Release Version 0.6.0.

Update german translation.
Fix Bug #1286 : typo.
2023-10-10 17:16:51 +00:00 · 2013-03-17 15:53:01 +01:00 · 2013-03-17 15:51:11 +01:00 · 2013-03-12 23:27:09 +01:00 · 2013-03-11 16:32:00 +01:00 · 2013-03-11 12:18:11 +01:00
19 changed files with 5351 additions and 1215 deletions
--- a/62
+++ b/62
@@ -1,3 +1,65 @@
+User johns
+Date: Sun Mar 17 15:52:42 CET 2013
+
+    Release Version 0.6.0
+    Adds H264 only hardware decoder for still-pictures.
+    Enable optional VDR-SPU deocder support.
+
+User anbr
+Date: Sun Mar 17 15:49:46 CET 2013
+
+    Update german translation.
+
+User cyril
+Date: Wed Mar 6 17:05:10 CET 2013
+
+    Adds raise softhddevice video window support.
+
+User johns
+Date: Wed Mar 6 10:30:27 CET 2013
+
+    Adds optional only complete mpeg packets support.
+    Fixes text of EAC-3 pass-through setup.
+    Try to start or connect to X11 server with -xx.
+    Try to use HBR (High Bit-Rate) for EAC3.
+    Improved pass-through (PCM+EAC3) support.
+    Support VDR 1.7.36 new build system.
+    Improves VDPAU display preemption handling.
+    Add modifiers to X11 remote key names (from Sibbi).
+    Add compatibility with >=ffmpeg 1.1.
+    Adds PIP (Picture-in-Picture) support.
+    Split mpeg packets in receiver thread.
+
+User horchi
+Date: Tue Jan  1 17:58:54 CET 2013
+
+    Adds VDR SeduAtmo Plugin support.
+
+User johns
+Date: Tue Jan  1 15:21:28 CET 2013
+
+    Support multiple streams with ScaleVideo.
+    Makes 4:3 and 16:9 display format configurable.
+    Don't use DVB display format.
+
+User Zoolook
+Date: Tue Jan  1 12:49:19 CET 2013
+
+    Add support for new vdr ScaleVideo API.
+
+User johns
+Date: Tue Jan  1 12:40:12 CET 2013
+
+    Add support for old PES HDTV recording.
+    Disable trickspeed hack, to prevent ffmpeg crash.
+    Makes X11 server arguments configurable.
+    Add german translation.
+
+User FireFly
+Date: Sun Nov 18 21:15:50 CET 2012
+
+    Add german translation.
+
 User johns
 Date: Thu Nov 15 22:28:55 CET 2012

--- a/218
+++ b/218
@@ -6,152 +6,183 @@
 # The official name of this plugin.
 # This name will be used in the '-P...' option of VDR to load the plugin.
 # By default the main source file also carries this name.
-# IMPORTANT: the presence of this macro is important for the Make.config
-# file. So it must be defined, even if it is not used here!
-#
+
 PLUGIN = softhddevice

+### Configuration (edit this for your needs)
+
+    # support alsa audio output module
+ALSA ?= $(shell pkg-config --exists alsa && echo 1)
+    # support OSS audio output module
+OSS ?= 1
+    # support VDPAU video output modue
+VDPAU ?= $(shell pkg-config --exists vdpau && echo 1)
+    # support VA-API video output modue
+VAAPI ?= $(shell pkg-config --exists libva && echo 1)
+    # screensaver disable/enable
+SCREENSAVER ?= 1
+    # use ffmpeg libswresample
+SWRESAMPLE ?= $(shell pkg-config --exists libswresample && echo 1)
+
+CONFIG := # -DDEBUG #-DOSD_DEBUG	# enable debug output+functions
+#CONFIG += -DSTILL_DEBUG=2		# still picture debug verbose level
+
+CONFIG += -DAV_INFO -DAV_INFO_TIME=3000	# info/debug a/v sync
+CONFIG += -DUSE_PIP			# PIP support
+#CONFIG += -DHAVE_PTHREAD_NAME		# supports new pthread_setname_np
+#CONFIG += -DNO_TS_AUDIO		# disable ts audio parser
+#CONFIG += -DUSE_TS_VIDEO		# build new ts video parser
+#CONFIG += -DUSE_MPEG_COMPLETE		# support only complete mpeg packets
+#CONFIG += -DUSE_VDR_SPU		# use VDR SPU decoder.
+
+ifeq ($(ALSA),1)
+CONFIG += -DUSE_ALSA
+_CFLAGS += $(shell pkg-config --cflags alsa)
+LIBS += $(shell pkg-config --libs alsa)
+endif
+ifeq ($(OSS),1)
+CONFIG += -DUSE_OSS
+endif
+ifeq ($(VDPAU),1)
+CONFIG += -DUSE_VDPAU
+_CFLAGS += $(shell pkg-config --cflags vdpau)
+LIBS += $(shell pkg-config --libs vdpau)
+endif
+ifeq ($(VAAPI),1)
+CONFIG += -DUSE_VAAPI
+_CFLAGS += $(shell pkg-config --cflags libva-x11 libva)
+LIBS += $(shell pkg-config --libs libva-x11 libva)
+endif
+ifeq ($(SCREENSAVER),1)
+CONFIG += -DUSE_SCREENSAVER
+_CFLAGS += $(shell pkg-config --cflags xcb-screensaver xcb-dpms)
+LIBS += $(shell pkg-config --libs xcb-screensaver xcb-dpms)
+endif
+ifeq ($(SWRESAMPLE),1)
+CONFIG += -DUSE_SWRESAMPLE
+_CFLAGS += $(shell pkg-config --cflags libswresample)
+LIBS += $(shell pkg-config --libs libswresample)
+endif
+
+_CFLAGS += $(shell pkg-config --cflags libavcodec x11 x11-xcb xcb xcb-icccm)
+LIBS += -lrt $(shell pkg-config --libs libavcodec x11 x11-xcb xcb xcb-icccm)
+
 ### The version number of this plugin (taken from the main source file):

 VERSION = $(shell grep 'static const char \*const VERSION *=' $(PLUGIN).cpp | awk '{ print $$7 }' | sed -e 's/[";]//g')
 GIT_REV = $(shell git describe --always 2>/dev/null)

-### Configuration (edit this for your needs)
-
-CONFIG := #-DDEBUG #-DOSD_DEBUG
-CONFIG += -DAV_INFO -DAV_INFO_TIME=3000	# debug a/v sync
-#CONFIG += -DHAVE_PTHREAD_NAME		# supports new pthread_setname_np
-#CONFIG += -DNO_TS_AUDIO		# disable ts audio parser
-#CONFIG += -DUSE_TS_VIDEO		# build new ts video parser
-CONFIG += $(shell pkg-config --exists vdpau && echo "-DUSE_VDPAU")
-CONFIG += $(shell pkg-config --exists libva && echo "-DUSE_VAAPI")
-CONFIG += $(shell pkg-config --exists alsa && echo "-DUSE_ALSA")
-CONFIG += -DUSE_OSS
-
-### The C++ compiler and options:
-
-CC	 ?= gcc
-CXX	 ?= g++
-CFLAGS	 ?=	-g -O2 -W -Wall -Wextra -Winit-self \
-		-Wdeclaration-after-statement \
-		-ftree-vectorize -msse3 -flax-vector-conversions
-CXXFLAGS ?= -g -O2 -W -Wall -Wextra -Werror=overloaded-virtual
-
 ### The directory environment:

-VDRDIR ?= ../../..
-LIBDIR ?= ../../lib
+# Use package data if installed...otherwise assume we're under the VDR source directory:
+PKGCFG = $(if $(VDRDIR),$(shell pkg-config --variable=$(1) $(VDRDIR)/vdr.pc),$(shell pkg-config --variable=$(1) vdr || pkg-config --variable=$(1) ../../../vdr.pc))
+LIBDIR = $(call PKGCFG,libdir)
+LOCDIR = $(call PKGCFG,locdir)
+PLGCFG = $(call PKGCFG,plgcfg)
+#
 TMPDIR ?= /tmp

-### Make sure that necessary options are included:
+### The compiler options:

-include $(VDRDIR)/Make.global
+export CFLAGS	= $(call PKGCFG,cflags)
+export CXXFLAGS = $(call PKGCFG,cxxflags)
+
+ifeq ($(CFLAGS),)
+$(error CFLAGS not set)
+endif
+ifeq ($(CXXFLAGS),)
+$(error CXXFLAGS not set)
+endif
+
+### The version number of VDR's plugin API:
+
+APIVERSION = $(call PKGCFG,apiversion)

 ### Allow user defined options to overwrite defaults:

-include $(VDRDIR)/Make.config
-
-### The version number of VDR's plugin API (taken from VDR's "config.h"):
-
-APIVERSION = $(shell sed -ne '/define APIVERSION/s/^.*"\(.*\)".*$$/\1/p' $(VDRDIR)/config.h)
+-include $(PLGCFG)

 ### The name of the distribution archive:

 ARCHIVE = $(PLUGIN)-$(VERSION)
 PACKAGE = vdr-$(ARCHIVE)

-### Includes, Defines and dependencies (add further entries here):
+### The name of the shared object file:

-INCLUDES += -I$(VDRDIR)/include
+SOFILE = libvdr-$(PLUGIN).so

-DEFINES += $(CONFIG) -D_GNU_SOURCE -DPLUGIN_NAME_I18N='"$(PLUGIN)"' \
+### Includes and Defines (add further entries here):
+
+INCLUDES +=
+
+DEFINES += -DPLUGIN_NAME_I18N='"$(PLUGIN)"' -D_GNU_SOURCE $(CONFIG) \
 	$(if $(GIT_REV), -DGIT_REV='"$(GIT_REV)"')

-_CFLAGS = $(DEFINES) $(INCLUDES) \
-	$(shell pkg-config --cflags libavcodec) \
-	`pkg-config --cflags x11 x11-xcb xcb xcb-xv xcb-shm xcb-dpms xcb-atom\
-		xcb-screensaver xcb-randr xcb-glx xcb-icccm xcb-keysyms`\
-	`pkg-config --cflags gl glu` \
-	$(if $(findstring USE_VDPAU,$(CONFIG)), \
-		    `pkg-config --cflags vdpau`) \
-	$(if $(findstring USE_VAAPI,$(CONFIG)), \
-		    `pkg-config --cflags libva-x11 libva-glx libva`) \
-	$(if $(findstring USE_ALSA,$(CONFIG)), \
-		    `pkg-config --cflags alsa`)
+### Make it standard

-#override _CFLAGS  += -Werror
-override CXXFLAGS += $(_CFLAGS)
-override CFLAGS	  += $(_CFLAGS)
-
-LIBS += -lrt \
-	$(shell pkg-config --libs libavcodec) \
-	`pkg-config --libs x11 x11-xcb xcb xcb-xv xcb-shm xcb-dpms xcb-atom\
-		xcb-screensaver xcb-randr xcb-glx xcb-icccm xcb-keysyms`\
-	`pkg-config --libs gl glu` \
-	$(if $(findstring USE_VDPAU,$(CONFIG)), \
-		    `pkg-config --libs vdpau`) \
-	$(if $(findstring USE_VAAPI,$(CONFIG)), \
-		    `pkg-config --libs libva-x11 libva-glx libva`) \
-	$(if $(findstring USE_ALSA,$(CONFIG)), \
-		    `pkg-config --libs alsa`)
+override CXXFLAGS += $(_CFLAGS) $(DEFINES) $(INCLUDES) \
+    -g -W -Wall -Wextra -Winit-self -Werror=overloaded-virtual
+override CFLAGS	  += $(_CFLAGS) $(DEFINES) $(INCLUDES) \
+    -g -W -Wall -Wextra -Winit-self -Wdeclaration-after-statement

 ### The object files (add further files here):

 OBJS = $(PLUGIN).o softhddev.o video.o audio.o codec.o ringbuffer.o
+
 SRCS = $(wildcard $(OBJS:.o=.c)) $(PLUGIN).cpp

 ### The main target:

-all: libvdr-$(PLUGIN).so i18n
-
-### Implicit rules:
-#
-#%.o: %.cpp
-#	$(CXX) $(CXXFLAGS) -c $(DEFINES) $(INCLUDES) $<
+all: $(SOFILE) i18n

 ### Dependencies:

-MAKEDEP = $(CC) -MM -MG
+MAKEDEP = $(CXX) -MM -MG
 DEPFILE = .dependencies
 $(DEPFILE): Makefile
-	@$(MAKEDEP) $(DEFINES) $(INCLUDES) $(SRCS) >$@
-
-$(OBJS): Makefile
+	@$(MAKEDEP) $(CXXFLAGS) $(SRCS) > $@

 -include $(DEPFILE)

 ### Internationalization (I18N):

 PODIR	  = po
-LOCALEDIR = $(VDRDIR)/locale
 I18Npo	  = $(wildcard $(PODIR)/*.po)
-I18Nmsgs  = $(addprefix $(LOCALEDIR)/, $(addsuffix /LC_MESSAGES/vdr-$(PLUGIN).mo, $(notdir $(foreach file, $(I18Npo), $(basename $(file))))))
+I18Nmo	  = $(addsuffix .mo, $(foreach file, $(I18Npo), $(basename $(file))))
+I18Nmsgs  = $(addprefix $(DESTDIR)$(LOCDIR)/, $(addsuffix /LC_MESSAGES/vdr-$(PLUGIN).mo, $(notdir $(foreach file, $(I18Npo), $(basename $(file))))))
 I18Npot	  = $(PODIR)/$(PLUGIN).pot

 %.mo: %.po
 	msgfmt -c -o $@ $<

-$(I18Npot): $(wildcard *.cpp)  $(wildcard *.c)
+$(I18Npot): $(SRCS)
 	xgettext -C -cTRANSLATORS --no-wrap --no-location -k -ktr -ktrNOOP \
-	-k_ -k_N --package-name=VDR --package-version=$(VDRVERSION) \
-	--msgid-bugs-address='<see README>' -o $@ $^
+	-k_ -k_N --package-name=vdr-$(PLUGIN) --package-version=$(VERSION) \
+	--msgid-bugs-address='<see README>' -o $@ `ls $^`

 %.po: $(I18Npot)
-	msgmerge -U --no-wrap --no-location --backup=none -q $@ $<
+	msgmerge -U --no-wrap --no-location --backup=none -q -N $@ $<
 	@touch $@

-$(I18Nmsgs): $(LOCALEDIR)/%/LC_MESSAGES/vdr-$(PLUGIN).mo: $(PODIR)/%.mo
-	@mkdir -p $(dir $@)
-	cp $< $@
+$(I18Nmsgs): $(DESTDIR)$(LOCDIR)/%/LC_MESSAGES/vdr-$(PLUGIN).mo: $(PODIR)/%.mo
+	install -D -m644 $< $@

 .PHONY: i18n
-i18n: $(I18Nmsgs) $(I18Npot)
+i18n: $(I18Nmo) $(I18Npot)
+
+install-i18n: $(I18Nmsgs)

 ### Targets:

-libvdr-$(PLUGIN).so: $(OBJS) Makefile
-	$(CXX) $(CXXFLAGS) $(LDFLAGS) -shared -fPIC $(OBJS) -o $@ $(LIBS)
-	@cp --remove-destination $@ $(LIBDIR)/$@.$(APIVERSION)
+$(OBJS): Makefile
+
+$(SOFILE): $(OBJS)
+	$(CXX) $(CXXFLAGS) $(LDFLAGS) -shared $(OBJS) $(LIBS) -o $@
+
+install-lib: $(SOFILE)
+	install -D $^ $(DESTDIR)$(LIBDIR)/$^.$(APIVERSION)
+
+install: install-lib install-i18n

 dist: $(I18Npo) clean
 	@-rm -rf $(TMPDIR)/$(ARCHIVE)
@@ -162,19 +193,20 @@ dist: $(I18Npo) clean
 	@echo Distribution package created as $(PACKAGE).tgz

 clean:
-	@-rm -f $(OBJS) $(DEPFILE) *.so *.tgz core* *~ $(PODIR)/*.mo $(PODIR)/*.pot
+	@-rm -f $(PODIR)/*.mo $(PODIR)/*.pot
+	@-rm -f $(OBJS) $(DEPFILE) *.so *.tgz core* *~

-install:	libvdr-$(PLUGIN).so
-	cp --remove-destination libvdr-$(PLUGIN).so \
-		/usr/lib/vdr/plugins/libvdr-$(PLUGIN).so.$(APIVERSION)
+## Private Targets:

 HDRS=	$(wildcard *.h)

 indent:
-	for i in $(wildcard $(OBJS:.o=.c)) $(HDRS); do \
-		indent $$i; unexpand -a $$i > $$i.up; mv $$i.up $$i; \
+	for i in $(SRCS) $(HDRS); do \
+		indent $$i; \
+		unexpand -a $$i | sed -e s/constconst/const/ > $$i.up; \
+		mv $$i.up $$i; \
 	done

 video_test: video.c Makefile
-	$(CC) -DVIDEO_TEST -DVERSION='"$(VERSION)"' $(CFLAGS) $(LDFLAGS) $< $(LIBS) \
-	-o $@
+	$(CC) -DVIDEO_TEST -DVERSION='"$(VERSION)"' $(CFLAGS) $(LDFLAGS) $< \
+	$(LIBS) -o $@
--- a/Makefile-pre1.7.36
+++ b/Makefile-pre1.7.36
@@ -0,0 +1,188 @@
+#
+# Makefile for a Video Disk Recorder plugin
+#
+# $Id$
+
+# The official name of this plugin.
+# This name will be used in the '-P...' option of VDR to load the plugin.
+# By default the main source file also carries this name.
+# IMPORTANT: the presence of this macro is important for the Make.config
+# file. So it must be defined, even if it is not used here!
+#
+PLUGIN = softhddevice
+
+### The version number of this plugin (taken from the main source file):
+
+VERSION = $(shell grep 'static const char \*const VERSION *=' $(PLUGIN).cpp | awk '{ print $$7 }' | sed -e 's/[";]//g')
+GIT_REV = $(shell git describe --always 2>/dev/null)
+
+### Configuration (edit this for your needs)
+
+CONFIG := #-DDEBUG #-DOSD_DEBUG
+CONFIG += -DAV_INFO -DAV_INFO_TIME=3000	# debug a/v sync
+CONFIG += -DUSE_PIP			# PIP support
+#CONFIG += -DHAVE_PTHREAD_NAME		# supports new pthread_setname_np
+#CONFIG += -DNO_TS_AUDIO		# disable ts audio parser
+#CONFIG += -DUSE_TS_VIDEO		# build new ts video parser
+					# use ffmpeg libswresample
+CONFIG += $(shell pkg-config --exists libswresample && echo "-DUSE_SWRESAMPLE")
+CONFIG += -DUSE_SCREENSAVER		# use functions to disable screensaver
+CONFIG += $(shell pkg-config --exists vdpau && echo "-DUSE_VDPAU")
+CONFIG += $(shell pkg-config --exists libva && echo "-DUSE_VAAPI")
+CONFIG += $(shell pkg-config --exists alsa && echo "-DUSE_ALSA")
+CONFIG += -DUSE_OSS
+
+### The C++ compiler and options:
+
+CC	 ?= gcc
+CXX	 ?= g++
+CFLAGS	 ?=	-g -O2 -W -Wall -Wextra -Winit-self \
+		-Wdeclaration-after-statement \
+		-ftree-vectorize -msse3 -flax-vector-conversions -fPIC
+CXXFLAGS ?= -g -O2 -W -Wall -Wextra -Werror=overloaded-virtual -fPIC
+
+### The directory environment:
+
+VDRDIR ?= ../../..
+LIBDIR ?= ../../lib
+TMPDIR ?= /tmp
+
+### Make sure that necessary options are included:
+
+-include $(VDRDIR)/Make.global
+
+### Allow user defined options to overwrite defaults:
+
+-include $(VDRDIR)/Make.config
+
+### The version number of VDR's plugin API (taken from VDR's "config.h"):
+
+APIVERSION = $(shell sed -ne '/define APIVERSION/s/^.*"\(.*\)".*$$/\1/p' $(VDRDIR)/config.h)
+
+### The name of the distribution archive:
+
+ARCHIVE = $(PLUGIN)-$(VERSION)
+PACKAGE = vdr-$(ARCHIVE)
+
+### Includes, Defines and dependencies (add further entries here):
+
+INCLUDES += -I$(VDRDIR)/include
+
+DEFINES += $(CONFIG) -D_GNU_SOURCE -DPLUGIN_NAME_I18N='"$(PLUGIN)"' \
+	$(if $(GIT_REV), -DGIT_REV='"$(GIT_REV)"')
+
+_CFLAGS = $(DEFINES) $(INCLUDES) \
+	$(shell pkg-config --cflags libavcodec) \
+	`pkg-config --cflags x11 x11-xcb xcb xcb-xv xcb-shm xcb-dpms xcb-atom\
+		xcb-screensaver xcb-randr xcb-glx xcb-icccm xcb-keysyms`\
+	`pkg-config --cflags gl glu` \
+	$(if $(findstring USE_SWRESAMPLE,$(CONFIG)), \
+		$(shell pkg-config --cflags libswresample)) \
+	$(if $(findstring USE_VAAPI,$(CONFIG)), \
+		`pkg-config --cflags libva-x11 libva-glx libva`) \
+	$(if $(findstring USE_ALSA,$(CONFIG)), \
+		`pkg-config --cflags alsa`)
+
+#override _CFLAGS  += -Werror
+override CXXFLAGS += $(_CFLAGS)
+override CFLAGS	  += $(_CFLAGS)
+
+LIBS += -lrt \
+	$(shell pkg-config --libs libavcodec) \
+	`pkg-config --libs x11 x11-xcb xcb xcb-xv xcb-shm xcb-dpms xcb-atom\
+		xcb-screensaver xcb-randr xcb-glx xcb-icccm xcb-keysyms`\
+	`pkg-config --libs gl glu` \
+	$(if $(findstring USE_SWRESAMPLE,$(CONFIG)), \
+		$(shell pkg-config --libs libswresample)) \
+	$(if $(findstring USE_VDPAU,$(CONFIG)), \
+		`pkg-config --libs vdpau`) \
+	$(if $(findstring USE_VAAPI,$(CONFIG)), \
+		`pkg-config --libs libva-x11 libva-glx libva`) \
+	$(if $(findstring USE_ALSA,$(CONFIG)), \
+		`pkg-config --libs alsa`)
+
+### The object files (add further files here):
+
+OBJS = $(PLUGIN).o softhddev.o video.o audio.o codec.o ringbuffer.o
+SRCS = $(wildcard $(OBJS:.o=.c)) $(PLUGIN).cpp
+
+### The main target:
+
+all: libvdr-$(PLUGIN).so i18n
+
+### Implicit rules:
+#
+#%.o: %.cpp
+#	$(CXX) $(CXXFLAGS) -c $(DEFINES) $(INCLUDES) $<
+
+### Dependencies:
+
+MAKEDEP = $(CC) -MM -MG
+DEPFILE = .dependencies
+$(DEPFILE): Makefile
+	@$(MAKEDEP) $(DEFINES) $(INCLUDES) $(SRCS) >$@
+
+$(OBJS): Makefile
+
+-include $(DEPFILE)
+
+### Internationalization (I18N):
+
+PODIR	  = po
+LOCALEDIR = $(VDRDIR)/locale
+I18Npo	  = $(wildcard $(PODIR)/*.po)
+I18Nmsgs  = $(addprefix $(LOCALEDIR)/, $(addsuffix /LC_MESSAGES/vdr-$(PLUGIN).mo, $(notdir $(foreach file, $(I18Npo), $(basename $(file))))))
+I18Npot	  = $(PODIR)/$(PLUGIN).pot
+
+%.mo: %.po
+	msgfmt -c -o $@ $<
+
+$(I18Npot): $(wildcard *.cpp) $(wildcard *.c)
+	xgettext -C -cTRANSLATORS --no-wrap --no-location -k -ktr -ktrNOOP \
+	-k_ -k_N --package-name=VDR --package-version=$(VDRVERSION) \
+	--msgid-bugs-address='<see README>' -o $@ $^
+
+%.po: $(I18Npot)
+	msgmerge -U --no-wrap --no-location --backup=none -q $@ $<
+	@touch $@
+
+$(I18Nmsgs): $(LOCALEDIR)/%/LC_MESSAGES/vdr-$(PLUGIN).mo: $(PODIR)/%.mo
+	@mkdir -p $(dir $@)
+	cp $< $@
+
+.PHONY: i18n
+i18n: $(I18Nmsgs) $(I18Npot)
+
+### Targets:
+
+libvdr-$(PLUGIN).so: $(OBJS) Makefile
+	$(CXX) $(CXXFLAGS) $(LDFLAGS) -shared -fPIC $(OBJS) -o $@ $(LIBS)
+	@cp --remove-destination $@ $(LIBDIR)/$@.$(APIVERSION)
+
+dist: $(I18Npo) clean
+	@-rm -rf $(TMPDIR)/$(ARCHIVE)
+	@mkdir $(TMPDIR)/$(ARCHIVE)
+	@cp -a * $(TMPDIR)/$(ARCHIVE)
+	@tar czf $(PACKAGE).tgz -C $(TMPDIR) $(ARCHIVE)
+	@-rm -rf $(TMPDIR)/$(ARCHIVE)
+	@echo Distribution package created as $(PACKAGE).tgz
+
+clean:
+	@-rm -f $(OBJS) $(DEPFILE) *.so *.tgz core* *~ $(PODIR)/*.mo $(PODIR)/*.pot
+
+install:	libvdr-$(PLUGIN).so
+	cp --remove-destination libvdr-$(PLUGIN).so \
+		/usr/lib/vdr/plugins/libvdr-$(PLUGIN).so.$(APIVERSION)
+
+HDRS=	$(wildcard *.h)
+
+indent:
+	for i in $(SRCS) $(HDRS); do \
+		indent $$i; \
+		unexpand -a $$i | sed -e s/constconst/const/ > $$i.up; \
+		mv $$i.up $$i; \
+	done
+
+video_test: video.c Makefile
+	$(CC) -DVIDEO_TEST -DVERSION='"$(VERSION)"' $(CFLAGS) $(LDFLAGS) $< $(LIBS) \
+	-o $@
--- a/README.txt
+++ b/README.txt
@@ -1,6 +1,6 @@
@file README.txt		@brief A software HD output device for VDR

-Copyright (c) 2011, 2012 by Johns.  All Rights Reserved.
+Copyright (c) 2011 - 2013 by Johns.  All Rights Reserved.

 Contributor(s):

@@ -27,13 +27,14 @@ A software and GPU emulated HD output device plugin for VDR.
    o Audio FFMpeg / OSS / Analog
    o HDMI/SPDIF pass-through
    o Software volume, compression, normalize and channel resample
-    o YaepgHD support
+    o YaepgHD support / new >1.7.33 VDR ScaleVideo API support
    o Software deinterlacer Bob (VA-API only)
    o Autocrop
    o Grab image (VDPAU only)
-    o Suspend
+    o Suspend / Dettach
    o Letterbox, Stretch and Center cut-out video display modes
    o atmo light support with plugin http://github.com/durchflieger/DFAtmo
+    o PIP (Picture-in-Picture) (VDPAU only)

    o planned: Video decoder VA-API Branch: vaapi-ext/staging
    o planned: Video output XvBA / Opengl / Xv
@@ -56,8 +57,8 @@ Install:

 	git clone git://projects.vdr-developer.org/vdr-plugin-softhddevice.git
 	cd vdr-plugin-softhddevice
-	make VDRDIR=<path-to-your-vdr-files> LIBDIR=.
-	gentoo: make VDRDIR=/usr/include/vdr LIBDIR=.
+	make
+	make install

 	2a) tarball

@@ -66,10 +67,11 @@ Install:

 	tar vxf vdr-softhddevice-*.tar.bz2
 	cd softhddevice-*
-	make VDRDIR=<path-to-your-vdr-files> LIBDIR=.
+	make
+	make install

 	You can edit Makefile to enable/disable VDPAU / VA-API / Alsa / OSS
-	support.
+	support.  The default is to autodetect as much as possible.

 Setup:	environment
 ------
@@ -87,8 +89,8 @@ Setup:	environment
    only if alsa is configured
 	ALSA_DEVICE=default
 		alsa PCM device name
-	ALSA_AC3_DEVICE=
-		alsa AC3/pass-though device name
+	ALSA_PASSTHROUGH_DEVICE=
+		alsa pass-though (AC3,EAC3,DTS,...) device name
 	ALSA_MIXER=default
 		alsa control device name
 	ALSA_MIXER_CHANNEL=PCM
@@ -97,8 +99,8 @@ Setup:	environment
    only if oss is configured
 	OSS_AUDIODEV=/dev/dsp
 		oss dsp device name
-	OSS_AC3_AUDIODEV=
-		oss AC3/pass-though device name
+	OSS_PASSTHROUGHDEV=
+		oss pass-though (AC3,EAC3,DTS,...) device name
 	OSS_MIXERDEV=/dev/mixer
 		oss mixer device name
 	OSS_MIXER_CHANNEL=pcm
@@ -154,13 +156,17 @@ Setup: /etc/vdr/setup.conf
 	delay audio or delay video

 	softhddevice.AudioPassthrough = 0
-	0 = none, 1 = AC-3
+	0 = none, 1 = PCM, 2 = MPA, 4 = AC-3, 8 = EAC-3, -X disable

-	for AC-3 the pass-through device is used.
+	for PCM/AC-3/EAC-3 the pass-through device is used and the audio
+	stream is passed undecoded to the output device.
+	z.b. 12 = AC-3+EAC-3, 13 = PCM+AC-3+EAC-3
+	note: MPA/DTS/TrueHD/... aren't supported yet
+	negative values disable passthrough

 	softhddevice.AudioDownmix = 0
 	0 = none, 1 = downmix
-	downmix AC-3 to stero.
+	Use ffmpeg/libav downmix of AC-3/EAC-3 audio to stereo.

 	softhddevice.AudioSoftvol = 0
 	0 = off, use hardware volume control
@@ -228,11 +234,41 @@ Setup: /etc/vdr/setup.conf
 	0 disable black picture during channel switch
 	1 enable black picture during channel switch

-	VideoDisplayFormat = ?
+	softhddevice.Video4to3DisplayFormat = 1
 	0 pan and scan
 	1 letter box
 	2 center cut-out

+	softhddevice.VideoOtherDisplayFormat = 1
+	0 pan and scan
+	1 pillar box
+	2 center cut-out
+
+	softhddevice.pip.X = 79
+	softhddevice.pip.Y = 78
+	softhddevice.pip.Width = 18
+	softhddevice.pip.Height = 18
+	PIP pip window position and size in percent.
+
+	softhddevice.pip.VideoX = 0
+	softhddevice.pip.VideoY = 0
+	softhddevice.pip.VideoWidth = 0
+	softhddevice.pip.VideoHeight = 0
+	PIP video window position and size in percent.
+
+	softhddevice.pip.Alt.X = 0
+	softhddevice.pip.Alt.Y = 50
+	softhddevice.pip.Alt.Width = 0
+	softhddevice.pip.Alt.Height = 50
+	PIP alternative pip window position and size in percent.
+
+	softhddevice.pip.Alt.VideoX = 0
+	softhddevice.pip.Alt.VideoY = 0
+	softhddevice.pip.Alt.VideoWidth = 0
+	softhddevice.pip.Alt.VideoHeight = 50
+	PIP alternative video window position and size in percent.
+
+
 Setup: /etc/vdr/remote.conf
 ------

@@ -305,11 +341,16 @@ Warning:

 Known Bugs:
 -----------
-	VA-API doesn't v-sync 1080i streams
+	VA-API doesn't v-sync h264 interlaced streams
 	vdr-image not working

 Requires:
 ---------
+	media-video/vdr (version >=1.7.xx)
+		Video Disk Recorder - turns a pc into a powerful set top box
+		for DVB.
+		http://www.tvdr.de/
+
 	media-video/ffmpeg (version >=0.7)
 		Complete solution to record, convert and stream audio and
 		video. Includes libavcodec.
--- a/10
+++ b/10
@@ -39,6 +39,7 @@ video:
    some low-bandwidth tv channels have hiccups.
    check start with 24Hz display rate
    crash with ffmpeg without vaapi and vdpau.
+    still-picture of PES recordings should use VideoMpegEnqueue.

 vdpau:
    software deinterlace path not working.
@@ -54,6 +55,8 @@ libva:
    [drm:i915_hangcheck_elapsed] *ERROR* Hangcheck timer elapsed... GPU hung
    [drm:i915_wait_request] *ERROR* i915_wait_request returns -11 ...
    missing OSD support for 3d SBS / Top-Bottom streams, like VPDAU.
+    PIP support / multistream handling
+    VA-AP VaapiCleanup crash after channel without video.

 libva: branch vaapi-ext / staging
    add support for vaapi-ext / staging
@@ -110,23 +113,20 @@ plugins:
 setup:
    Setup of decoder type.
    Setup of output type.
-    Setup of display type.
-    Setup 4:3 zoom type
    Some setup parameters are not used until restart.
 	Can a notice be added to the setup menu?

 unsorted:
    stoping vdr while plugin is suspended opens and closes a window.
    svdrp prim: support plugin names for device numbers.
+    Workaround exists: hangup PipVideoStream -> Vdpau_get_format -> xcb -> poll
+	+ lock DecoderLockMutex

 future features (not planed for 1.0 - 1.5)

    video out with xv
    video out with opengl
-    video out with xvba
    software decoder for xv / opengl
-    multistream handling
-    pip support
    save and use auto-crop with channel zapping

    upmix stereo to AC-3 (supported by alsa plugin)
--- a/audio.c
+++ b/audio.c
@@ -1,7 +1,7 @@
 ///
 ///	@file audio.c		@brief Audio module
 ///
-///	Copyright (c) 2009 - 2012 by Johns.  All Rights Reserved.
+///	Copyright (c) 2009 - 2013 by Johns.  All Rights Reserved.
 ///
 ///	Contributor(s):
 ///
@@ -40,6 +40,7 @@
 //#define USE_ALSA			///< enable alsa support
 //#define USE_OSS			///< enable OSS support
 #define USE_AUDIO_THREAD		///< use thread for audio playback
+#define USE_AUDIO_MIXER			///< use audio module mixer

 #include <stdio.h>
 #include <stdint.h>
@@ -128,7 +129,7 @@ static const char *AudioModuleName;	///< which audio module to use
    /// Selected audio module.
 static const AudioModule *AudioUsedModule = &NoopModule;
 static const char *AudioPCMDevice;	///< PCM device name
-static const char *AudioAC3Device;	///< AC3 device name
+static const char *AudioPassthroughDevice;	///< Passthrough device name
 static const char *AudioMixerDevice;	///< mixer device name
 static const char *AudioMixerChannel;	///< mixer channel name
 static char AudioDoingInit;		///> flag in init, reduce error
@@ -163,7 +164,6 @@ static int AudioStereoDescent;		///< volume descent for stereo
 static int AudioVolume;			///< current volume (0 .. 1000)

 extern int VideoAudioDelay;		///< import audio/video delay
-extern int VideoGetBuffers(void);	///< Get number of input buffers.

    /// default ring buffer size ~2s 8ch 16bit (3 * 5 * 7 * 8)
 static const unsigned AudioRingBufferSize = 3 * 5 * 7 * 8 * 2 * 1000;
@@ -178,7 +178,7 @@ enum _audio_rates
    //Audio88200,				///< 88.2Khz
    //Audio96000,				///< 96.0Khz
    //Audio176400,				///< 176.4Khz
-    //Audio192000,				///< 192.0Khz
+    Audio192000,			///< 192.0Khz
    AudioRatesMax			///< max index
 };

@@ -188,9 +188,9 @@ static int AudioRatesInHw[AudioRatesMax];
    /// input to hardware channel matrix
 static int AudioChannelMatrix[AudioRatesMax][9];

-    /// rates tables
+    /// rates tables (must be sorted by frequency)
 static const unsigned AudioRatesTable[AudioRatesMax] = {
-    44100, 48000,
+    44100, 48000, 192000
 };

 //----------------------------------------------------------------------------
@@ -401,6 +401,8 @@ static void AudioSoftAmplifier(int16_t * samples, int count)
    }
 }

+#ifdef USE_AUDIO_MIXER
+
 /**
 **	Upmix mono to stereo.
 **
@@ -547,6 +549,13 @@ static void AudioUpmix(const int16_t * in, int in_chan, int frames,
 /**
 **	Resample ffmpeg sample format to hardware format.
 **
+**	FIXME: use libswresample for this and move it to codec.
+**	FIXME: ffmpeg to alsa conversion is already done in codec.c.
+**
+**	ffmpeg L  R  C	Ls Rs		-> alsa L R  Ls Rs C
+**	ffmpeg L  R  C	LFE Ls Rs	-> alsa L R  Ls Rs C  LFE
+**	ffmpeg L  R  C	LFE Ls Rs Rl Rr	-> alsa L R  Ls Rs C  LFE Rl Rr
+**
 **	@param in	input sample buffer
 **	@param in_chan	nr. of input channels
 **	@param frames	number of frames in sample buffer
@@ -582,6 +591,9 @@ static void AudioResample(const int16_t * in, int in_chan, int frames,
 	    AudioSurround2Stereo(in, in_chan, frames, out);
 	    break;
 	case 5 * 8 + 6:
+	case 3 * 8 + 8:
+	case 5 * 8 + 8:
+	case 6 * 8 + 8:
 	    AudioUpmix(in, in_chan, frames, out, out_chan);
 	    break;

@@ -594,6 +606,8 @@ static void AudioResample(const int16_t * in, int in_chan, int frames,
    }
 }

+#endif
+
 //----------------------------------------------------------------------------
 //	ring buffer
 //----------------------------------------------------------------------------
@@ -606,7 +620,7 @@ static void AudioResample(const int16_t * in, int in_chan, int frames,
 typedef struct _audio_ring_ring_
 {
    char FlushBuffers;			///< flag: flush buffers
-    char UseAc3;			///< flag: use ac3 pass-through
+    char Passthrough;			///< flag: use pass-through (AC3, ...)
    int16_t PacketSize;			///< packet size
    unsigned HwSampleRate;		///< hardware sample rate in Hz
    unsigned HwChannels;		///< hardware number of channels
@@ -624,29 +638,34 @@ static atomic_t AudioRingFilled;	///< how many of the ring is used
 static unsigned AudioStartThreshold;	///< start play, if filled

 /**
-**	Add sample-rate, number of channel change to ring.
+**	Add sample-rate, number of channels change to ring.
 **
 **	@param sample_rate	sample-rate frequency
 **	@param channels		number of channels
-**	@param use_ac3		use ac3/pass-through device
+**	@param passthrough	use /pass-through (AC3, ...) device
 **
 **	@retval -1	error
 **	@retval 0	okay
+**
+**	@note this function shouldn't fail.  Checks are done during AudoInit.
 */
-static int AudioRingAdd(unsigned sample_rate, int channels, int use_ac3)
+static int AudioRingAdd(unsigned sample_rate, int channels, int passthrough)
 {
    unsigned u;

    // search supported sample-rates
    for (u = 0; u < AudioRatesMax; ++u) {
 	if (AudioRatesTable[u] == sample_rate) {
+	    goto found;
+	}
+	if (AudioRatesTable[u] > sample_rate) {
 	    break;
 	}
    }
-    if (u == AudioRatesMax) {		// unsupported sample-rate
-	Error(_("audio: %dHz sample-rate unsupported\n"), sample_rate);
-	return -1;
-    }
+    Error(_("audio: %dHz sample-rate unsupported\n"), sample_rate);
+    return -1;				// unsupported sample-rate
+
+  found:
    if (!AudioChannelMatrix[u][channels]) {
 	Error(_("audio: %d channels unsupported\n"), channels);
 	return -1;			// unsupported nr. of channels
@@ -661,7 +680,7 @@ static int AudioRingAdd(unsigned sample_rate, int channels, int use_ac3)

    // FIXME: don't flush buffers here
    AudioRing[AudioRingWrite].FlushBuffers = 1;
-    AudioRing[AudioRingWrite].UseAc3 = use_ac3;
+    AudioRing[AudioRingWrite].Passthrough = passthrough;
    AudioRing[AudioRingWrite].PacketSize = 0;
    AudioRing[AudioRingWrite].InSampleRate = sample_rate;
    AudioRing[AudioRingWrite].InChannels = channels;
@@ -817,8 +836,9 @@ static int AlsaPlayRingbuffer(void)
 	if (!avail) {			// full or buffer empty
 	    break;
 	}
-	// muting ac3, can produce disturbance
-	if (AudioMute || (AudioSoftVolume && !AudioRing[AudioRingRead].UseAc3)) {
+	// muting pass-through ac3, can produce disturbance
+	if (AudioMute || (AudioSoftVolume
+		&& !AudioRing[AudioRingRead].Passthrough)) {
 	    // FIXME: quick&dirty cast
 	    AudioSoftAmplifier((int16_t *) p, avail);
 	    // FIXME: if not all are written, we double amplify them
@@ -965,23 +985,23 @@ static int AlsaThread(void)
 /**
 **	Open alsa pcm device.
 **
-**	@param use_ac3	use ac3/pass-through device
+**	@param passthrough	use pass-through (AC3, ...) device
 */
-static snd_pcm_t *AlsaOpenPCM(int use_ac3)
+static snd_pcm_t *AlsaOpenPCM(int passthrough)
 {
    const char *device;
    snd_pcm_t *handle;
    int err;

    // &&|| hell
-    if (!(use_ac3 && ((device = AudioAC3Device)
-		|| (device = getenv("ALSA_AC3_DEVICE"))))
+    if (!(passthrough && ((device = AudioPassthroughDevice)
+		|| (device = getenv("ALSA_PASSTHROUGH_DEVICE"))))
 	&& !(device = AudioPCMDevice) && !(device = getenv("ALSA_DEVICE"))) {
 	device = "default";
    }
    if (!AudioDoingInit) {		// reduce blabla during init
-	Info(_("audio/alsa: using %sdevice '%s'\n"), use_ac3 ? "ac3 " : "",
-	    device);
+	Info(_("audio/alsa: using %sdevice '%s'\n"),
+	    passthrough ? "pass-through " : "", device);
    }
    // open none blocking; if device is already used, we don't want wait
    if ((err =
@@ -1148,7 +1168,7 @@ static int64_t AlsaGetDelay(void)
 **
 **	@param freq		sample frequency
 **	@param channels		number of channels
-**	@param use_ac3		use ac3/pass-through device
+**	@param passthrough	use pass-through (AC3, ...) device
 **
 **	@retval 0	everything ok
 **	@retval 1	didn't support frequency/channels combination
@@ -1156,7 +1176,7 @@ static int64_t AlsaGetDelay(void)
 **
 **	@todo FIXME: remove pointer for freq + channels
 */
-static int AlsaSetup(int *freq, int *channels, int use_ac3)
+static int AlsaSetup(int *freq, int *channels, int passthrough)
 {
    snd_pcm_uframes_t buffer_size;
    snd_pcm_uframes_t period_size;
@@ -1164,7 +1184,7 @@ static int AlsaSetup(int *freq, int *channels, int use_ac3)
    int delay;

    if (!AlsaPCMHandle) {		// alsa not running yet
-	// FIXME: if open fails for ac3, we never recover
+	// FIXME: if open fails for fe. pass-through, we never recover
 	return -1;
    }
    if (1) {				// close+open to fix HDMI no sound bug
@@ -1174,7 +1194,7 @@ static int AlsaSetup(int *freq, int *channels, int use_ac3)
 	// FIXME: need lock
 	AlsaPCMHandle = NULL;		// other threads should check handle
 	snd_pcm_close(handle);
-	if (!(handle = AlsaOpenPCM(use_ac3))) {
+	if (!(handle = AlsaOpenPCM(passthrough))) {
 	    return -1;
 	}
 	AlsaPCMHandle = handle;
@@ -1434,7 +1454,7 @@ static int OssPlayRingbuffer(void)
 	    break;			// bi.bytes could become negative!
 	}

-	if (AudioSoftVolume && !AudioRing[AudioRingRead].UseAc3) {
+	if (AudioSoftVolume && !AudioRing[AudioRingRead].Passthrough) {
 	    // FIXME: quick&dirty cast
 	    AudioSoftAmplifier((int16_t *) p, bi.bytes);
 	    // FIXME: if not all are written, we double amplify them
@@ -1541,22 +1561,22 @@ static int OssThread(void)
 /**
 **	Open OSS pcm device.
 **
-**	@param use_ac3	use ac3/pass-through device
+**	@param passthrough	use pass-through (AC3, ...) device
 */
-static int OssOpenPCM(int use_ac3)
+static int OssOpenPCM(int passthrough)
 {
    const char *device;
    int fildes;

    // &&|| hell
-    if (!(use_ac3 && ((device = AudioAC3Device)
-		|| (device = getenv("OSS_AC3_AUDIODEV"))))
+    if (!(passthrough && ((device = AudioPassthroughDevice)
+		|| (device = getenv("OSS_PASSTHROUGHDEV"))))
 	&& !(device = AudioPCMDevice) && !(device = getenv("OSS_AUDIODEV"))) {
 	device = "/dev/dsp";
    }
    if (!AudioDoingInit) {
-	Info(_("audio/oss: using %sdevice '%s'\n"), use_ac3 ? "ac3 " : "",
-	    device);
+	Info(_("audio/oss: using %sdevice '%s'\n"),
+	    passthrough ? "pass-through " : "", device);
    }

    if ((fildes = open(device, O_WRONLY)) < 0) {
@@ -1703,15 +1723,15 @@ static int64_t OssGetDelay(void)
 /**
 **	Setup OSS audio for requested format.
 **
-**	@param sample_rate		sample rate/frequency
+**	@param sample_rate	sample rate/frequency
 **	@param channels		number of channels
-**	@param use_ac3		use ac3/pass-through device
+**	@param passthrough	use pass-through (AC3, ...) device
 **
 **	@retval 0	everything ok
 **	@retval 1	didn't support frequency/channels combination
 **	@retval -1	something gone wrong
 */
-static int OssSetup(int *sample_rate, int *channels, int use_ac3)
+static int OssSetup(int *sample_rate, int *channels, int passthrough)
 {
    int ret;
    int tmp;
@@ -1719,7 +1739,7 @@ static int OssSetup(int *sample_rate, int *channels, int use_ac3)
    audio_buf_info bi;

    if (OssPcmFildes == -1) {		// OSS not ready
-	// FIXME: if open fails for ac3, we never recover
+	// FIXME: if open fails for fe. pass-through, we never recover
 	return -1;
    }

@@ -1729,7 +1749,7 @@ static int OssSetup(int *sample_rate, int *channels, int use_ac3)
 	fildes = OssPcmFildes;
 	OssPcmFildes = -1;
 	close(fildes);
-	if (!(fildes = OssOpenPCM(use_ac3))) {
+	if (!(fildes = OssOpenPCM(passthrough))) {
 	    return -1;
 	}
 	OssPcmFildes = fildes;
@@ -1910,13 +1930,14 @@ static void NoopSetVolume( __attribute__ ((unused))
 /**
 **	Noop setup.
 **
-**	@param freq	sample frequency
-**	@param channels	number of channels
+**	@param freq		sample frequency
+**	@param channels		number of channels
+**	@param passthrough	use pass-through (AC3, ...) device
 */
 static int NoopSetup( __attribute__ ((unused))
    int *channels, __attribute__ ((unused))
    int *freq, __attribute__ ((unused))
-    int use_ac3)
+    int passthrough)
 {
    return -1;
 }
@@ -1954,16 +1975,17 @@ static const AudioModule NoopModule = {
 */
 static int AudioNextRing(void)
 {
-    int use_ac3;
+    int passthrough;
    int sample_rate;
    int channels;
+    size_t used;

    // update audio format
    // not always needed, but check if needed is too complex
-    use_ac3 = AudioRing[AudioRingRead].UseAc3;
+    passthrough = AudioRing[AudioRingRead].Passthrough;
    sample_rate = AudioRing[AudioRingRead].HwSampleRate;
    channels = AudioRing[AudioRingRead].HwChannels;
-    if (AudioUsedModule->Setup(&sample_rate, &channels, use_ac3)) {
+    if (AudioUsedModule->Setup(&sample_rate, &channels, passthrough)) {
 	Error(_("audio: can't set channels %d sample-rate %dHz\n"), channels,
 	    sample_rate);
 	// FIXME: handle error
@@ -1976,12 +1998,18 @@ static int AudioNextRing(void)
    AudioResetCompressor();
    AudioResetNormalizer();

+    Debug(3, "audio: a/v next buf(%d,%4zdms)\n", atomic_read(&AudioRingFilled),
+	(RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer) * 1000)
+	/ (AudioRing[AudioRingWrite].HwSampleRate *
+	    AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample));
+
    // stop, if not enough in next buffer
-    if (AudioStartThreshold >=
-	RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer)) {
-	return 1;
+    used = RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer);
+    if (AudioStartThreshold * 4 < used || (AudioVideoIsReady
+	    && AudioStartThreshold < used)) {
+	return 0;
    }
-    return 0;
+    return 1;
 }

 /**
@@ -2030,19 +2058,22 @@ static void *AudioPlayHandlerThread(void *dummy)
 	    }

 	    if (flush) {
+		Debug(3, "audio: flush\n");
 		AudioUsedModule->FlushBuffers();
 		if (AudioNextRing()) {
+		    Debug(3, "audio: break after flush\n");
 		    break;
 		}
+		Debug(3, "audio: continue after flush\n");
 	    }
 	    // try to play some samples
 	    err = AudioUsedModule->Thread();
 	    // underrun, check if new ring buffer is available
 	    if (!err) {
-		int use_ac3;
+		int passthrough;
 		int sample_rate;
 		int channels;
-		int old_use_ac3;
+		int old_passthrough;
 		int old_sample_rate;
 		int old_channels;

@@ -2052,20 +2083,21 @@ static void *AudioPlayHandlerThread(void *dummy)
 		}

 		Debug(3, "audio: next ring buffer\n");
-		old_use_ac3 = AudioRing[AudioRingRead].UseAc3;
+		old_passthrough = AudioRing[AudioRingRead].Passthrough;
 		old_sample_rate = AudioRing[AudioRingRead].HwSampleRate;
 		old_channels = AudioRing[AudioRingRead].HwChannels;

 		atomic_dec(&AudioRingFilled);
 		AudioRingRead = (AudioRingRead + 1) % AUDIO_RING_MAX;

-		use_ac3 = AudioRing[AudioRingRead].UseAc3;
+		passthrough = AudioRing[AudioRingRead].Passthrough;
 		sample_rate = AudioRing[AudioRingRead].HwSampleRate;
 		channels = AudioRing[AudioRingRead].HwChannels;
 		Debug(3, "audio: thread channels %d frequency %dHz %s\n",
-		    channels, sample_rate, use_ac3 ? "ac3" : "pcm");
+		    channels, sample_rate, passthrough ? "pass-through" : "");
 		// audio config changed?
-		if (old_use_ac3 != use_ac3 || old_sample_rate != sample_rate
+		if (old_passthrough != passthrough
+		    || old_sample_rate != sample_rate
 		    || old_channels != channels) {
 		    // FIXME: wait for buffer drain
 		    if (AudioNextRing()) {
@@ -2144,7 +2176,6 @@ void AudioEnqueue(const void *samples, int count)
 {
    size_t n;
    int16_t *buffer;
-    int frames;

 #ifdef noDEBUG
    static uint32_t last_tick;
@@ -2166,28 +2197,40 @@ void AudioEnqueue(const void *samples, int count)
 	AudioRing[AudioRingWrite].PacketSize = count;
 	Debug(3, "audio: a/v packet size %d bytes\n", count);
    }
-    if (AudioRing[AudioRingWrite].UseAc3) {
-	buffer = (void *)samples;
-    } else {
-	//
-	//	Convert / resample input to hardware format
-	//
+    // audio sample modification allowed and needed?
+    buffer = (void *)samples;
+    if (!AudioRing[AudioRingWrite].Passthrough && (AudioCompression
+	    || AudioNormalize
+	    || AudioRing[AudioRingWrite].InChannels !=
+	    AudioRing[AudioRingWrite].HwChannels)) {
+	int frames;
+
+	// resample into ring-buffer is too complex in the case of a roundabout
+	// just use a temporary buffer
 	frames =
 	    count / (AudioRing[AudioRingWrite].InChannels *
 	    AudioBytesProSample);
 	buffer =
 	    alloca(frames * AudioRing[AudioRingWrite].HwChannels *
 	    AudioBytesProSample);
+#ifdef USE_AUDIO_MIXER
+	// Convert / resample input to hardware format
 	AudioResample(samples, AudioRing[AudioRingWrite].InChannels, frames,
 	    buffer, AudioRing[AudioRingWrite].HwChannels);
-
+#else
+#ifdef DEBUG
+	if (AudioRing[AudioRingWrite].InChannels !=
+	    AudioRing[AudioRingWrite].HwChannels) {
+	    Debug(3, "audio: internal failure channels mismatch\n");
+	    return;
+	}
+#endif
+	memcpy(buffer, samples, count);
+#endif
 	count =
 	    frames * AudioRing[AudioRingWrite].HwChannels *
 	    AudioBytesProSample;

-	// resample into ring-buffer is too complex in the case of a roundabout
-	// just use a temporary buffer
-
 	if (AudioCompression) {		// in place operation
 	    AudioCompressor(buffer, count);
 	}
@@ -2276,11 +2319,12 @@ void AudioVideoReady(int64_t pts)
 	(used * 90 * 1000) / (AudioRing[AudioRingWrite].HwSampleRate *
 	AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample);

-    Debug(3, "audio: a/v buf:%4zdms %s|%s = %dms video ready\n",
+    Debug(3, "audio: a/v sync buf(%d,%4zdms) %s|%s = %dms %s\n",
+	atomic_read(&AudioRingFilled),
 	(used * 1000) / (AudioRing[AudioRingWrite].HwSampleRate *
 	    AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample),
-	Timestamp2String(audio_pts), Timestamp2String(pts),
-	(int)(pts - audio_pts) / 90);
+	Timestamp2String(pts), Timestamp2String(audio_pts),
+	(int)(pts - audio_pts) / 90, AudioRunning ? "running" : "ready");

    if (!AudioRunning) {
 	int skip;
@@ -2291,7 +2335,7 @@ void AudioVideoReady(int64_t pts)
 	    pts - 15 * 20 * 90 - AudioBufferTime * 90 - audio_pts +
 	    VideoAudioDelay;
 #ifdef DEBUG
-	printf("%dms %dms %dms\n", (int)(pts - audio_pts) / 90,
+	fprintf(stderr, "%dms %dms %dms\n", (int)(pts - audio_pts) / 90,
 	    VideoAudioDelay / 90, skip / 90);
 #endif
 	// guard against old PTS
@@ -2304,7 +2348,7 @@ void AudioVideoReady(int64_t pts)
 		AudioSkip = skip - used;
 		skip = used;
 	    }
-	    Debug(3, "audio: advance %dms %d/%zd\n",
+	    Debug(3, "audio: sync advance %dms %d/%zd\n",
 		(skip * 1000) / (AudioRing[AudioRingWrite].HwSampleRate *
 		    AudioRing[AudioRingWrite].HwChannels *
 		    AudioBytesProSample), skip, used);
@@ -2375,7 +2419,7 @@ void AudioFlushBuffers(void)
    old = AudioRingWrite;
    AudioRingWrite = (AudioRingWrite + 1) % AUDIO_RING_MAX;
    AudioRing[AudioRingWrite].FlushBuffers = 1;
-    AudioRing[AudioRingWrite].UseAc3 = AudioRing[old].UseAc3;
+    AudioRing[AudioRingWrite].Passthrough = AudioRing[old].Passthrough;
    AudioRing[AudioRingWrite].HwSampleRate = AudioRing[old].HwSampleRate;
    AudioRing[AudioRingWrite].HwChannels = AudioRing[old].HwChannels;
    AudioRing[AudioRingWrite].InSampleRate = AudioRing[old].InSampleRate;
@@ -2487,7 +2531,7 @@ int64_t AudioGetClock(void)

 	// delay zero, if no valid time stamp
 	if ((delay = AudioGetDelay())) {
-	    if (AudioRing[AudioRingRead].UseAc3) {
+	    if (AudioRing[AudioRingRead].Passthrough) {
 		return AudioRing[AudioRingRead].PTS + 0 * 90 - delay;
 	    }
 	    return AudioRing[AudioRingRead].PTS + 0 * 90 - delay;
@@ -2507,7 +2551,7 @@ void AudioSetVolume(int volume)
    AudioMute = !volume;
    // reduce loudness for stereo output
    if (AudioStereoDescent && AudioRing[AudioRingRead].InChannels == 2
-	&& !AudioRing[AudioRingRead].UseAc3) {
+	&& !AudioRing[AudioRingRead].Passthrough) {
 	volume -= AudioStereoDescent;
 	if (volume < 0) {
 	    volume = 0;
@@ -2524,18 +2568,20 @@ void AudioSetVolume(int volume)
 /**
 **	Setup audio for requested format.
 **
-**	@param freq	sample frequency
-**	@param channels	number of channels
-**	@param use_ac3	use ac3/pass-through device
+**	@param freq		sample frequency
+**	@param channels		number of channels
+**	@param passthrough	use pass-through (AC3, ...) device
 **
 **	@retval 0	everything ok
 **	@retval 1	didn't support frequency/channels combination
 **	@retval -1	something gone wrong
+**
+**	@todo add support to report best fitting format.
 */
-int AudioSetup(int *freq, int *channels, int use_ac3)
+int AudioSetup(int *freq, int *channels, int passthrough)
 {
    Debug(3, "audio: setup channels %d frequency %dHz %s\n", *channels, *freq,
-	use_ac3 ? "ac3" : "pcm");
+	passthrough ? "pass-through" : "");

    // invalid parameter
    if (!freq || !channels || !*freq || !*channels) {
@@ -2543,7 +2589,7 @@ int AudioSetup(int *freq, int *channels, int use_ac3)
 	// FIXME: set flag invalid setup
 	return -1;
    }
-    return AudioRingAdd(*freq, *channels, use_ac3);
+    return AudioRingAdd(*freq, *channels, passthrough);
 }

 /**
@@ -2679,7 +2725,7 @@ void AudioSetDevice(const char *device)
 **
 **	@note this is currently usable with alsa only.
 */
-void AudioSetDeviceAC3(const char *device)
+void AudioSetPassthroughDevice(const char *device)
 {
    if (!AudioModuleName) {
 	AudioModuleName = "alsa";	// detect alsa/OSS
@@ -2689,7 +2735,7 @@ void AudioSetDeviceAC3(const char *device)
 	    AudioModuleName = "oss";
 	}
    }
-    AudioAC3Device = device;
+    AudioPassthroughDevice = device;
 }

 /**
@@ -2748,10 +2794,16 @@ void AudioInit(void)
    //	Check which channels/rates/formats are supported
    //	FIXME: we force 44.1Khz and 48Khz must be supported equal
    //	FIXME: should use bitmap of channels supported in RatesInHw
+    //	FIXME: use loop over sample-rates
    freq = 44100;
    AudioRatesInHw[Audio44100] = 0;
    for (chan = 1; chan < 9; ++chan) {
-	if (AudioUsedModule->Setup(&freq, &chan, 0)) {
+	int tchan;
+	int tfreq;
+
+	tchan = chan;
+	tfreq = freq;
+	if (AudioUsedModule->Setup(&tfreq, &tchan, 0)) {
 	    AudioChannelsInHw[chan] = 0;
 	} else {
 	    AudioChannelsInHw[chan] = chan;
@@ -2761,16 +2813,39 @@ void AudioInit(void)
    freq = 48000;
    AudioRatesInHw[Audio48000] = 0;
    for (chan = 1; chan < 9; ++chan) {
+	int tchan;
+	int tfreq;
+
 	if (!AudioChannelsInHw[chan]) {
 	    continue;
 	}
-	if (AudioUsedModule->Setup(&freq, &chan, 0)) {
-	    AudioChannelsInHw[chan] = 0;
+	tchan = chan;
+	tfreq = freq;
+	if (AudioUsedModule->Setup(&tfreq, &tchan, 0)) {
+	    //AudioChannelsInHw[chan] = 0;
 	} else {
 	    AudioChannelsInHw[chan] = chan;
 	    AudioRatesInHw[Audio48000] |= (1 << chan);
 	}
    }
+    freq = 192000;
+    AudioRatesInHw[Audio192000] = 0;
+    for (chan = 1; chan < 9; ++chan) {
+	int tchan;
+	int tfreq;
+
+	if (!AudioChannelsInHw[chan]) {
+	    continue;
+	}
+	tchan = chan;
+	tfreq = freq;
+	if (AudioUsedModule->Setup(&tfreq, &tchan, 0)) {
+	    //AudioChannelsInHw[chan] = 0;
+	} else {
+	    AudioChannelsInHw[chan] = chan;
+	    AudioRatesInHw[Audio192000] |= (1 << chan);
+	}
+    }
    //	build channel support and conversion table
    for (u = 0; u < AudioRatesMax; ++u) {
 	for (chan = 1; chan < 9; ++chan) {
@@ -2852,13 +2927,16 @@ void AudioInit(void)
 */
 void AudioExit(void)
 {
+    const AudioModule *module;
+
 #ifdef USE_AUDIO_THREAD
    if (AudioUsedModule->Thread) {	// supports threads
 	AudioExitThread();
    }
 #endif
-    AudioUsedModule->Exit();
+    module = AudioUsedModule;
    AudioUsedModule = &NoopModule;
+    module->Exit();
    AudioRingExit();
    AudioRunning = 0;
    AudioPaused = 0;
@@ -2905,7 +2983,7 @@ static void PrintVersion(void)
 #ifdef GIT_REV
 	"(GIT-" GIT_REV ")"
 #endif
-	",\n\t(c) 2009 - 2012 by Johns\n"
+	",\n\t(c) 2009 - 2013 by Johns\n"
 	"\tLicense AGPLv3: GNU Affero General Public License version 3\n");
 }

@@ -2962,7 +3040,7 @@ int main(int argc, char *const argv[])
 		return -1;
 	    default:
 		PrintVersion();
-		fprintf(stderr, "Unkown option '%c'\n", optopt);
+		fprintf(stderr, "Unknown option '%c'\n", optopt);
 		return -1;
 	}
 	break;
--- a/audio.h
+++ b/audio.h
@@ -1,7 +1,7 @@
 ///
 ///	@file audio.h		@brief Audio module headerfile
 ///
-///	Copyright (c) 2009 - 2012 by Johns.  All Rights Reserved.
+///	Copyright (c) 2009 - 2013 by Johns.  All Rights Reserved.
 ///
 ///	Contributor(s):
 ///
@@ -48,7 +48,9 @@ extern void AudioSetCompression(int, int);	///< set compression parameters
 extern void AudioSetStereoDescent(int);	///< set stereo loudness descent

 extern void AudioSetDevice(const char *);	///< set PCM audio device
-extern void AudioSetDeviceAC3(const char *);	///< set pass-through device
+
+    /// set pass-through device
+extern void AudioSetPassthroughDevice(const char *);
 extern void AudioSetChannel(const char *);	///< set mixer channel
 extern void AudioInit(void);		///< setup audio module
 extern void AudioExit(void);		///< cleanup and exit audio module
--- a/codec.c
+++ b/codec.c
@@ -1,7 +1,7 @@
 ///
 ///	@file codec.c	@brief Codec functions
 ///
-///	Copyright (c) 2009 - 2012 by Johns.  All Rights Reserved.
+///	Copyright (c) 2009 - 2013 by Johns.  All Rights Reserved.
 ///
 ///	Contributor(s):
 ///
@@ -30,12 +30,14 @@
 ///		many bugs and incompatiblity in it.  Don't use this shit.
 ///

-    /// compile with passthrough support (stable, ac3 only)
+    /// compile with pass-through support (stable, AC-3, E-AC-3 only)
 #define USE_PASSTHROUGH
-    /// compile audio drift correction support (experimental)
+    /// compile audio drift correction support (very experimental)
 #define USE_AUDIO_DRIFT_CORRECTION
-    /// compile AC3 audio drift correction support (experimental)
+    /// compile AC-3 audio drift correction support (very experimental)
 #define USE_AC3_DRIFT_CORRECTION
+    /// use ffmpeg libswresample API (autodected, Makefile)
+#define noUSE_SWRESAMPLE

 #include <stdio.h>
 #include <unistd.h>
@@ -58,6 +60,9 @@
 #ifdef USE_VDPAU
 #include <libavcodec/vdpau.h>
 #endif
+#ifdef USE_SWRESAMPLE
+#include <libswresample/swresample.h>
+#endif

 #ifndef __USE_GNU
 #define __USE_GNU
@@ -115,9 +120,11 @@ struct _video_decoder_
 /**
 **	Callback to negotiate the PixelFormat.
 **
-**	@param fmt	is the list of formats which are supported by the codec,
-**			it is terminated by -1 as 0 is a valid format, the
-**			formats are ordered by quality.
+**	@param video_ctx	codec context
+**	@param fmt		is the list of formats which are supported by
+**				the codec, it is terminated by -1 as 0 is a
+**				valid format, the formats are ordered by
+**				quality.
 */
 static enum PixelFormat Codec_get_format(AVCodecContext * video_ctx,
    const enum PixelFormat *fmt)
@@ -125,7 +132,24 @@ static enum PixelFormat Codec_get_format(AVCodecContext * video_ctx,
    VideoDecoder *decoder;

    decoder = video_ctx->opaque;
-    //Debug(3, "codec: %s: %18p\n", __FUNCTION__, decoder);
+#if LIBAVCODEC_VERSION_INT == AV_VERSION_INT(54,86,100)
+    // this begins to stink, 1.1.2 calls get_format for each frame
+    // 1.1.3 has the same version, but works again
+    if (decoder->GetFormatDone) {
+	if (decoder->GetFormatDone < 10) {
+	    ++decoder->GetFormatDone;
+	    Error
+		("codec/video: ffmpeg/libav buggy: get_format called again\n");
+	}
+	return *fmt;			// FIXME: this is hack
+    }
+#endif
+
+    // bug in ffmpeg 1.1.1, called with zero width or height
+    if (!video_ctx->width || !video_ctx->height) {
+	Error("codec/video: ffmpeg/libav buggy: width or height zero\n");
+    }
+
    decoder->GetFormatDone = 1;
    return Video_get_format(decoder->HwDecoder, video_ctx, fmt);
 }
@@ -143,11 +167,15 @@ static int Codec_get_buffer(AVCodecContext * video_ctx, AVFrame * frame)
    VideoDecoder *decoder;

    decoder = video_ctx->opaque;
+#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(54,86,100)
+    // ffmpeg has this already fixed
+    // libav 0.8.5 53.35.0 still needs this
+#endif
    if (!decoder->GetFormatDone) {	// get_format missing
 	enum PixelFormat fmts[2];

-	fprintf(stderr, "codec: buggy ffmpeg/libav\n");
-	Warning(_("codec: buggy ffmpeg/libav\n"));
+	fprintf(stderr, "codec: buggy libav, use ffmpeg\n");
+	Warning(_("codec: buggy libav, use ffmpeg\n"));
 	fmts[0] = video_ctx->pix_fmt;
 	fmts[1] = PIX_FMT_NONE;
 	Codec_get_format(video_ctx, fmts);
@@ -160,7 +188,7 @@ static int Codec_get_buffer(AVCodecContext * video_ctx, AVFrame * frame)
 	unsigned surface;
 	struct vdpau_render_state *vrs;

-	surface = VideoGetSurface(decoder->HwDecoder);
+	surface = VideoGetSurface(decoder->HwDecoder, video_ctx);
 	vrs = av_mallocz(sizeof(struct vdpau_render_state));
 	vrs->surface = surface;

@@ -189,7 +217,7 @@ static int Codec_get_buffer(AVCodecContext * video_ctx, AVFrame * frame)
    if (video_ctx->hwaccel_context) {
 	unsigned surface;

-	surface = VideoGetSurface(decoder->HwDecoder);
+	surface = VideoGetSurface(decoder->HwDecoder, video_ctx);

 	//Debug(3, "codec: use surface %#010x\n", surface);

@@ -413,8 +441,11 @@ void CodecVideoOpen(VideoDecoder * decoder, const char *name, int codec_id)
    }
    if (video_codec->capabilities & CODEC_CAP_TRUNCATED) {
 	Debug(3, "codec: video can use truncated packets\n");
+#ifndef USE_MPEG_COMPLETE
 	// we send incomplete frames, for old PES recordings
+	// this breaks the decoder for some stations
 	decoder->VideoCtx->flags |= CODEC_FLAG_TRUNCATED;
+#endif
    }
    // FIXME: own memory management for video frames.
    if (video_codec->capabilities & CODEC_CAP_DR1) {
@@ -619,14 +650,27 @@ struct _audio_decoder_
    AVCodec *AudioCodec;		///< audio codec
    AVCodecContext *AudioCtx;		///< audio codec context

-    int PassthroughAC3;			///< current ac-3 pass-through
+    char Passthrough;			///< current pass-through flags
    int SampleRate;			///< current stream sample rate
    int Channels;			///< current stream channels

    int HwSampleRate;			///< hw sample rate
    int HwChannels;			///< hw channels

+#ifndef USE_SWRESAMPLE
    ReSampleContext *ReSample;		///< audio resampling context
+#endif
+#ifdef USE_SWRESAMPLE
+#if LIBSWRESAMPLE_VERSION_INT < AV_VERSION_INT(0, 15, 100)
+    struct SwrContext *Resample;	///< audio software resample context
+#else
+    SwrContext *Resample;		///< audio software resample context
+#endif
+#endif
+
+    uint16_t Spdif[24576 / 2];		///< SPDIF output buffer
+    int SpdifIndex;			///< index into SPDIF output buffer
+    int SpdifCount;			///< SPDIF repeat counter

    int64_t LastDelay;			///< last delay
    struct timespec LastTime;		///< last time
@@ -636,6 +680,7 @@ struct _audio_decoder_
    int DriftCorr;			///< audio drift correction value
    int DriftFrac;			///< audio drift fraction for ac3

+#ifndef USE_SWRESAMPLE
    struct AVResampleContext *AvResample;	///< second audio resample context
 #define MAX_CHANNELS 8			///< max number of channels supported
    int16_t *Buffer[MAX_CHANNELS];	///< deinterleave sample buffers
@@ -643,24 +688,35 @@ struct _audio_decoder_
    int16_t *Remain[MAX_CHANNELS];	///< filter remaining samples
    int RemainSize;			///< size of remain buffer
    int RemainCount;			///< number of remaining samples
+#endif
+};
+
+///
+///	IEC Data type enumeration.
+///
+enum IEC61937
+{
+    IEC61937_AC3 = 0x01,		///< AC-3 data
+    // FIXME: more data types
+    IEC61937_EAC3 = 0x15,		///< E-AC-3 data
 };

 #ifdef USE_AUDIO_DRIFT_CORRECTION
+#define CORRECT_PCM	1		///< do PCM audio-drift correction
+#define CORRECT_AC3	2		///< do AC3 audio-drift correction
 static char CodecAudioDrift;		///< flag: enable audio-drift correction
 #else
 static const int CodecAudioDrift = 0;
 #endif
 #ifdef USE_PASSTHROUGH
-//static char CodecPassthroughPCM;	///< pass pcm through (unsupported)
-static char CodecPassthroughAC3;	///< pass ac3 through
-
-//static char CodecPassthroughDTS;	///< pass dts through (unsupported)
-//static char CodecPassthroughMPA;	///< pass mpa through (unsupported)
+    ///
+    /// Pass-through flags: CodecPCM, CodecAC3, CodecEAC3, ...
+    ///
+static char CodecPassthrough;
 #else
-
-static const int CodecPassthroughAC3 = 0;
+static const int CodecPassthrough = 0;
 #endif
-static char CodecDownmix;		///< enable ac-3 downmix
+static char CodecDownmix;		///< enable AC-3 decoder downmix

 /**
 **	Allocate a new audio decoder context.
@@ -715,10 +771,16 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name,
    }

    if (CodecDownmix) {
+#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(53,61,100) || FF_API_REQUEST_CHANNELS
 	audio_decoder->AudioCtx->request_channels = 2;
+#endif
 	audio_decoder->AudioCtx->request_channel_layout =
 	    AV_CH_LAYOUT_STEREO_DOWNMIX;
    }
+#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53,61,100)
+    // this has no effect
+    // audio_decoder->AudioCtx->request_sample_fmt = AV_SAMPLE_FMT_S16;
+#endif
    pthread_mutex_lock(&CodecLockMutex);
    // open codec
 #if LIBAVCODEC_VERSION_INT <= AV_VERSION_INT(53,5,0)
@@ -765,6 +827,7 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name,
 void CodecAudioClose(AudioDecoder * audio_decoder)
 {
    // FIXME: output any buffered data
+#ifndef USE_SWRESAMPLE
    if (audio_decoder->AvResample) {
 	int ch;

@@ -784,6 +847,12 @@ void CodecAudioClose(AudioDecoder * audio_decoder)
 	audio_resample_close(audio_decoder->ReSample);
 	audio_decoder->ReSample = NULL;
    }
+#endif
+#ifdef USE_SWRESAMPLE
+    if (audio_decoder->Resample) {
+	swr_free(&audio_decoder->Resample);
+    }
+#endif
    if (audio_decoder->AudioCtx) {
 	pthread_mutex_lock(&CodecLockMutex);
 	avcodec_close(audio_decoder->AudioCtx);
@@ -800,7 +869,7 @@ void CodecAudioClose(AudioDecoder * audio_decoder)
 void CodecSetAudioDrift(int mask)
 {
 #ifdef USE_AUDIO_DRIFT_CORRECTION
-    CodecAudioDrift = mask & 3;
+    CodecAudioDrift = mask & (CORRECT_PCM | CORRECT_AC3);
 #endif
    (void)mask;
 }
@@ -808,12 +877,12 @@ void CodecSetAudioDrift(int mask)
 /**
 **	Set audio pass-through.
 **
-**	@param mask	enable mask (PCM, AC3)
+**	@param mask	enable mask (PCM, AC3, EAC3)
 */
 void CodecSetAudioPassthrough(int mask)
 {
 #ifdef USE_PASSTHROUGH
-    CodecPassthroughAC3 = mask & 1 ? 1 : 0;
+    CodecPassthrough = mask & (CodecPCM | CodecAC3 | CodecEAC3);
 #endif
    (void)mask;
 }
@@ -892,6 +961,194 @@ static void CodecReorderAudioFrame(int16_t * buf, int size, int channels)
    }
 }

+/**
+**	Handle audio format changes helper.
+**
+**	@param audio_decoder	audio decoder data
+**	@param[out] passthrough	pass-through output
+*/
+static int CodecAudioUpdateHelper(AudioDecoder * audio_decoder,
+    int *passthrough)
+{
+    const AVCodecContext *audio_ctx;
+    int err;
+
+    audio_ctx = audio_decoder->AudioCtx;
+    Debug(3, "codec/audio: format change %s %dHz *%d channels%s%s%s%s%s\n",
+	av_get_sample_fmt_name(audio_ctx->sample_fmt), audio_ctx->sample_rate,
+	audio_ctx->channels, CodecPassthrough & CodecPCM ? " PCM" : "",
+	CodecPassthrough & CodecMPA ? " MPA" : "",
+	CodecPassthrough & CodecAC3 ? " AC3" : "",
+	CodecPassthrough & CodecEAC3 ? " EAC3" : "",
+	CodecPassthrough ? " pass-through" : "");
+
+    *passthrough = 0;
+    audio_decoder->SampleRate = audio_ctx->sample_rate;
+    audio_decoder->HwSampleRate = audio_ctx->sample_rate;
+    audio_decoder->Channels = audio_ctx->channels;
+    audio_decoder->HwChannels = audio_ctx->channels;
+    audio_decoder->Passthrough = CodecPassthrough;
+
+    // SPDIF/HDMI pass-through
+    if ((CodecPassthrough & CodecAC3 && audio_ctx->codec_id == CODEC_ID_AC3)
+	|| (CodecPassthrough & CodecEAC3
+	    && audio_ctx->codec_id == CODEC_ID_EAC3)) {
+	if (audio_ctx->codec_id == CODEC_ID_EAC3) {
+	    // EAC3 over HDMI some receivers need HBR
+	    audio_decoder->HwSampleRate *= 4;
+	}
+	audio_decoder->HwChannels = 2;
+	audio_decoder->SpdifIndex = 0;	// reset buffer
+	audio_decoder->SpdifCount = 0;
+	*passthrough = 1;
+    }
+    // channels/sample-rate not support?
+    if ((err =
+	    AudioSetup(&audio_decoder->HwSampleRate,
+		&audio_decoder->HwChannels, *passthrough))) {
+
+	// try EAC3 none HBR
+	audio_decoder->HwSampleRate /= 4;
+	if (audio_ctx->codec_id != CODEC_ID_EAC3
+	    || (err =
+		AudioSetup(&audio_decoder->HwSampleRate,
+		    &audio_decoder->HwChannels, *passthrough))) {
+
+	    Debug(3, "codec/audio: audio setup error\n");
+	    // FIXME: handle errors
+	    audio_decoder->HwChannels = 0;
+	    audio_decoder->HwSampleRate = 0;
+	    return err;
+	}
+    }
+
+    Debug(3, "codec/audio: resample %s %dHz *%d -> %s %dHz *%d\n",
+	av_get_sample_fmt_name(audio_ctx->sample_fmt), audio_ctx->sample_rate,
+	audio_ctx->channels, av_get_sample_fmt_name(AV_SAMPLE_FMT_S16),
+	audio_decoder->HwSampleRate, audio_decoder->HwChannels);
+
+    return 0;
+}
+
+/**
+**	Audio pass-through decoder helper.
+**
+**	@param audio_decoder	audio decoder data
+**	@param avpkt		undecoded audio packet
+*/
+static int CodecAudioPassthroughHelper(AudioDecoder * audio_decoder,
+    const AVPacket * avpkt)
+{
+#ifdef USE_PASSTHROUGH
+    const AVCodecContext *audio_ctx;
+
+    audio_ctx = audio_decoder->AudioCtx;
+    // SPDIF/HDMI passthrough
+    if (CodecPassthrough & CodecAC3 && audio_ctx->codec_id == CODEC_ID_AC3) {
+	uint16_t *spdif;
+	int spdif_sz;
+
+	spdif = audio_decoder->Spdif;
+	spdif_sz = 6144;
+
+#ifdef USE_AC3_DRIFT_CORRECTION
+	// FIXME: this works with some TVs/AVReceivers
+	// FIXME: write burst size drift correction, which should work with all
+	if (CodecAudioDrift & CORRECT_AC3) {
+	    int x;
+
+	    x = (audio_decoder->DriftFrac +
+		(audio_decoder->DriftCorr * spdif_sz)) / (10 *
+		audio_decoder->HwSampleRate * 100);
+	    audio_decoder->DriftFrac =
+		(audio_decoder->DriftFrac +
+		(audio_decoder->DriftCorr * spdif_sz)) % (10 *
+		audio_decoder->HwSampleRate * 100);
+	    // round to word border
+	    x *= audio_decoder->HwChannels * 4;
+	    if (x < -64) {		// limit correction
+		x = -64;
+	    } else if (x > 64) {
+		x = 64;
+	    }
+	    spdif_sz += x;
+	}
+#endif
+
+	// build SPDIF header and append A52 audio to it
+	// avpkt is the original data
+	if (spdif_sz < avpkt->size + 8) {
+	    Error(_("codec/audio: decoded data smaller than encoded\n"));
+	    return -1;
+	}
+	spdif[0] = htole16(0xF872);	// iec 61937 sync word
+	spdif[1] = htole16(0x4E1F);
+	spdif[2] = htole16(IEC61937_AC3 | (avpkt->data[5] & 0x07) << 8);
+	spdif[3] = htole16(avpkt->size * 8);
+	// copy original data for output
+	// FIXME: not 100% sure, if endian is correct on not intel hardware
+	swab(avpkt->data, spdif + 4, avpkt->size);
+	// FIXME: don't need to clear always
+	memset(spdif + 4 + avpkt->size / 2, 0, spdif_sz - 8 - avpkt->size);
+	// don't play with the ac-3 samples
+	AudioEnqueue(spdif, spdif_sz);
+	return 1;
+    }
+    if (CodecPassthrough & CodecEAC3 && audio_ctx->codec_id == CODEC_ID_EAC3) {
+	uint16_t *spdif;
+	int spdif_sz;
+	int repeat;
+
+	// build SPDIF header and append A52 audio to it
+	// avpkt is the original data
+	spdif = audio_decoder->Spdif;
+	spdif_sz = 24576;		// 4 * 6144
+	if (audio_decoder->HwSampleRate == 48000) {
+	    spdif_sz = 6144;
+	}
+	if (spdif_sz < audio_decoder->SpdifIndex + avpkt->size + 8) {
+	    Error(_("codec/audio: decoded data smaller than encoded\n"));
+	    return -1;
+	}
+	// check if we must pack multiple packets
+	repeat = 1;
+	if ((avpkt->data[4] & 0xc0) != 0xc0) {	// fscod
+	    static const uint8_t eac3_repeat[4] = { 6, 3, 2, 1 };
+
+	    // fscod2
+	    repeat = eac3_repeat[(avpkt->data[4] & 0x30) >> 4];
+	}
+	// fprintf(stderr, "repeat %d %d\n", repeat, avpkt->size);
+
+	// copy original data for output
+	// pack upto repeat EAC-3 pakets into one IEC 61937 burst
+	// FIXME: not 100% sure, if endian is correct on not intel hardware
+	swab(avpkt->data, spdif + 4 + audio_decoder->SpdifIndex, avpkt->size);
+	audio_decoder->SpdifIndex += avpkt->size;
+	if (++audio_decoder->SpdifCount < repeat) {
+	    return 1;
+	}
+
+	spdif[0] = htole16(0xF872);	// iec 61937 sync word
+	spdif[1] = htole16(0x4E1F);
+	spdif[2] = htole16(IEC61937_EAC3);
+	spdif[3] = htole16(audio_decoder->SpdifIndex * 8);
+	memset(spdif + 4 + audio_decoder->SpdifIndex / 2, 0,
+	    spdif_sz - 8 - audio_decoder->SpdifIndex);
+
+	// don't play with the eac-3 samples
+	AudioEnqueue(spdif, spdif_sz);
+
+	audio_decoder->SpdifIndex = 0;
+	audio_decoder->SpdifCount = 0;
+	return 1;
+    }
+#endif
+    return 0;
+}
+
+#ifndef USE_SWRESAMPLE
+
 /**
 **	Set/update audio pts clock.
 **
@@ -913,14 +1170,15 @@ static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
    if (!delay) {
 	return;
    }
-    clock_gettime(CLOCK_REALTIME, &nowtime);
+    clock_gettime(CLOCK_MONOTONIC, &nowtime);
    if (!audio_decoder->LastDelay) {
 	audio_decoder->LastTime = nowtime;
 	audio_decoder->LastPTS = pts;
 	audio_decoder->LastDelay = delay;
 	audio_decoder->Drift = 0;
 	audio_decoder->DriftFrac = 0;
-	Debug(3, "codec/audio: inital delay %" PRId64 "ms\n", delay / 90);
+	Debug(3, "codec/audio: inital drift delay %" PRId64 "ms\n",
+	    delay / 90);
 	return;
    }
    // collect over some time
@@ -964,8 +1222,10 @@ static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
 	audio_decoder->Drift = drift;
 	corr = (10 * audio_decoder->HwSampleRate * drift) / (90 * 1000);
 	// SPDIF/HDMI passthrough
-	if ((CodecAudioDrift & 2) && (!CodecPassthroughAC3
-		|| audio_decoder->AudioCtx->codec_id != CODEC_ID_AC3)) {
+	if ((CodecAudioDrift & CORRECT_AC3) && (!(CodecPassthrough & CodecAC3)
+		|| audio_decoder->AudioCtx->codec_id != CODEC_ID_AC3)
+	    && (!(CodecPassthrough & CodecEAC3)
+		|| audio_decoder->AudioCtx->codec_id != CODEC_ID_EAC3)) {
 	    audio_decoder->DriftCorr = -corr;
 	}

@@ -1002,14 +1262,15 @@ static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
 **	Handle audio format changes.
 **
 **	@param audio_decoder	audio decoder data
+**
+**	@note this is the old not good supported version
 */
 static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
 {
+    int passthrough;
    const AVCodecContext *audio_ctx;
    int err;
-    int isAC3;

-    // FIXME: use swr_convert from swresample (only in ffmpeg!)
    if (audio_decoder->ReSample) {
 	audio_resample_close(audio_decoder->ReSample);
 	audio_decoder->ReSample = NULL;
@@ -1021,28 +1282,8 @@ static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
    }

    audio_ctx = audio_decoder->AudioCtx;
-    Debug(3, "codec/audio: format change %dHz %d channels %s\n",
-	audio_ctx->sample_rate, audio_ctx->channels,
-	CodecPassthroughAC3 ? "pass-through" : "");
+    if ((err = CodecAudioUpdateHelper(audio_decoder, &passthrough))) {

-    audio_decoder->SampleRate = audio_ctx->sample_rate;
-    audio_decoder->HwSampleRate = audio_ctx->sample_rate;
-    audio_decoder->Channels = audio_ctx->channels;
-    audio_decoder->PassthroughAC3 = CodecPassthroughAC3;
-
-    // SPDIF/HDMI passthrough
-    if (CodecPassthroughAC3 && audio_ctx->codec_id == CODEC_ID_AC3) {
-	audio_decoder->HwChannels = 2;
-	isAC3 = 1;
-    } else {
-	audio_decoder->HwChannels = audio_ctx->channels;
-	isAC3 = 0;
-    }
-
-    // channels not support?
-    if ((err =
-	    AudioSetup(&audio_decoder->HwSampleRate,
-		&audio_decoder->HwChannels, isAC3))) {
 	Debug(3, "codec/audio: resample %dHz *%d -> %dHz *%d\n",
 	    audio_ctx->sample_rate, audio_ctx->channels,
 	    audio_decoder->HwSampleRate, audio_decoder->HwChannels);
@@ -1058,19 +1299,21 @@ static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
 		Error(_("codec/audio: resample setup error\n"));
 		audio_decoder->HwChannels = 0;
 		audio_decoder->HwSampleRate = 0;
-		return;
 	    }
-	} else {
-	    Debug(3, "codec/audio: audio setup error\n");
-	    // FIXME: handle errors
-	    audio_decoder->HwChannels = 0;
-	    audio_decoder->HwSampleRate = 0;
 	    return;
 	}
+	Debug(3, "codec/audio: audio setup error\n");
+	// FIXME: handle errors
+	audio_decoder->HwChannels = 0;
+	audio_decoder->HwSampleRate = 0;
+	return;
+    }
+    if (passthrough) {			// pass-through no conversion allowed
+	return;
    }
    // prepare audio drift resample
 #ifdef USE_AUDIO_DRIFT_CORRECTION
-    if ((CodecAudioDrift & 1) && !isAC3) {
+    if (CodecAudioDrift & CORRECT_PCM) {
 	if (audio_decoder->AvResample) {
 	    Error(_("codec/audio: overwrite resample\n"));
 	}
@@ -1101,7 +1344,7 @@ static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
 void CodecAudioEnqueue(AudioDecoder * audio_decoder, int16_t * data, int count)
 {
 #ifdef USE_AUDIO_DRIFT_CORRECTION
-    if ((CodecAudioDrift & 1) && audio_decoder->AvResample) {
+    if ((CodecAudioDrift & CORRECT_PCM) && audio_decoder->AvResample) {
 	int16_t buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 +
 	    FF_INPUT_BUFFER_PADDING_SIZE] __attribute__ ((aligned(16)));
 	int16_t buftmp[MAX_CHANNELS][(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4];
@@ -1162,12 +1405,16 @@ void CodecAudioEnqueue(AudioDecoder * audio_decoder, int16_t * data, int count)
 	n *= 2;

 	n *= audio_decoder->HwChannels;
-	CodecReorderAudioFrame(buf, n, audio_decoder->HwChannels);
+	if (!(audio_decoder->Passthrough & CodecPCM)) {
+	    CodecReorderAudioFrame(buf, n, audio_decoder->HwChannels);
+	}
 	AudioEnqueue(buf, n);
 	return;
    }
 #endif
-    CodecReorderAudioFrame(data, count, audio_decoder->HwChannels);
+    if (!(audio_decoder->Passthrough & CodecPCM)) {
+	CodecReorderAudioFrame(data, count, audio_decoder->HwChannels);
+    }
    AudioEnqueue(data, count);
 }

@@ -1189,6 +1436,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)

    audio_ctx = audio_decoder->AudioCtx;

+    // FIXME: don't need to decode pass-through codecs
    buf_sz = sizeof(buf);
    l = avcodec_decode_audio3(audio_ctx, buf, &buf_sz, (AVPacket *) avpkt);
    if (avpkt->size != l) {
@@ -1202,20 +1450,12 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
 	}
 	Error(_("codec: error more than one frame data\n"));
    }
-#ifdef notyetFF_API_OLD_DECODE_AUDIO
-    // FIXME: ffmpeg git comeing
-    int got_frame;
-
-    avcodec_decode_audio4(audio_ctx, frame, &got_frame, avpkt);
-#else
-#endif
-
    // update audio clock
    if (avpkt->pts != (int64_t) AV_NOPTS_VALUE) {
 	CodecAudioSetClock(audio_decoder, avpkt->pts);
    }
    // FIXME: must first play remainings bytes, than change and play new.
-    if (audio_decoder->PassthroughAC3 != CodecPassthroughAC3
+    if (audio_decoder->Passthrough != CodecPassthrough
 	|| audio_decoder->SampleRate != audio_ctx->sample_rate
 	|| audio_decoder->Channels != audio_ctx->channels) {
 	CodecAudioUpdateFormat(audio_decoder);
@@ -1248,48 +1488,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
 		CodecAudioEnqueue(audio_decoder, outbuf, outlen);
 	    }
 	} else {
-#ifdef USE_PASSTHROUGH
-	    // SPDIF/HDMI passthrough
-	    if (CodecPassthroughAC3 && audio_ctx->codec_id == CODEC_ID_AC3) {
-		// build SPDIF header and append A52 audio to it
-		// avpkt is the original data
-		buf_sz = 6144;
-
-#ifdef USE_AC3_DRIFT_CORRECTION
-		if (CodecAudioDrift & 2) {
-		    int x;
-
-		    x = (audio_decoder->DriftFrac +
-			(audio_decoder->DriftCorr * buf_sz)) / (10 *
-			audio_decoder->HwSampleRate * 100);
-		    audio_decoder->DriftFrac =
-			(audio_decoder->DriftFrac +
-			(audio_decoder->DriftCorr * buf_sz)) % (10 *
-			audio_decoder->HwSampleRate * 100);
-		    x *= audio_decoder->HwChannels * 4;
-		    if (x < -64) {	// limit correction
-			x = -64;
-		    } else if (x > 64) {
-			x = 64;
-		    }
-		    buf_sz += x;
-		}
-#endif
-		if (buf_sz < avpkt->size + 8) {
-		    Error(_
-			("codec/audio: decoded data smaller than encoded\n"));
-		    return;
-		}
-		// copy original data for output
-		// FIXME: not 100% sure, if endian is correct
-		buf[0] = htole16(0xF872);	// iec 61937 sync word
-		buf[1] = htole16(0x4E1F);
-		buf[2] = htole16(0x01 | (avpkt->data[5] & 0x07) << 8);
-		buf[3] = htole16(avpkt->size * 8);
-		swab(avpkt->data, buf + 4, avpkt->size);
-		memset(buf + 4 + avpkt->size / 2, 0, buf_sz - 8 - avpkt->size);
-		// don't play with the ac-3 samples
-		AudioEnqueue(buf, buf_sz);
+	    if (CodecAudioPassthroughHelper(audio_decoder, avpkt)) {
 		return;
 	    }
 #if 0
@@ -1342,13 +1541,269 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
 	    }
 	    // DTS HD?
 	    // True HD?
-#endif
 #endif
 	    CodecAudioEnqueue(audio_decoder, buf, buf_sz);
 	}
    }
 }

+#endif
+
+#ifdef USE_SWRESAMPLE
+
+/**
+**	Set/update audio pts clock.
+**
+**	@param audio_decoder	audio decoder data
+**	@param pts		presentation timestamp
+*/
+static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
+{
+    struct timespec nowtime;
+    int64_t delay;
+    int64_t tim_diff;
+    int64_t pts_diff;
+    int drift;
+    int corr;
+
+    AudioSetClock(pts);
+
+    delay = AudioGetDelay();
+    if (!delay) {
+	return;
+    }
+    clock_gettime(CLOCK_MONOTONIC, &nowtime);
+    if (!audio_decoder->LastDelay) {
+	audio_decoder->LastTime = nowtime;
+	audio_decoder->LastPTS = pts;
+	audio_decoder->LastDelay = delay;
+	audio_decoder->Drift = 0;
+	audio_decoder->DriftFrac = 0;
+	Debug(3, "codec/audio: inital drift delay %" PRId64 "ms\n",
+	    delay / 90);
+	return;
+    }
+    // collect over some time
+    pts_diff = pts - audio_decoder->LastPTS;
+    if (pts_diff < 10 * 1000 * 90) {
+	return;
+    }
+
+    tim_diff = (nowtime.tv_sec - audio_decoder->LastTime.tv_sec)
+	* 1000 * 1000 * 1000 + (nowtime.tv_nsec -
+	audio_decoder->LastTime.tv_nsec);
+
+    drift =
+	(tim_diff * 90) / (1000 * 1000) - pts_diff + delay -
+	audio_decoder->LastDelay;
+
+    // adjust rounding error
+    nowtime.tv_nsec -= nowtime.tv_nsec % (1000 * 1000 / 90);
+    audio_decoder->LastTime = nowtime;
+    audio_decoder->LastPTS = pts;
+    audio_decoder->LastDelay = delay;
+
+    if (0) {
+	Debug(3,
+	    "codec/audio: interval P:%5" PRId64 "ms T:%5" PRId64 "ms D:%4"
+	    PRId64 "ms %f %d\n", pts_diff / 90, tim_diff / (1000 * 1000),
+	    delay / 90, drift / 90.0, audio_decoder->DriftCorr);
+    }
+    // underruns and av_resample have the same time :(((
+    if (abs(drift) > 10 * 90) {
+	// drift too big, pts changed?
+	Debug(3, "codec/audio: drift(%6d) %3dms reset\n",
+	    audio_decoder->DriftCorr, drift / 90);
+	audio_decoder->LastDelay = 0;
+#ifdef DEBUG
+	corr = 0;			// keep gcc happy
+#endif
+    } else {
+
+	drift += audio_decoder->Drift;
+	audio_decoder->Drift = drift;
+	corr = (10 * audio_decoder->HwSampleRate * drift) / (90 * 1000);
+	// SPDIF/HDMI passthrough
+	if ((CodecAudioDrift & CORRECT_AC3) && (!(CodecPassthrough & CodecAC3)
+		|| audio_decoder->AudioCtx->codec_id != CODEC_ID_AC3)
+	    && (!(CodecPassthrough & CodecEAC3)
+		|| audio_decoder->AudioCtx->codec_id != CODEC_ID_EAC3)) {
+	    audio_decoder->DriftCorr = -corr;
+	}
+
+	if (audio_decoder->DriftCorr < -20000) {	// limit correction
+	    audio_decoder->DriftCorr = -20000;
+	} else if (audio_decoder->DriftCorr > 20000) {
+	    audio_decoder->DriftCorr = 20000;
+	}
+    }
+
+    if (audio_decoder->Resample && audio_decoder->DriftCorr) {
+	int distance;
+
+	// try workaround for buggy ffmpeg 0.10
+	if (abs(audio_decoder->DriftCorr) < 2000) {
+	    distance = (pts_diff * audio_decoder->HwSampleRate) / (900 * 1000);
+	} else {
+	    distance = (pts_diff * audio_decoder->HwSampleRate) / (90 * 1000);
+	}
+	if (swr_set_compensation(audio_decoder->Resample,
+		audio_decoder->DriftCorr / 10, distance)) {
+	    Debug(3, "codec/audio: swr_set_compensation failed\n");
+	}
+    }
+    if (1) {
+	static int c;
+
+	if (!(c++ % 10)) {
+	    Debug(3, "codec/audio: drift(%6d) %8dus %5d\n",
+		audio_decoder->DriftCorr, drift * 1000 / 90, corr);
+	}
+    }
+}
+
+/**
+**	Handle audio format changes.
+**
+**	@param audio_decoder	audio decoder data
+*/
+static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
+{
+    int passthrough;
+    const AVCodecContext *audio_ctx;
+
+    if (CodecAudioUpdateHelper(audio_decoder, &passthrough)) {
+	// FIXME: handle swresample format conversions.
+	return;
+    }
+    if (passthrough) {			// pass-through no conversion allowed
+	return;
+    }
+
+    audio_ctx = audio_decoder->AudioCtx;
+
+#ifdef DEBUG
+    if (audio_ctx->sample_fmt == AV_SAMPLE_FMT_S16
+	&& audio_ctx->sample_rate == audio_decoder->HwSampleRate
+	&& !CodecAudioDrift) {
+	// FIXME: use Resample only, when it is needed!
+	fprintf(stderr, "no resample needed\n");
+    }
+#endif
+
+    audio_decoder->Resample =
+	swr_alloc_set_opts(audio_decoder->Resample, audio_ctx->channel_layout,
+	AV_SAMPLE_FMT_S16, audio_decoder->HwSampleRate,
+	audio_ctx->channel_layout, audio_ctx->sample_fmt,
+	audio_ctx->sample_rate, 0, NULL);
+    if (audio_decoder->Resample) {
+	swr_init(audio_decoder->Resample);
+    } else {
+	Error(_("codec/audio: can't setup resample\n"));
+    }
+}
+
+/**
+**	Decode an audio packet.
+**
+**	PTS must be handled self.
+**
+**	@note the caller has not aligned avpkt and not cleared the end.
+**
+**	@param audio_decoder	audio decoder data
+**	@param avpkt		audio packet
+*/
+void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
+{
+    AVCodecContext *audio_ctx;
+    AVFrame frame;
+    int got_frame;
+    int n;
+
+    audio_ctx = audio_decoder->AudioCtx;
+
+    // FIXME: don't need to decode pass-through codecs
+    frame.data[0] = NULL;
+    n = avcodec_decode_audio4(audio_ctx, &frame, &got_frame,
+	(AVPacket *) avpkt);
+    if (n != avpkt->size) {
+	if (n == AVERROR(EAGAIN)) {
+	    Error(_("codec/audio: latm\n"));
+	    return;
+	}
+	if (n < 0) {			// no audio frame could be decompressed
+	    Error(_("codec/audio: bad audio frame\n"));
+	    return;
+	}
+	Error(_("codec/audio: error more than one frame data\n"));
+    }
+    if (!got_frame) {
+	Error(_("codec/audio: no frame\n"));
+	return;
+    }
+    // update audio clock
+    if (avpkt->pts != (int64_t) AV_NOPTS_VALUE) {
+	CodecAudioSetClock(audio_decoder, avpkt->pts);
+    }
+    // format change
+    if (audio_decoder->Passthrough != CodecPassthrough
+	|| audio_decoder->SampleRate != audio_ctx->sample_rate
+	|| audio_decoder->Channels != audio_ctx->channels) {
+	CodecAudioUpdateFormat(audio_decoder);
+    }
+
+    if (!audio_decoder->HwSampleRate || !audio_decoder->HwChannels) {
+	return;				// unsupported sample format
+    }
+
+    if (CodecAudioPassthroughHelper(audio_decoder, avpkt)) {
+	return;
+    }
+
+    if (0) {
+	char strbuf[32];
+	int data_sz;
+	int plane_sz;
+
+	data_sz =
+	    av_samples_get_buffer_size(&plane_sz, audio_ctx->channels,
+	    frame.nb_samples, audio_ctx->sample_fmt, 1);
+	fprintf(stderr, "codec/audio: sample_fmt %s\n",
+	    av_get_sample_fmt_name(audio_ctx->sample_fmt));
+	av_get_channel_layout_string(strbuf, 32, audio_ctx->channels,
+	    audio_ctx->channel_layout);
+	fprintf(stderr, "codec/audio: layout %s\n", strbuf);
+	fprintf(stderr,
+	    "codec/audio: channels %d samples %d plane %d data %d\n",
+	    audio_ctx->channels, frame.nb_samples, plane_sz, data_sz);
+    }
+
+    if (audio_decoder->Resample) {
+	uint8_t outbuf[8192 * 2 * 8];
+	uint8_t *out[1];
+
+	out[0] = outbuf;
+	n = swr_convert(audio_decoder->Resample, out,
+	    sizeof(outbuf) / (2 * audio_decoder->HwChannels),
+	    (const uint8_t **)frame.extended_data, frame.nb_samples);
+	if (n > 0) {
+	    if (!(audio_decoder->Passthrough & CodecPCM)) {
+		CodecReorderAudioFrame((int16_t *) outbuf,
+		    n * 2 * audio_decoder->HwChannels,
+		    audio_decoder->HwChannels);
+	    }
+	    AudioEnqueue(outbuf, n * 2 * audio_decoder->HwChannels);
+	}
+	return;
+    }
+#ifdef DEBUG
+    // should be never reached
+    fprintf(stderr, "oops\n");
+#endif
+}
+
+#endif
+
 /**
 **	Flush the audio decoder.
 **
--- a/codec.h
+++ b/codec.h
@@ -1,7 +1,7 @@
 ///
 ///	@file codec.h	@brief Codec module headerfile
 ///
-///	Copyright (c) 2009 - 2012 by Johns.  All Rights Reserved.
+///	Copyright (c) 2009 - 2013 by Johns.  All Rights Reserved.
 ///
 ///	Contributor(s):
 ///
@@ -23,6 +23,16 @@
 /// @addtogroup Codec
 /// @{

+//----------------------------------------------------------------------------
+//	Defines
+//----------------------------------------------------------------------------
+
+#define CodecPCM 0x01			///< PCM bit mask
+#define CodecMPA 0x02			///< MPA bit mask (planned)
+#define CodecAC3 0x04			///< AC-3 bit mask
+#define CodecEAC3 0x08			///< EAC-3 bit mask
+#define CodecDTS 0x10			///< DTS bit mask (planned)
+
 //----------------------------------------------------------------------------
 //	Typedefs
 //----------------------------------------------------------------------------
--- a/make.sh
+++ b/make.sh
@@ -1,2 +0,0 @@
-#!/bin/sh
-exec make VDRDIR=/usr/include/vdr LIBDIR=. "$@"
--- a/po/de_DE.po
+++ b/po/de_DE.po
--- a/softhddev.c
+++ b/softhddev.c
--- a/softhddev.h
+++ b/softhddev.h
@@ -1,7 +1,7 @@
 ///
 ///	@file softhddev.h	@brief software HD device plugin header file.
 ///
-///	Copyright (c) 2011 - 2012 by Johns.  All Rights Reserved.
+///	Copyright (c) 2011 - 2013 by Johns.  All Rights Reserved.
 ///
 ///	Contributor(s):
 ///
@@ -95,6 +95,19 @@ extern "C"

    /// Get decoder statistics
    extern void GetStats(int *, int *, int *, int *);
+    /// C plugin scale video
+    extern void ScaleVideo(int, int, int, int);
+
+    /// Set Pip position
+    extern void PipSetPosition(int, int, int, int, int, int, int, int);
+    /// Pip start
+    extern void PipStart(int, int, int, int, int, int, int, int);
+    /// Pip stop
+    extern void PipStop(void);
+    /// Pip play video packet
+    extern int PipPlayVideo(const uint8_t *, int);
+
+    extern const char *X11DisplayName;	///< x11 display name
 #ifdef __cplusplus
 }
 #endif
--- a/softhddevice.cpp
+++ b/softhddevice.cpp
--- a/softhddevice_service.h
+++ b/softhddevice_service.h
@@ -23,6 +23,7 @@
 #pragma once

 #define ATMO_GRAB_SERVICE	"SoftHDDevice-AtmoGrabService-v1.0"
+#define ATMO1_GRAB_SERVICE	"SoftHDDevice-AtmoGrabService-v1.1"
 #define OSD_3DMODE_SERVICE	"SoftHDDevice-Osd3DModeService-v1.0"

 enum
@@ -48,3 +49,17 @@ typedef struct
 {
    int Mode;
 } SoftHDDevice_Osd3DModeService_v1_0_t;
+
+typedef struct
+{
+    // request/reply data
+
+    int width;
+    int height;
+
+    // reply data
+
+    int size;
+
+    void *img;
+} SoftHDDevice_AtmoGrabService_v1_1_t;
--- a/vdr-softhddevice-9999-pre1.7.36.ebuild
+++ b/vdr-softhddevice-9999-pre1.7.36.ebuild
@@ -0,0 +1,74 @@
+# Copyright 1999-2012 Gentoo Foundation
+# Distributed under the terms of the GNU General Public License v2
+# $Header: $
+
+EAPI="4"
+
+inherit eutils vdr-plugin-2
+
+if [[ ${PV} == "9999" ]] ; then
+		inherit git-2
+		EGIT_REPO_URI="git://projects.vdr-developer.org/vdr-plugin-softhddevice.git"
+else
+		SRC_URI="http://projects.vdr-developer.org/attachments/download/838/${P}.tgz"
+fi
+
+
+DESCRIPTION="A software and GPU emulated HD output device plugin for VDR."
+HOMEPAGE="http://projects.vdr-developer.org/projects/show/plg-softhddevice"
+SRC_URI=""
+
+LICENSE="AGPL-3"
+SLOT="0"
+KEYWORDS="~x86 ~amd64"
+IUSE="vaapi vdpau alsa oss yaepg opengl debug"
+
+DEPEND=">=x11-libs/libxcb-1.8
+		x11-libs/xcb-util
+		x11-libs/xcb-util-wm
+		x11-libs/xcb-util-keysyms
+		x11-libs/xcb-util-renderutil
+		x11-libs/libX11
+		opengl? ( virtual/opengl )
+		>=virtual/ffmpeg-0.7
+		sys-devel/gettext
+		sys-devel/make
+		dev-util/pkgconfig
+		yaepg? ( >=media-video/vdr-1.7.23[yaepg] )
+		!yaepg? ( >=media-video/vdr-1.7.23 )
+		vdpau? ( x11-libs/libvdpau virtual/ffmpeg[vdpau] )
+		vaapi? ( x11-libs/libva virtual/ffmpeg[vaapi] )
+		alsa? ( media-libs/alsa-lib )
+		oss? ( sys-kernel/linux-headers )
+"
+
+src_prepare() {
+		vdr-plugin-2_src_prepare
+}
+
+src_compile() {
+		local myconf
+
+		myconf="-DHAVE_PTHREAD_NAME -DAV_INFO -DAV_INFO_TIME=15000"
+		use vdpau && myconf="${myconf} -DUSE_VDPAU"
+		use vaapi && myconf="${myconf} -DUSE_VAAPI"
+		use alsa && myconf="${myconf} -DUSE_ALSA"
+		use oss && myconf="${myconf} -DUSE_OSS"
+		use debug && myconf="${myconf} -DDEBUG"
+
+		#vdr-plugin-2_src_compile
+		cd "${S}"
+
+		BUILD_TARGETS=${BUILD_TARGETS:-${VDRPLUGIN_MAKE_TARGET:-all}}
+
+		emake ${BUILD_PARAMS} CONFIG="${myconf}" \
+				${BUILD_TARGETS} \
+				LOCALEDIR="${TMP_LOCALE_DIR}" \
+				LIBDIR="${S}" \
+				TMPDIR="${T}" \
+		|| die "emake failed"
+}
+
+src_install() {
+		vdr-plugin-2_src_install
+}
--- a/vdr-softhddevice-9999.ebuild
+++ b/vdr-softhddevice-9999.ebuild
@@ -2,73 +2,64 @@
 # Distributed under the terms of the GNU General Public License v2
 # $Header: $

-EAPI="3"
+EAPI="5"

-inherit eutils vdr-plugin
+inherit flag-o-matic toolchain-funcs vdr-plugin-2 eutils

-if [[ ${PV} == "9999" ]] ; then
-		inherit git-2
-		EGIT_REPO_URI="git://projects.vdr-developer.org/vdr-plugin-softhddevice.git"
+if [ "${PV}" = "9999" ]; then
+	inherit git-2
+	EGIT_REPO_URI="git://projects.vdr-developer.org/vdr-plugin-softhddevice.git"
+	KEYWORDS=""
 else
-		SRC_URI="http://projects.vdr-developer.org/attachments/download/838/${P}.tgz"
+	SRC_URI="mirror://vdr-developerorg/889/${P}.tgz"
+	KEYWORDS="~amd64 ~x86"
 fi

-
-DESCRIPTION="A software and GPU emulated HD output device plugin for VDR."
+DESCRIPTION="Software and GPU emulated HD output device plugin for VDR"
 HOMEPAGE="http://projects.vdr-developer.org/projects/show/plg-softhddevice"
-SRC_URI=""

 LICENSE="AGPL-3"
 SLOT="0"
-KEYWORDS="~x86 ~amd64"
-IUSE="vaapi vdpau alsa oss yaepg opengl debug"
+IUSE="alsa oss vaapi vdpau yaepg xscreensaver debug"

-DEPEND=">=x11-libs/libxcb-1.8
-		x11-libs/xcb-util
-		x11-libs/xcb-util-wm
-		x11-libs/xcb-util-keysyms
-		x11-libs/xcb-util-renderutil
-		x11-libs/libX11
-		opengl? ( virtual/opengl )
-		>=virtual/ffmpeg-0.7
-		sys-devel/gettext
-		sys-devel/make
-		dev-util/pkgconfig
-		yaepg? ( >=media-video/vdr-1.7.23[yaepg] )
-		!yaepg? ( >=media-video/vdr-1.7.23 )
-		vdpau? ( x11-libs/libvdpau virtual/ffmpeg[vdpau] )
-		vaapi? ( x11-libs/libva virtual/ffmpeg[vaapi] )
-		alsa? ( media-libs/alsa-lib )
-		oss? ( sys-kernel/linux-headers )
-"
-
-src_prepare() {
-		vdr-plugin_src_prepare
-}
+RDEPEND=">=media-video/vdr-1.7
+	>=virtual/ffmpeg-0.7[vdpau?,vaapi?]
+	x11-libs/libX11
+	>=x11-libs/libxcb-1.8
+	x11-libs/xcb-util-wm
+	alsa? ( media-libs/alsa-lib )
+	vdpau? ( x11-libs/libvdpau )
+	vaapi? ( x11-libs/libva )
+	alsa? ( media-libs/alsa-lib )
+	yaepg? ( >=media-video/vdr-1.7[yaepg] )"
+DEPEND="${RDEPEND}
+	x11-libs/xcb-util
+	sys-devel/gettext
+	virtual/pkgconfig
+	oss? ( sys-kernel/linux-headers )"

 src_compile() {
-		local myconf
+	local myconf

-		myconf="-DHAVE_PTHREAD_NAME"
-		use vdpau && myconf="${myconf} -DUSE_VDPAU"
-		use vaapi && myconf="${myconf} -DUSE_VAAPI"
-		use alsa && myconf="${myconf} -DUSE_ALSA"
-		use oss && myconf="${myconf} -DUSE_OSS"
-		use debug && myconf="${myconf} -DDEBUG"
+	myconf+=" ALSA=$(usex alsa 1 0)"
+	myconf+=" OSS=$(usex oss 1 0)"
+	myconf+=" VDPAU=$(usex vdpau 1 0)"
+	myconf+=" VAAPI=$(usex vaapi 1 0)"
+	myconf+=" SCREENSAVER=$(usex xscreensaver 1 0)"
+	if has_version ">=media-video/ffmpeg-0.8" ; then
+		myconf+=" SWRESAMPLE=1"
+	fi

-		emake all CC="$(tc-getCC)" CFLAGS="${CFLAGS}" \
-			LDFLAGS="${LDFLAGS}" CONFIG="${myconf}" LIBDIR="." || die
+	append-cflags -DHAVE_PTHREAD_NAME -D_GNU_SOURCE
+	append-cxxflags -DHAVE_PTHREAD_NAME -D_GNU_SOURCE
+	tc-export CC CXX
+
+	BUILD_PARAMS="${myconf}"
+	vdr-plugin-2_src_compile
 }

 src_install() {
-		vdr-plugin_src_install
+	vdr-plugin-2_src_install

-		dodoc README.txt
-
-		#dodir /etc/vdr/plugins || die
-		#insinto /etc/vdr/plugins
-		#fowners -R vdr:vdr /etc/vdr || die
-
-		#insinto /etc/conf.d
-		#doins vdr.softhddevice
+	dodoc ChangeLog README.txt
 }
--- a/video.c
+++ b/video.c
--- a/video.h
+++ b/video.h
@@ -1,7 +1,7 @@
 ///
 ///	@file video.h	@brief Video module header file
 ///
-///	Copyright (c) 2009 - 2012 by Johns.  All Rights Reserved.
+///	Copyright (c) 2009 - 2013 by Johns.  All Rights Reserved.
 ///
 ///	Contributor(s):
 ///
@@ -30,6 +30,9 @@
    /// Video hardware decoder typedef
 typedef struct _video_hw_decoder_ VideoHwDecoder;

+    /// Video output stream typedef
+typedef struct __video_stream__ VideoStream;
+
 //----------------------------------------------------------------------------
 //	Variables
 //----------------------------------------------------------------------------
@@ -37,24 +40,25 @@ typedef struct _video_hw_decoder_ VideoHwDecoder;
 extern char VideoHardwareDecoder;	///< flag use hardware decoder
 extern char VideoIgnoreRepeatPict;	///< disable repeat pict warning
 extern int VideoAudioDelay;		///< audio/video delay
+extern char ConfigStartX11Server;	///< flag start the x11 server

 //----------------------------------------------------------------------------
 //	Prototypes
 //----------------------------------------------------------------------------

    /// Allocate new video hardware decoder.
-extern VideoHwDecoder *VideoNewHwDecoder(void);
+extern VideoHwDecoder *VideoNewHwDecoder(VideoStream *);

    /// Deallocate video hardware decoder.
 extern void VideoDelHwDecoder(VideoHwDecoder *);

+#ifdef LIBAVCODEC_VERSION
    /// Get and allocate a video hardware surface.
-extern unsigned VideoGetSurface(VideoHwDecoder *);
+extern unsigned VideoGetSurface(VideoHwDecoder *, const AVCodecContext *);

    /// Release a video hardware surface
 extern void VideoReleaseSurface(VideoHwDecoder *, unsigned);

-#ifdef LIBAVCODEC_VERSION
    /// Callback to negotiate the PixelFormat.
 extern enum PixelFormat Video_get_format(VideoHwDecoder *, AVCodecContext *,
    const enum PixelFormat *);
@@ -110,13 +114,16 @@ extern void VideoSetSaturation(int);
 extern void VideoSetHue(int);

    /// Set video output position.
-extern void VideoSetOutputPosition(int, int, int, int);
+extern void VideoSetOutputPosition(VideoHwDecoder *, int, int, int, int);

    /// Set video mode.
 extern void VideoSetVideoMode(int, int, int, int);

-    /// Set display format.
-extern void VideoSetDisplayFormat(int);
+    /// Set 4:3 display format.
+extern void VideoSet4to3DisplayFormat(int);
+
+    /// Set other display format.
+extern void VideoSetOtherDisplayFormat(int);

    /// Set video fullscreen mode.
 extern void VideoSetFullscreen(int);
@@ -205,8 +212,16 @@ extern void VideoOsdExit(void);		///< Cleanup osd.
 extern void VideoInit(const char *);	///< Setup video module.
 extern void VideoExit(void);		///< Cleanup and exit video module.

-extern int VideoPollInput(void);	///< Poll video input buffers.
-extern int VideoDecodeInput(void);	///< Decode video input buffers.
-extern int VideoGetBuffers(void);	///< Get number of input buffers.
+    /// Poll video input buffers.
+extern int VideoPollInput(VideoStream *);
+
+    /// Decode video input buffers.
+extern int VideoDecodeInput(VideoStream *);
+
+    /// Get number of input buffers.
+extern int VideoGetBuffers(const VideoStream *);
+
+       /// Raise the frontend window
+extern int VideoRaiseWindow();

 /// @}
Author	SHA1	Message	Date
Johns	eab051f5e6	Release Version 0.6.0.	2013-03-17 15:53:01 +01:00
anbr	e058302a08	Update german translation.	2013-03-17 15:51:11 +01:00
Johns	efcf3a1d2d	Fix Bug #1286 : typo.	2013-03-12 23:27:09 +01:00
Johns	1f0d5878b1	Adds H264 only hardware decoder for still-pictures.	2013-03-11 16:32:00 +01:00
Johns	f09a37a941	Enable optional VDR-SPU deocder support.	2013-03-11 12:18:11 +01:00
Johns	a7562eb2be	Removes "static" warning.	2013-03-06 17:31:01 +01:00
cyril	a747829ffb	Adds raise softhddevice video window support.	2013-03-06 17:07:30 +01:00
Johns	7db63875d0	Adds optional only complete mpeg packets support.	2013-03-06 10:30:27 +01:00
Johns	637c04655a	Fixes text of EAC-3 pass-through setup.	2013-03-05 14:39:13 +01:00
Johns	9954b939ef	Try to connect to X11 server, when start fails.	2013-02-27 16:48:42 +01:00
Johns	e65572c2e2	Updated ebuild for new vdr-plugin-2.eclass.	2013-02-25 16:40:44 +01:00
Johns	b3ddc47cb5	Removes "unused parameter" warnings.	2013-02-25 16:39:32 +01:00
Johns	4d9e3a71f4	More informations in short description.	2013-02-25 16:34:28 +01:00
Johns	21e4f4ee04	Workaround for ffmpeg 1.1.x get_format bug.	2013-02-25 16:33:40 +01:00
Johns	746746d5b7	Workaround for ffmpeg 1.1.2 bug.	2013-02-25 16:32:52 +01:00
Johns	936566c642	Show vaapi droped/missed messages with info level.	2013-02-23 15:56:24 +01:00
Johns	d6f557c6f7	Removes warnings, when compiles without VDPAU.	2013-02-15 15:58:49 +01:00
Johns	2d2ea53d16	Quote spaces to get correct ouput.	2013-02-13 22:47:16 +01:00
Johns	bccd959833	Fix .pot file generation, support _ and _N macros.	2013-02-13 22:46:08 +01:00
Johns	240fc17471	Store pass-through on/off state in setup.	2013-02-13 17:23:18 +01:00
Johns	7b2caac901	Fix bug: still old CodecPassthroughAC3 used.	2013-02-12 22:03:52 +01:00
Johns	56aae221e7	Adds missing USE_SCREENSAVER to old Makefile.	2013-02-12 18:11:40 +01:00
Johns	54255e7b57	Try to use HBR (High Bit-Rate) for EAC3.	2013-02-11 23:40:09 +01:00
Johns	2cd667fb44	Improved pass-through (PCM+EAC3) support.	2013-02-11 16:53:51 +01:00
Johns	145d65ff01	Debug time for channel switch.	2013-02-11 15:11:52 +01:00
Johns	8faff0fd1e	Try to detect wrong Makefile use.	2013-02-11 14:41:19 +01:00
Johns	d31ff55b12	Clear buffers quicker, when replay stops.	2013-02-08 21:03:06 +01:00
Johns	d4535a34c9	Fix xcb deadlock while closing PIP decoder. Close video decoder from inside the decoder thread, otherwise xcb hangs in a lock.	2013-02-07 16:54:16 +01:00
Johns	ebd2f85f90	Update for new Makefiles.	2013-02-07 10:17:48 +01:00
Johns	dce7ef9110	Adds PIP hot-key support.	2013-02-06 16:27:25 +01:00
Johns	780e2989ae	Makes audio mixer optional.	2013-02-06 16:02:22 +01:00
Johns	2661fdf333	Plugin needs -D_GNU_SOURCE.	2013-01-30 18:04:53 +01:00
Johns	2557418e81	No longer needed.	2013-01-29 20:48:04 +01:00
Johns	b48e0d0638	Support VDR 1.7.36 new build system.	2013-01-29 17:04:50 +01:00
Johns	2c27d83b9e	Makes buffers soft limit configurable.	2013-01-28 23:13:23 +01:00
Johns	8b22585748	Never trust the docs: fix ffmpeg downsample.	2013-01-26 00:18:59 +01:00
Johns	7ed6975330	Support upsampling 3,5 to 8 channels. Improve next ring buffer handling. Improve audio/video sync debug.	2013-01-25 17:05:53 +01:00
Johns	d5e111238d	Improves VDPAU dissplay preemption handling.	2013-01-25 16:55:25 +01:00
Sibbi	2733e47af7	Add modifiers to X11 remote key names.	2013-01-25 16:52:23 +01:00
Johns	f9998e7664	Add audio compatibility with >=ffmpeg 1.1.	2013-01-24 21:42:39 +01:00
Johns	04286fb2ad	Add video compatibility with >=ffmpeg 1.1.	2013-01-23 21:54:56 +01:00
Johns	cd82ee8e4a	Fix bug: no sound with video output "none".	2013-01-22 17:18:18 +01:00
Johns	a1f17199d6	Crop setup is done by VdpauSetupOutput.	2013-01-22 17:08:44 +01:00
Johns	849ca7cfd6	Stronger H264/Mpeg detection tests.	2013-01-12 19:13:43 +01:00
Johns	308742a927	React faster only with mpeg.	2013-01-11 18:54:45 +01:00
Johns	1a730ef90b	Fix build without USE_PIP.	2013-01-09 00:39:27 +01:00
Johns	d59c2ad40b	Must guard CodecVideoDecode.	2013-01-08 23:52:41 +01:00
Johns	3366faece2	Fixes close PIP threading bug.	2013-01-08 21:11:46 +01:00
Johns	2bb2875cd7	Update PIP only, when visible.	2013-01-07 19:31:05 +01:00
Johns	e88403d044	Close pip before switching channel.	2013-01-07 17:47:05 +01:00
Johns	11293e8dc1	Split mpeg packets in receiver thread.	2013-01-07 16:55:42 +01:00
Johns	3bcc3d280e	fprints only when building debug version.	2013-01-06 19:17:02 +01:00
Johns	55587c86f0	Switch PIP to only available channels.	2013-01-06 19:15:46 +01:00
Johns	7cc74795be	Fix bug: missing external reference with -DUSE_PIP.	2013-01-06 19:14:46 +01:00
Johns	534b4094b5	Fix thread exit problem.	2013-01-06 14:47:00 +01:00
Johns	7cd025a023	Cosmetic cleanups.	2013-01-06 14:06:08 +01:00
Johns	b14a67b601	New features documented.	2013-01-05 23:16:03 +01:00
Johns	b24409323d	Fix detach and attach bug.	2013-01-05 22:48:01 +01:00
Johns	9ec2716026	PIP function update.	2013-01-05 20:47:14 +01:00
Johns	fa09d940c5	Fix PIP threading problems.	2013-01-05 20:44:54 +01:00
Johns	d4702b9a9e	Include softhddevice.cpp in indent target. Fix constconst after indent. Hello GNU who many years until it will be fixed?	2013-01-05 19:49:59 +01:00
Johns	6a3a560857	Adds PIP channel switch functions.	2013-01-05 17:17:50 +01:00
Johns	8ee1e84b2e	Restore video position, when PIP closes.	2013-01-05 16:37:25 +01:00
Johns	7e96a292eb	First video surface clears complete window.	2013-01-05 15:41:57 +01:00
Johns	d89ada9aad	Makes PIP+Video position configurable.	2013-01-04 21:50:55 +01:00
Johns	59d1a6b1f2	Adds function to stop PIP.	2013-01-04 18:50:14 +01:00
Johns	1a744a8eb8	Sync to audio only with the first video stream.	2013-01-04 18:04:53 +01:00
Johns	78100cba00	Use second stream for PIP.	2013-01-04 15:56:26 +01:00
Johns	b54ddd4549	Enable more than 1 vdpau video stream.	2013-01-04 15:54:32 +01:00
Johns	d42475f2dc	Prepared reentrant video stream.	2013-01-03 18:52:34 +01:00
Johns	7cf6c1ab2b	va-api has problems with h264 interlace.	2013-01-03 01:03:22 +01:00
Johns	06b8e3d784	Fix video size not updated for VAAPI.	2013-01-02 23:59:46 +01:00
horchi	63c22a13cf	Adds VDR SeduAtmo Plugin support.	2013-01-01 17:58:54 +01:00
Johns	3d5e59a6e5	Support multiple streams with ScaleVideo.	2013-01-01 15:21:28 +01:00
Johns	acf377ec60	Makes 4:3 and 16:9 display format configurable. Don't use DVB display format.	2013-01-01 14:48:51 +01:00
Zoolook	87c1c7be84	Add support for new vdr ScaleVideo API.	2013-01-01 12:51:08 +01:00
Johns	5bacd0cf7b	Comments cleanup.	2013-01-01 12:40:12 +01:00
Johns	b1beb49798	Add support for old PES HDTV recording.	2013-01-01 12:09:54 +01:00
Johns	33b14be516	Change VideoSetOutputPosition to work OSD relative.	2012-12-31 17:51:43 +01:00
Johns	3907f3eb2f	Change +%d to %+d, which is more general.	2012-12-31 17:11:22 +01:00
Johns	00d314dcd1	Disable trickspeed hack, to prevent ffmpeg crash.	2012-12-31 14:03:19 +01:00
Johns	9719fef55b	Fix video size not updated, when window size change.	2012-12-30 17:05:42 +01:00
Johns	9b69045a20	Use 1 surface reserve.	2012-12-24 12:59:35 +01:00
Johns	966ff4229a	Better video stream window position support.	2012-12-24 12:23:16 +01:00
Johns	e0c0c2021f	Need LOCALEDIR to build as user.	2012-12-24 11:56:10 +01:00
Johns	a3907d11a5	Spam syslog with a/v sync infos.	2012-12-24 11:55:32 +01:00
Johns	b94d19e7b8	Make clear what downmix is enabled.	2012-12-24 11:53:16 +01:00
Johns	943ee22aa5	Makes X11 server arguments configurable.	2012-12-14 17:24:54 +01:00
Johns	b6154a988f	Fix build with NO_TS_AUDIO.	2012-12-14 17:04:50 +01:00
Johns	932871dea9	Use vdr-plugin-2.	2012-11-30 20:46:13 +01:00
Johns	5ec6963398	Missing space added.	2012-11-25 00:38:32 +01:00
Johns	b04323704d	Missing space added.	2012-11-25 00:35:39 +01:00
FireFly	38011d51b6	Add german translation.	2012-11-18 21:17:09 +01:00