaPlayback();
~AlsaPlayback();
virtual void initAudio( long sampleRate, int channels );
virtual float volume();
virtual bool isPaused() { return m_paused; }
virtual bool isPlaying() { return m_playing; }
virtual bool haveData();
virtual bool needData();
virtual void processData( const QByteArray& );
virtual void setBufferCapacity( int size );
virtual int bufferSize();
virtual QStringList soundSystems();
virtual QStringList devices();
public slots:
virtual void clearBuffers();
virtual bool startPlayback();
virtual void stopPlayback();
virtual void pause();
virtual void resume();
virtual void setVolume( int volume );
virtual void triggerTimers();
signals:
void timeElapsed( unsigned int seconds );
private:
class AlsaAudio *m_audio;
int m_bufferCapacity;
bool m_paused;
bool m_playing;
float m_volume;
int m_deviceNum;
QString internalSoundCardID( int settingsID );
};
#endif

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/*
* Copyright (C) 2001-2003 Haavard Kvaalen <havardk@xmms.org>
*
* Licensed under GNU LGPL version 2.
*/
#include <stdlib.h>
#include <stdint.h>
#include "xconvert.h"
// These are adapted from defines in gtypes.h and glibconfig.h
#ifndef FALSE
#define FALSE ( 0 )
#endif
#ifndef TRUE
#define TRUE ( !FALSE )
#endif
# define GUINT16_SWAP_LE_BE( val ) \
( ( uint16_t ) \
( \
( uint16_t ) ( ( uint16_t ) ( val ) >> 8 ) | \
( uint16_t ) ( ( uint16_t ) ( val ) << 8 ) \
) \
)
# define GINT16_SWAP_LE_BE( val ) ( ( int16_t ) GUINT16_SWAP_LE_BE ( val ) )
#ifdef WORDS_BIGENDIAN
# define IS_BIG_ENDIAN TRUE
# define GINT16_TO_BE( val ) ( ( int16_t ) ( val ) )
# define GINT16_FROM_BE( val ) ( ( int16_t ) ( val ) )
# define GUINT16_TO_BE( val ) ( ( uint16_t ) ( val ) )
# define GUINT16_FROM_BE( val ) ( ( uint16_t ) ( val ) )
# define GUINT16_TO_LE( val ) ( GUINT16_SWAP_LE_BE ( val ) )
# define GUINT16_FROM_LE( val ) ( GUINT16_SWAP_LE_BE ( val ) )
# define GINT16_TO_LE( val ) ( ( int16_t ) GUINT16_SWAP_LE_BE ( val ) )
# define GINT16_FROM_LE( val ) ( ( int16_t ) GUINT16_SWAP_LE_BE ( val ) )
#else
# define IS_BIG_ENDIAN FALSE
# define GINT16_TO_LE( val ) ( ( int16_t ) ( val ) )
# define GINT16_FROM_LE( val ) ( ( int16_t ) ( val ) )
# define GUINT16_TO_LE( val ) ( ( uint16_t ) ( val ) )
# define GUINT16_FROM_LE( val ) ( ( uint16_t ) ( val ) )
# define GUINT16_TO_BE( val ) ( GUINT16_SWAP_LE_BE ( val ) )
# define GUINT16_FROM_BE( val ) ( GUINT16_SWAP_LE_BE ( val ) )
# define GINT16_TO_BE( val ) ( ( int16_t ) GUINT16_SWAP_LE_BE ( val ) )
# define GINT16_FROM_BE( val ) ( ( int16_t ) GUINT16_SWAP_LE_BE ( val ) )
#endif
struct buffer {
void *buffer;
uint size;
};
struct xmms_convert_buffers {
struct buffer format_buffer, stereo_buffer, freq_buffer;
};
struct xmms_convert_buffers* xmms_convert_buffers_new( void )
{
return calloc( 1, sizeof( struct xmms_convert_buffers ) );
}
static void* convert_get_buffer( struct buffer *buffer, size_t size )
{
if ( size > 0 && size <= buffer->size )
return buffer->buffer;
buffer->size = size;
buffer->buffer = realloc( buffer->buffer, size );
return buffer->buffer;
}
void xmms_convert_buffers_free( struct xmms_convert_buffers* buf )
{
convert_get_buffer( &buf->format_buffer, 0 );
convert_get_buffer( &buf->stereo_buffer, 0 );
convert_get_buffer( &buf->freq_buffer, 0 );
}
void xmms_convert_buffers_destroy( struct xmms_convert_buffers* buf )
{
if ( !buf )
return;
xmms_convert_buffers_free( buf );
free( buf );
}
static int convert_swap_endian( struct xmms_convert_buffers* buf, void **data, int length )
{
uint16_t *ptr = *data;
int i;
for ( i = 0; i < length; i += 2, ptr++ )
*ptr = GUINT16_SWAP_LE_BE( *ptr );
return i;
}
static int convert_swap_sign_and_endian_to_native( struct xmms_convert_buffers* buf, void **data, int length )
{
uint16_t *ptr = *data;
int i;
for ( i = 0; i < length; i += 2, ptr++ )
*ptr = GUINT16_SWAP_LE_BE( *ptr ) ^ 1 << 15;
return i;
}
static int convert_swap_sign_and_endian_to_alien( struct xmms_convert_buffers* buf, void **data, int length )
{
uint16_t *ptr = *data;
int i;
for ( i = 0; i < length; i += 2, ptr++ )
*ptr = GUINT16_SWAP_LE_BE( *ptr ^ 1 << 15 );
return i;
}
static int convert_swap_sign16( struct xmms_convert_buffers* buf, void **data, int length )
{
int16_t *ptr = *data;
int i;
for ( i = 0; i < length; i += 2, ptr++ )
*ptr ^= 1 << 15;
return i;
}
static int convert_swap_sign8( struct xmms_convert_buffers* buf, void **data, int length )
{
int8_t *ptr = *data;
int i;
for ( i = 0; i < length; i++ )
*ptr++ ^= 1 << 7;
return i;
}
static int convert_to_8_native_endian( struct xmms_convert_buffers* buf, void **data, int length )
{
int8_t *output = *data;
int16_t *input = *data;
int i;
for ( i = 0; i < length / 2; i++ )
*output++ = *input++ >> 8;
return i;
}
static int convert_to_8_native_endian_swap_sign( struct xmms_convert_buffers* buf, void **data, int length )
{
int8_t *output = *data;
int16_t *input = *data;
int i;
for ( i = 0; i < length / 2; i++ )
*output++ = ( *input++ >> 8 ) ^ ( 1 << 7 );
return i;
}
static int convert_to_8_alien_endian( struct xmms_convert_buffers* buf, void **data, int length )
{
int8_t *output = *data;
int16_t *input = *data;
int i;
for ( i = 0; i < length / 2; i++ )
*output++ = *input++ & 0xff;
return i;
}
static int convert_to_8_alien_endian_swap_sign( struct xmms_convert_buffers* buf, void **data, int length )
{
int8_t *output = *data;
int16_t *input = *data;
int i;
for ( i = 0; i < length / 2; i++ )
*output++ = ( *input++ & 0xff ) ^ ( 1 << 7 );
return i;
}
static int convert_to_16_native_endian( struct xmms_convert_buffers* buf, void **data, int length )
{
uint8_t *input = *data;
uint16_t *output;
int i;
*data = convert_get_buffer( &buf->format_buffer, length * 2 );
output = *data;
for ( i = 0; i < length; i++ )
*output++ = *input++ << 8;
return i * 2;
}
static int convert_to_16_native_endian_swap_sign( struct xmms_convert_buffers* buf, void **data, int length )
{
uint8_t *input = *data;
uint16_t *output;
int i;
*data = convert_get_buffer( &buf->format_buffer, length * 2 );
output = *data;
for ( i = 0; i < length; i++ )
*output++ = ( *input++ << 8 ) ^ ( 1 << 15 );
return i * 2;
}
static int convert_to_16_alien_endian( struct xmms_convert_buffers* buf, void **data, int length )
{
uint8_t *input = *data;
uint16_t *output;
int i;
*data = convert_get_buffer( &buf->format_buffer, length * 2 );
output = *data;
for ( i = 0; i < length; i++ )
*output++ = *input++;
return i * 2;
}
static int convert_to_16_alien_endian_swap_sign( struct xmms_convert_buffers* buf, void **data, int length )
{
uint8_t *input = *data;
uint16_t *output;
int i;
*data = convert_get_buffer( &buf->format_buffer, length * 2 );
output = *data;
for ( i = 0; i < length; i++ )
*output++ = *input++ ^ ( 1 << 7 );
return i * 2;
}
static AFormat unnativize( AFormat fmt )
{
if ( fmt == FMT_S16_NE )
{
if ( IS_BIG_ENDIAN )
return FMT_S16_BE;
else
return FMT_S16_LE;
}
if ( fmt == FMT_U16_NE )
{
if ( IS_BIG_ENDIAN )
return FMT_U16_BE;
else
return FMT_U16_LE;
}
return fmt;
}
convert_func_t xmms_convert_get_func( AFormat output, AFormat input )
{
output = unnativize( output );
input = unnativize( input );
if ( output == input )
return NULL;
if ( ( output == FMT_U16_BE && input == FMT_U16_LE ) ||
( output == FMT_U16_LE && input == FMT_U16_BE ) ||
( output == FMT_S16_BE && input == FMT_S16_LE ) ||
( output == FMT_S16_LE && input == FMT_S16_BE ) )
return convert_swap_endian;
if ( ( output == FMT_U16_BE && input == FMT_S16_BE ) ||
( output == FMT_U16_LE && input == FMT_S16_LE ) ||
( output == FMT_S16_BE && input == FMT_U16_BE ) ||
( output == FMT_S16_LE && input == FMT_U16_LE ) )
return convert_swap_sign16;
if ( ( IS_BIG_ENDIAN &&
( ( output == FMT_U16_BE && input == FMT_S16_LE ) ||
( output == FMT_S16_BE && input == FMT_U16_LE ) ) ) ||
( !IS_BIG_ENDIAN &&
( ( output == FMT_U16_LE && input == FMT_S16_BE ) ||
( output == FMT_S16_LE && input == FMT_U16_BE ) ) ) )
return convert_swap_sign_and_endian_to_native;
if ( ( !IS_BIG_ENDIAN &&
( ( output == FMT_U16_BE && input == FMT_S16_LE ) ||
( output == FMT_S16_BE && input == FMT_U16_LE ) ) ) ||
( IS_BIG_ENDIAN &&
( ( output == FMT_U16_LE && input == FMT_S16_BE ) ||
( output == FMT_S16_LE && input == FMT_U16_BE ) ) ) )
return convert_swap_sign_and_endian_to_alien;
if ( ( IS_BIG_ENDIAN &&
( ( output == FMT_U8 && input == FMT_U16_BE ) ||
( output == FMT_S8 && input == FMT_S16_BE ) ) ) ||
( !IS_BIG_ENDIAN &&
( ( output == FMT_U8 && input == FMT_U16_LE ) ||
( output == FMT_S8 && input == FMT_S16_LE ) ) ) )
return convert_to_8_native_endian;
if ( ( IS_BIG_ENDIAN &&
( ( output == FMT_U8 && input == FMT_S16_BE ) ||
( output == FMT_S8 && input == FMT_U16_BE ) ) ) ||
( !IS_BIG_ENDIAN &&
( ( output == FMT_U8 && input == FMT_S16_LE ) ||
( output == FMT_S8 && input == FMT_U16_LE ) ) ) )
return convert_to_8_native_endian_swap_sign;
if ( ( !IS_BIG_ENDIAN &&
( ( output == FMT_U8 && input == FMT_U16_BE ) ||
( output == FMT_S8 && input == FMT_S16_BE ) ) ) ||
( IS_BIG_ENDIAN &&
( ( output == FMT_U8 && input == FMT_U16_LE ) ||
( output == FMT_S8 && input == FMT_S16_LE ) ) ) )
return convert_to_8_alien_endian;
if ( ( !IS_BIG_ENDIAN &&
( ( output == FMT_U8 && input == FMT_S16_BE ) ||
( output == FMT_S8 && input == FMT_U16_BE ) ) ) ||
( IS_BIG_ENDIAN &&
( ( output == FMT_U8 && input == FMT_S16_LE ) ||
( output == FMT_S8 && input == FMT_U16_LE ) ) ) )
return convert_to_8_alien_endian_swap_sign;
if ( ( output == FMT_U8 && input == FMT_S8 ) ||
( output == FMT_S8 && input == FMT_U8 ) )
return convert_swap_sign8;
if ( ( IS_BIG_ENDIAN &&
( ( output == FMT_U16_BE && input == FMT_U8 ) ||
( output == FMT_S16_BE && input == FMT_S8 ) ) ) ||
( !IS_BIG_ENDIAN &&
( ( output == FMT_U16_LE && input == FMT_U8 ) ||
( output == FMT_S16_LE && input == FMT_S8 ) ) ) )
return convert_to_16_native_endian;
if ( ( IS_BIG_ENDIAN &&
( ( output == FMT_U16_BE && input == FMT_S8 ) ||
( output == FMT_S16_BE && input == FMT_U8 ) ) ) ||
( !IS_BIG_ENDIAN &&
( ( output == FMT_U16_LE && input == FMT_S8 ) ||
( output == FMT_S16_LE && input == FMT_U8 ) ) ) )
return convert_to_16_native_endian_swap_sign;
if ( ( !IS_BIG_ENDIAN &&
( ( output == FMT_U16_BE && input == FMT_U8 ) ||
( output == FMT_S16_BE && input == FMT_S8 ) ) ) ||
( IS_BIG_ENDIAN &&
( ( output == FMT_U16_LE && input == FMT_U8 ) ||
( output == FMT_S16_LE && input == FMT_S8 ) ) ) )
return convert_to_16_alien_endian;
if ( ( !IS_BIG_ENDIAN &&
( ( output == FMT_U16_BE && input == FMT_S8 ) ||
( output == FMT_S16_BE && input == FMT_U8 ) ) ) ||
( IS_BIG_ENDIAN &&
( ( output == FMT_U16_LE && input == FMT_S8 ) ||
( output == FMT_S16_LE && input == FMT_U8 ) ) ) )
return convert_to_16_alien_endian_swap_sign;
//g_warning( "Translation needed, but not available.\n"
// "Input: %d; Output %d.", input, output );
return NULL;
}
static int convert_mono_to_stereo( struct xmms_convert_buffers* buf, void **data, int length, int b16 )
{
int i;
void *outbuf = convert_get_buffer( &buf->stereo_buffer, length * 2 );
if ( b16 )
{
uint16_t *output = outbuf, *input = *data;
for ( i = 0; i < length / 2; i++ )
{
*output++ = *input;
*output++ = *input;
input++;
}
}
else
{
uint8_t *output = outbuf, *input = *data;
for ( i = 0; i < length; i++ )
{
*output++ = *input;
*output++ = *input;
input++;
}
}
*data = outbuf;
return length * 2;
}
static int convert_mono_to_stereo_8( struct xmms_convert_buffers* buf, void **data, int length )
{
return convert_mono_to_stereo( buf, data, length, FALSE );
}
static int convert_mono_to_stereo_16( struct xmms_convert_buffers* buf, void **data, int length )
{
return convert_mono_to_stereo( buf, data, length, TRUE );
}
static int convert_stereo_to_mono_u8( struct xmms_convert_buffers* buf, void **data, int length )
{
uint8_t *output = *data, *input = *data;
int i;
for ( i = 0; i < length / 2; i++ )
{
uint16_t tmp;
tmp = *input++;
tmp += *input++;
*output++ = tmp / 2;
}
return length / 2;
}
static int convert_stereo_to_mono_s8( struct xmms_convert_buffers* buf, void **data, int length )
{
int8_t *output = *data, *input = *data;
int i;
for ( i = 0; i < length / 2; i++ )
{
int16_t tmp;
tmp = *input++;
tmp += *input++;
*output++ = tmp / 2;
}
return length / 2;
}
static int convert_stereo_to_mono_u16le( struct xmms_convert_buffers* buf, void **data, int length )
{
uint16_t *output = *data, *input = *data;
int i;
for ( i = 0; i < length / 4; i++ )
{
uint32_t tmp;
uint16_t stmp;
tmp = GUINT16_FROM_LE( *input );
input++;
tmp += GUINT16_FROM_LE( *input );
input++;
stmp = tmp / 2;
*output++ = GUINT16_TO_LE( stmp );
}
return length / 2;
}
static int convert_stereo_to_mono_u16be( struct xmms_convert_buffers* buf, void **data, int length )
{
uint16_t *output = *data, *input = *data;
int i;
for ( i = 0; i < length / 4; i++ )
{
uint32_t tmp;
uint16_t stmp;
tmp = GUINT16_FROM_BE( *input );
input++;
tmp += GUINT16_FROM_BE( *input );
input++;
stmp = tmp / 2;
*output++ = GUINT16_TO_BE( stmp );
}
return length / 2;
}
static int convert_stereo_to_mono_s16le( struct xmms_convert_buffers* buf, void **data, int length )
{
int16_t *output = *data, *input = *data;
int i;
for ( i = 0; i < length / 4; i++ )
{
int32_t tmp;
int16_t stmp;
tmp = GINT16_FROM_LE( *input );
input++;
tmp += GINT16_FROM_LE( *input );
input++;
stmp = tmp / 2;
*output++ = GINT16_TO_LE( stmp );
}
return length / 2;
}
static int convert_stereo_to_mono_s16be( struct xmms_convert_buffers* buf, void **data, int length )
{
int16_t *output = *data, *input = *data;
int i;
for ( i = 0; i < length / 4; i++ )
{
int32_t tmp;
int16_t stmp;
tmp = GINT16_FROM_BE( *input );
input++;
tmp += GINT16_FROM_BE( *input );
input++;
stmp = tmp / 2;
*output++ = GINT16_TO_BE( stmp );
}
return length / 2;
}
convert_channel_func_t xmms_convert_get_channel_func( AFormat fmt, int output, int input )
{
fmt = unnativize( fmt );
if ( output == input )
return NULL;
if ( input == 1 && output == 2 )
switch ( fmt )
{
case FMT_U8:
case FMT_S8:
return convert_mono_to_stereo_8;
case FMT_U16_LE:
case FMT_U16_BE:
case FMT_S16_LE:
case FMT_S16_BE:
return convert_mono_to_stereo_16;
default:
//g_warning( "Unknown format: %d"
// "No conversion available.", fmt );
return NULL;
}
if ( input == 2 && output == 1 )
switch ( fmt )
{
case FMT_U8:
return convert_stereo_to_mono_u8;
case FMT_S8:
return convert_stereo_to_mono_s8;
case FMT_U16_LE:
return convert_stereo_to_mono_u16le;
case FMT_U16_BE:
return convert_stereo_to_mono_u16be;
case FMT_S16_LE:
return convert_stereo_to_mono_s16le;
case FMT_S16_BE:
return convert_stereo_to_mono_s16be;
default:
//g_warning( "Unknown format: %d. "
// "No conversion available.", fmt );
return NULL;
}
//g_warning( "Input has %d channels, soundcard uses %d channels\n"
// "No conversion is available", input, output );
return NULL;
}
#define RESAMPLE_STEREO( sample_type, bswap ) \
do { \
const int shift = sizeof ( sample_type ); \
int i, in_samples, out_samples, x, delta; \
sample_type *inptr = *data, *outptr; \
uint nlen = ( ( ( length >> shift ) * ofreq ) / ifreq ); \
void *nbuf; \
if ( nlen == 0 ) \
break; \
nlen <<= shift; \
if ( bswap ) \
convert_swap_endian( NULL, data, length ); \
nbuf = convert_get_buffer( &buf->freq_buffer, nlen ); \
outptr = nbuf; \
in_samples = length >> shift; \
out_samples = nlen >> shift; \
delta = ( in_samples << 12 ) / out_samples; \
for ( x = 0, i = 0; i < out_samples; i++ ) \
{ \
int x1, frac; \
x1 = ( x >> 12 ) << 12; \
frac = x - x1; \
*outptr++ = \
( ( inptr[( x1 >> 12 ) << 1] * \
( ( 1<<12 ) - frac ) + \
inptr[( ( x1 >> 12 ) + 1 ) << 1] * \
frac ) >> 12 ); \
*outptr++ = \
( ( inptr[( ( x1 >> 12 ) << 1 ) + 1] * \
( ( 1<<12 ) - frac ) + \
inptr[( ( ( x1 >> 12 ) + 1 ) << 1 ) + 1] * \
frac ) >> 12 ); \
x += delta; \
} \
if ( bswap ) \
convert_swap_endian( NULL, &nbuf, nlen ); \
*data = nbuf; \
return nlen; \
} while ( 0 )
#define RESAMPLE_MONO( sample_type, bswap ) \
do { \
const int shift = sizeof ( sample_type ) - 1; \
int i, x, delta, in_samples, out_samples; \
sample_type *inptr = *data, *outptr; \
uint nlen = ( ( ( length >> shift ) * ofreq ) / ifreq ); \
void *nbuf; \
if ( nlen == 0 ) \
break; \
nlen <<= shift; \
if ( bswap ) \
convert_swap_endian( NULL, data, length ); \
nbuf = convert_get_buffer( &buf->freq_buffer, nlen ); \
outptr = nbuf; \
in_samples = length >> shift; \
out_samples = nlen >> shift; \
delta = ( ( length >> shift ) << 12 ) / out_samples; \
for ( x = 0, i = 0; i < out_samples; i++ ) \
{ \
int x1, frac; \
x1 = ( x >> 12 ) << 12; \
frac = x - x1; \
*outptr++ = \
( ( inptr[x1 >> 12] * ( ( 1<<12 ) - frac ) + \
inptr[( x1 >> 12 ) + 1] * frac ) >> 12 ); \
x += delta; \
} \
if ( bswap ) \
convert_swap_endian( NULL, &nbuf, nlen ); \
*data = nbuf; \
return nlen; \
} while ( 0 )
static int convert_resample_stereo_s16ne( struct xmms_convert_buffers* buf, void **data, int length, int ifreq, int ofreq )
{
RESAMPLE_STEREO( int16_t, FALSE );
return 0;
}
static int convert_resample_stereo_s16ae( struct xmms_convert_buffers* buf, void **data, int length, int ifreq, int ofreq )
{
RESAMPLE_STEREO( int16_t, TRUE );
return 0;
}
static int convert_resample_stereo_u16ne( struct xmms_convert_buffers* buf, void **data, int length, int ifreq, int ofreq )
{
RESAMPLE_STEREO( uint16_t, FALSE );
return 0;
}
static int convert_resample_stereo_u16ae( struct xmms_convert_buffers* buf, void **data, int length, int ifreq, int ofreq )
{
RESAMPLE_STEREO( uint16_t, TRUE );
return 0;
}
static int convert_resample_mono_s16ne( struct xmms_convert_buffers* buf, void **data, int length, int ifreq, int ofreq )
{
RESAMPLE_MONO( int16_t, FALSE );
return 0;
}
static int convert_resample_mono_s16ae( struct xmms_convert_buffers* buf, void **data, int length, int ifreq, int ofreq )
{
RESAMPLE_MONO( int16_t, TRUE );
return 0;
}
static int convert_resample_mono_u16ne( struct xmms_convert_buffers* buf, void **data, int length, int ifreq, int ofreq )
{
RESAMPLE_MONO( uint16_t, FALSE );
return 0;
}
static int convert_resample_mono_u16ae( struct xmms_convert_buffers* buf, void **data, int length, int ifreq, int ofreq )
{
RESAMPLE_MONO( uint16_t, TRUE );
return 0;
}
static int convert_resample_stereo_u8( struct xmms_convert_buffers* buf, void **data, int length, int ifreq, int ofreq )
{
RESAMPLE_STEREO( uint8_t, FALSE );
return 0;
}
static int convert_resample_mono_u8( struct xmms_convert_buffers* buf, void **data, int length, int ifreq, int ofreq )
{
RESAMPLE_MONO( uint8_t, FALSE );
return 0;
}
static int convert_resample_stereo_s8( struct xmms_convert_buffers* buf, void **data, int length, int ifreq, int ofreq )
{
RESAMPLE_STEREO( int8_t, FALSE );
return 0;
}
static int convert_resample_mono_s8( struct xmms_convert_buffers* buf, void **data, int length, int ifreq, int ofreq )
{
RESAMPLE_MONO( int8_t, FALSE );
return 0;
}
convert_freq_func_t xmms_convert_get_frequency_func( AFormat fmt, int channels )
{
fmt = unnativize( fmt );
//g_message( "fmt %d, channels: %d", fmt, channels );
if ( channels < 1 || channels > 2 )
{
//g_warning( "Unsupported number of channels: %d. "
// "Resample function not available", channels );
return NULL;
}
if ( ( IS_BIG_ENDIAN && fmt == FMT_U16_BE ) ||
( !IS_BIG_ENDIAN && fmt == FMT_U16_LE ) )
{
if ( channels == 1 )
return convert_resample_mono_u16ne;
else
return convert_resample_stereo_u16ne;
}
if ( ( IS_BIG_ENDIAN && fmt == FMT_S16_BE ) ||
( !IS_BIG_ENDIAN && fmt == FMT_S16_LE ) )
{
if ( channels == 1 )
return convert_resample_mono_s16ne;
else
return convert_resample_stereo_s16ne;
}
if ( ( !IS_BIG_ENDIAN && fmt == FMT_U16_BE ) ||
( IS_BIG_ENDIAN && fmt == FMT_U16_LE ) )
{
if ( channels == 1 )
return convert_resample_mono_u16ae;
else
return convert_resample_stereo_u16ae;
}
if ( ( !IS_BIG_ENDIAN && fmt == FMT_S16_BE ) ||
( IS_BIG_ENDIAN && fmt == FMT_S16_LE ) )
{
if ( channels == 1 )
return convert_resample_mono_s16ae;
else
return convert_resample_stereo_s16ae;
}
if ( fmt == FMT_U8 )
{
if ( channels == 1 )
return convert_resample_mono_u8;
else
return convert_resample_stereo_u8;
}
if ( fmt == FMT_S8 )
{
if ( channels == 1 )
return convert_resample_mono_s8;
else
return convert_resample_stereo_s8;
}
//g_warning( "Resample function not available"
// "Format %d.", fmt );
return NULL;
}

43
alsa-playback/xconvert.h Normal file
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@@ -0,0 +1,43 @@
/*
* Copyright (C) 2003 Haavard Kvaalen <havardk@xmms.org>
*
* Licensed under GNU LGPL version 2.
*/
#if BYTE_ORDER == BIG_ENDIAN
#define WORDS_BIGENDIAN 1
#endif
#ifdef __cplusplus
extern "C" {
#endif
typedef enum
{
FMT_U8, FMT_S8, FMT_U16_LE, FMT_U16_BE, FMT_U16_NE, FMT_S16_LE, FMT_S16_BE, FMT_S16_NE
}
AFormat;
struct xmms_convert_buffers;
struct xmms_convert_buffers* xmms_convert_buffers_new(void);
/*
* Free the data assosiated with the buffers, without destroying the
* context. The context can be reused.
*/
void xmms_convert_buffers_free(struct xmms_convert_buffers* buf);
void xmms_convert_buffers_destroy(struct xmms_convert_buffers* buf);
typedef int (*convert_func_t)(struct xmms_convert_buffers* buf, void **data, int length);
typedef int (*convert_channel_func_t)(struct xmms_convert_buffers* buf, void **data, int length);
typedef int (*convert_freq_func_t)(struct xmms_convert_buffers* buf, void **data, int length, int ifreq, int ofreq);
convert_func_t xmms_convert_get_func(AFormat output, AFormat input);
convert_channel_func_t xmms_convert_get_channel_func(AFormat fmt, int output, int input);
convert_freq_func_t xmms_convert_get_frequency_func(AFormat fmt, int channels);
#ifdef __cplusplus
}
#endif

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55
data/topbar-radiobuttons.css Normal file
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QWidget#widgetRadio {
margin:0;
padding:0;
border: 0;
}
QRadioButton {
border: 0;
margin:0;
padding:0;
background-repeat: none;
/*width:0; height:0;*/
}
QRadioButton::indicator {
width: 29px;
height: 30px;
}
QRadioButton::indicator::unchecked {
background-image: url(:/data/images/view-toggle-inactive-centre.png);
image: url(:/data/images/view-toggle-icon-artist-inactive.png);
}
QRadioButton::indicator::checked {
background-image: url(:/data/images/view-toggle-active-centre.png);
image: url(:/data/images/view-toggle-icon-artist-active.png);
}
QRadioButton::indicator::pressed {
background-image: url(:/data/images/view-toggle-pressed-centre.png);
image: url(:/data/images/view-toggle-icon-artist-active.png);
}
QRadioButton#radioNormal::indicator::unchecked {
background-image: url(:/data/images/view-toggle-inactive-left.png);
image: url(:/data/images/view-toggle-icon-list-inactive.png);
}
QRadioButton#radioNormal::indicator::checked {
background-image: url(:/data/images/view-toggle-active-left.png);
image: url(:/data/images/view-toggle-icon-list-active.png);
}
QRadioButton#radioNormal::indicator::pressed {
background-image: url(:/data/images/view-toggle-pressed-left.png);
image: url(:/data/images/view-toggle-icon-list-active.png);
}
QRadioButton#radioCloud::indicator::unchecked {
background-image: url(:/data/images/view-toggle-inactive-right.png);
image: url(:/data/images/view-toggle-icon-cloud-inactive.png);
}
QRadioButton#radioCloud::indicator::checked {
background-image: url(:/data/images/view-toggle-active-right.png);
image: url(:/data/images/view-toggle-icon-cloud-active.png);
}
QRadioButton#radioCloud::indicator::pressed {
background-image: url(:/data/images/view-toggle-pressed-right.png);
image: url(:/data/images/view-toggle-icon-cloud-active.png);
}

13
gen_resources.sh Executable file
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#!/bin/bash
echo "<!DOCTYPE RCC><RCC version=\"1.0\"><qresource>"
datadir="`pwd`/data"
cd "$datadir"
(find -type f | sed 's/^\.\///g') | while read f
do
ff="${datadir}/$f"
echo "<file alias=\"$f\">$ff</file>"
done
echo "</qresource></RCC>"

35
include/tomahawk/album.h Normal file
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#ifndef TOMAHAWKALBUM_H
#define TOMAHAWKALBUM_H
#include <QObject>
#include <QSharedPointer>
#include "artist.h"
namespace Tomahawk
{
class Album;
typedef QSharedPointer<Album> album_ptr;
class Album : public QObject
{
Q_OBJECT
public:
Album( artist_ptr artist, const QString& name )
: m_name( name )
, m_artist( artist )
{}
const QString& name() const { return m_name; }
const artist_ptr artist() const { return m_artist; }
private:
QString m_name;
artist_ptr m_artist;
};
}; // ns
#endif

30
include/tomahawk/artist.h Normal file
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#ifndef TOMAHAWKARTIST_H
#define TOMAHAWKARTIST_H
#include <QObject>
#include <QSharedPointer>
namespace Tomahawk
{
class Artist;
typedef QSharedPointer<Artist> artist_ptr;
class Artist : public QObject
{
Q_OBJECT
public:
Artist( const QString& name )
: m_name( name )
{};
const QString& name() const { return m_name; }
private:
QString m_name;
};
}; // ns
#endif

100
include/tomahawk/collection.h Normal file
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/*
The collection - acts as container for someones music library
load() -> async populate by calling addArtists etc,
then finishedLoading() is emitted.
then use artists() etc to get the data.
*/
#ifndef TOMAHAWK_COLLECTION_H
#define TOMAHAWK_COLLECTION_H
#include <QHash>
#include <QList>
#include <QSharedPointer>
#include <QDebug>
#include "tomahawk/functimeout.h"
#include "tomahawk/playlist.h"
#include "tomahawk/source.h"
#include "tomahawk/typedefs.h"
namespace Tomahawk
{
/*
Call load(), then wait for the finishedLoading() signal,
then call tracks() to get all tracks.
*/
class Collection : public QObject
{
Q_OBJECT
public:
Collection( const source_ptr& source, const QString& name, QObject* parent = 0 );
virtual ~Collection();
void invokeSlotTracks( QObject* obj, const char* slotname, const QList<QVariant>& val, collection_ptr collection );
virtual QString name() const;
virtual void loadPlaylists() = 0;
virtual Tomahawk::playlist_ptr playlist( const QString& guid );
virtual void addPlaylist( const Tomahawk::playlist_ptr& p );
virtual void deletePlaylist( const Tomahawk::playlist_ptr& p );
/// async calls that fetch data from DB/whatever:
void loadTracks( QObject* obj, const char* slotname );
virtual const QList< Tomahawk::playlist_ptr >& playlists() const { return m_playlists; }
bool isLoaded() const { return m_loaded; }
const source_ptr& source() const { return m_source; }
unsigned int lastmodified() const { return m_lastmodified; }
static bool trackSorter( const QVariant& left, const QVariant &right );
signals:
void tracksAdded( const QList<QVariant>&, Tomahawk::collection_ptr );
void tracksRemoved( const QList<QVariant>&, Tomahawk::collection_ptr );
void playlistsAdded( const QList<Tomahawk::playlist_ptr>& );
void playlistsDeleted( const QList<Tomahawk::playlist_ptr>& );
public slots:
virtual void addTracks( const QList<QVariant> &newitems ) = 0;
virtual void removeTracks( const QList<QVariant> &olditems ) = 0;
void setPlaylists( const QList<Tomahawk::playlist_ptr>& plists )
{
qDebug() << Q_FUNC_INFO << plists.length();
m_playlists.append( plists );
if( !m_loaded )
{
m_loaded = true;
emit playlistsAdded( plists );
}
}
protected:
virtual void loadAllTracks( boost::function<void( const QList<QVariant>&, collection_ptr )> callback ) = 0;
QString m_name;
bool m_loaded;
unsigned int m_lastmodified; // unix time of last change to collection
private:
source_ptr m_source;
QList< Tomahawk::playlist_ptr > m_playlists;
};
}; // ns
inline uint qHash( const QSharedPointer<Tomahawk::Collection>& key )
{
return qHash( (void *)key.data() );
}
#endif // TOMAHAWK_COLLECTION_H

51
include/tomahawk/functimeout.h Normal file
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#ifndef FUNCTIMEOUT_H
#define FUNCTIMEOUT_H
#include <QObject>
#include <QTimer>
#include <QDebug>
#include "boost/function.hpp"
#include "boost/bind.hpp"
/*
I want to do:
QTimer::singleShot(1000, this, SLOT(doSomething(x)));
instead, I'm doing:
new FuncTimeout(1000, boost::bind(&MyClass::doSomething, this, x));
*/
namespace Tomahawk
{
class FuncTimeout : public QObject
{
Q_OBJECT
public:
FuncTimeout( int ms, boost::function<void()> func )
: m_func( func )
{
//qDebug() << Q_FUNC_INFO;
QTimer::singleShot( ms, this, SLOT(exec() ) );
};
~FuncTimeout()
{
//qDebug() << Q_FUNC_INFO;
};
public slots:
void exec()
{
m_func();
this->deleteLater();
};
private:
boost::function<void()> m_func;
};
}; // ns
#endif // FUNCTIMEOUT_H

70
include/tomahawk/pipeline.h Normal file
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#ifndef PIPELINE_H
#define PIPELINE_H
#include <QObject>
#include <QList>
#include <QMap>
#include <QMutex>
#include "tomahawk/typedefs.h"
#include "tomahawk/query.h"
#include "tomahawk/result.h"
#include "tomahawk/resolver.h"
namespace Tomahawk
{
class Resolver;
class Pipeline : public QObject
{
Q_OBJECT
public:
explicit Pipeline( QObject* parent = 0 );
// const query_ptr& query( QID qid ) const;
// result_ptr result( RID rid ) const;
void reportResults( QID qid, const QList< result_ptr >& results );
/// sorter to rank resolver priority
static bool resolverSorter( const Resolver* left, const Resolver* right );
void addResolver( Resolver* r, bool sort = true );
void removeResolver( Resolver* r );
query_ptr query( const QID& qid ) const
{
return m_qids.value( qid );
}
result_ptr result( const RID& rid ) const
{
return m_rids.value( rid );
}
public slots:
void add( const query_ptr& q );
void add( const QList<query_ptr>& qlist );
void databaseReady();
private slots:
void shunt( const query_ptr& q );
void indexReady();
private:
QList< Resolver* > m_resolvers;
QMap< QID, query_ptr > m_qids;
QMap< RID, result_ptr > m_rids;
QMutex m_mut; // for m_qids, m_rids
// store queries here until DB index is loaded, then shunt them all
QList< query_ptr > m_queries_pending;
bool m_index_ready;
};
}; //ns
#endif // PIPELINE_H

190
include/tomahawk/playlist.h Normal file
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#ifndef PLAYLIST_H
#define PLAYLIST_H
#include <QObject>
#include <QList>
#include <QSharedPointer>
#include "tomahawk/query.h"
#include "tomahawk/typedefs.h"
class DatabaseCommand_LoadAllPlaylists;
class DatabaseCommand_SetPlaylistRevision;
class DatabaseCommand_CreatePlaylist;
namespace Tomahawk
{
class PlaylistEntry : public QObject
{
Q_OBJECT
Q_PROPERTY( QString guid READ guid WRITE setGuid )
Q_PROPERTY( QString annotation READ annotation WRITE setAnnotation )
Q_PROPERTY( QString resulthint READ resulthint WRITE setResulthint )
Q_PROPERTY( unsigned int duration READ duration WRITE setDuration )
Q_PROPERTY( unsigned int lastmodified READ lastmodified WRITE setLastmodified )
Q_PROPERTY( QVariant query READ queryvariant WRITE setQueryvariant )
public:
void setQuery( const Tomahawk::query_ptr& q ) { m_query = q; }
const Tomahawk::query_ptr& query() const { return m_query; }
// I wish Qt did this for me once i specified the Q_PROPERTIES:
void setQueryvariant( const QVariant& v );
QVariant queryvariant() const;
QString guid() const { return m_guid; }
void setGuid( const QString& s ) { m_guid = s; }
QString annotation() const { return m_annotation; }
void setAnnotation( const QString& s ) { m_annotation = s; }
QString resulthint() const { return m_resulthint; }
void setResulthint( const QString& s ) { m_resulthint= s; }
unsigned int duration() const { return m_duration; }
void setDuration( unsigned int i ) { m_duration = i; }
unsigned int lastmodified() const { return m_lastmodified; }
void setLastmodified( unsigned int i ) { m_lastmodified = i; }
source_ptr lastsource() const { return m_lastsource; }
void setLastsource( source_ptr s ) { m_lastsource = s ; }
private:
QString m_guid;
Tomahawk::query_ptr m_query;
QString m_annotation;
unsigned int m_duration;
unsigned int m_lastmodified;
source_ptr m_lastsource;
QString m_resulthint;
};
struct PlaylistRevision
{
QString revisionguid;
QString oldrevisionguid;
QList<plentry_ptr> newlist;
QList<plentry_ptr> added;
QList<plentry_ptr> removed;
bool applied; // false if conflict
};
class Playlist : public QObject
{
Q_OBJECT
Q_PROPERTY( QString guid READ guid WRITE setGuid )
Q_PROPERTY( QString currentrevision READ currentrevision WRITE setCurrentrevision )
Q_PROPERTY( QString title READ title WRITE setTitle )
Q_PROPERTY( QString info READ info WRITE setInfo )
Q_PROPERTY( QString creator READ creator WRITE setCreator )
Q_PROPERTY( bool shared READ shared WRITE setShared )
friend class ::DatabaseCommand_LoadAllPlaylists;
friend class ::DatabaseCommand_SetPlaylistRevision;
friend class ::DatabaseCommand_CreatePlaylist;
public:
// one CTOR is private, only called by DatabaseCommand_LoadAllPlaylists
static Tomahawk::playlist_ptr create( const source_ptr& author,
const QString& guid,
const QString& title,
const QString& info,
const QString& creator,
bool shared );
static bool remove( const playlist_ptr& playlist );
virtual void loadRevision( const QString& rev = "" );
const source_ptr& author() { return m_source; }
const QString& currentrevision() { return m_currentrevision; }
const QString& title() { return m_title; }
const QString& info() { return m_info; }
const QString& creator() { return m_creator; }
unsigned int lastmodified() { return m_lastmodified; }
const QString& guid() { return m_guid; }
bool shared() const { return m_shared; }
const QList< plentry_ptr >& entries() { return m_entries; }
void addEntry( const Tomahawk::query_ptr& query, const QString& oldrev );
void addEntries( const QList<Tomahawk::query_ptr>& queries, const QString& oldrev );
// <IGNORE hack="true">
// these need to exist and be public for the json serialization stuff
// you SHOULD NOT call them. They are used for an alternate CTOR method from json.
// maybe friend QObjectHelper and make them private?
Playlist( const source_ptr& author ) :
m_source( author )
, m_lastmodified( 0 )
{
qDebug() << Q_FUNC_INFO << "JSON";
}
void setCurrentrevision( const QString& s ) { m_currentrevision = s; }
void setTitle( const QString& s ) { m_title= s; }
void setInfo( const QString& s ) { m_info = s; }
void setCreator( const QString& s ) { m_creator = s; }
void setGuid( const QString& s ) { m_guid = s; }
void setShared( bool b ) { m_shared = b; }
// </IGNORE>
signals:
/// emitted when the playlist revision changes (whenever the playlist changes)
void revisionLoaded( Tomahawk::PlaylistRevision );
/// watch for this to see when newly created playlist is synced to DB (if you care)
void created();
public slots:
// want to update the playlist from the model?
// generate a newrev using uuid() and call this:
void createNewRevision( const QString& newrev, const QString& oldrev, const QList< plentry_ptr >& entries );
void reportCreated( const Tomahawk::playlist_ptr& self );
void reportDeleted( const Tomahawk::playlist_ptr& self );
void setRevision( const QString& rev,
const QList<QString>& neworderedguids,
const QList<QString>& oldorderedguids,
bool is_newest_rev,
const QMap< QString, Tomahawk::plentry_ptr >& addedmap,
bool applied );
void resolve();
private:
// called from loadAllPlaylists DB cmd:
explicit Playlist( const source_ptr& src,
const QString& currentrevision,
const QString& title,
const QString& info,
const QString& creator,
bool shared,
int lastmod,
const QString& guid = "" ); // populate db
// called when creating new playlist
explicit Playlist( const source_ptr& author,
const QString& guid,
const QString& title,
const QString& info,
const QString& creator,
bool shared );
void rundb();
source_ptr m_source;
QString m_currentrevision;
QString m_guid, m_title, m_info, m_creator;
unsigned int m_lastmodified;
bool m_shared;
QList< plentry_ptr > m_entries;
};
};
#endif // PLAYLIST_H

34
include/tomahawk/playlistmodelinterface.h Normal file
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#ifndef PLAYLISTINTERFACE_H
#define PLAYLISTINTERFACE_H
#include "playlistitem.h"
#include "tomahawk/collection.h"
#include "tomahawk/source.h"
class PlaylistModelInterface
{
public:
enum RepeatMode { NoRepeat, RepeatOne, RepeatAll };
virtual ~PlaylistModelInterface() {}
virtual PlaylistItem* previousItem() = 0;
virtual PlaylistItem* nextItem() = 0;
virtual PlaylistItem* siblingItem( int itemsAway ) = 0;
virtual void setCurrentItem( const QModelIndex& index ) = 0;
virtual unsigned int sourceCount() = 0;
virtual unsigned int collectionCount() = 0;
virtual unsigned int trackCount() = 0;
virtual unsigned int artistCount() = 0;
public slots:
virtual void setRepeatMode( RepeatMode mode ) = 0;
virtual void setShuffled( bool enabled ) = 0;
signals:
virtual void repeatModeChanged( PlaylistModelInterface::RepeatMode mode ) = 0;
virtual void shuffleModeChanged( bool enabled ) = 0;
};
#endif // PLAYLISTINTERFACE_H

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