Release Version 0.5.1

avcodec_flush_buffers crash fix.

ChangeLog

@@ -1,3 +1,174 @@
+User johns
+Date: Tue Jul  3 16:35:46 CEST 2012
+
+    Release Version 0.5.1
+    Add commandline support to disable hardware decoder.
+    Display frame statistics in plugin menu.
+    Fix bug: 100% CPU use during playback.
+    Fix bug: audio use 100% CPU during pause.
+    Guard audio skip against old PTS values.
+    Improved audio skip, after channel switch.
+    Setup add preview of collapsed tree.
+    Fix bug: dvd plugin times out.
+    Makes OSD size configurable.
+    Support display with non-square pixels.
+    Initial ADTS (AAC+) support.
+    Fix bug #959: log flooded with service messages.
+
+User durchflieger
+Date: Mon Apr 30 14:46:51 CEST 2012
+
+    Adds VDR DFAtmo Plugin support.
+
+User johns
+Date: Mon Apr 30 13:56:26 CEST 2012
+
+    Fix bug: don't normalize or compress pass-through samples.
+    Make audio ring buffer size a multiple of 3,5,7,8.
+    Add reset ring buffer support.
+    Fix bug: alloca wrong size for audio buffer.
+    Handle jump in stream like stream start.
+    Always compile audio drift correction.
+    Add audio drift correction configuration to the setup.
+
+User mini73
+Date: Fri Apr 20 16:51:14 CEST 2012
+
+    Add support for svdr command "stat".
+
+User johns
+Date: Fri Apr 20 16:05:40 CEST 2012
+
+    Made showing black picture configurable.
+    Show black picture, if no video stream is available.
+    Setup split into foldable sections.
+    Adds show cursor on pointer move and hide after 200ms.
+    Adds Hot-key support for auto-crop enable/disable/toggle.
+    Adds detached start mode.
+    Fix bug: VDPAU looses preemption callback.
+    Fix bug: X11 server keeps sending USR1 signals, which confuses suspend.
+    Show message for hot-keys.
+    Fix bug: playback errors with old PES recordings.
+    Adds Hot-key support for 4:3 zoom modes.
+
+User johns
+Date: Sat Apr  7 20:21:16 CEST 2012
+
+    Release Version 0.5.0
+    Change audio/video delay with hot-key.
+    Enable/disable/toggle fullscreen with hot-key (Feature #930).
+
+User: CafeDelMar
+Date: Thu Apr  5 22:44:06 CEST 2012
+
+    Cutting pixels are now configured for each resolution.
+
+User johns
+Date: Thu Apr 5 15:47:59 CEST 2012
+
+    Buffer less video and audio.
+    Fix 100% cpu use, with mp3 plugin.
+    Audio/Video sync rewrite, trick-speed support moved to video.
+    Faster VdpauBlackSurface version.
+    Fix bug: VideoSetPts wrong position for multi frame packets.
+
+User: CafeDelMar
+Date: Mon Mar 26 20:45:54 CEST 2012
+
+    Add VideoSkipPixels support.
+
+User johns
+Date: Fri Mar 23 18:43:20 CET 2012
+
+    Add optional argument (display) to ATTA svdrp commmand.
+    Wakeup display to show OSD for remote learning mode.
+    Support switching the primary device with svdrp.
+    Disable and reenable screen saver and DPMS.
+    Video source code cleanup.
+    Fix fast backward with some h264 streams.
+    Make soft start sync setup menu configurable.
+    Fix bug: StillPicture NAL end of sequence is 10 and not 0x10.
+    Fix bug: AudioEnqueue crash without sound card.
+
+User johns
+Date: Sun Mar  4 22:35:36 CET 2012
+
+    Release Version 0.4.9
+    Experimental ac3 audio drift correction support.
+    Removes LPCM detection from TS parser.
+    Rewrote video/audio start code.
+    Add support for attach/detach plugin.
+    OSS needs bigger audio buffers.
+    Improved audio drift correction support.
+    Experimental audio drift correction support.
+    Add SVDRP HOTK command support.
+    Increased audio buffer time for PES packets.
+    Support configuration and set of video background.
+    Survive lost X11 display.
+    Fix bug: 100% cpu use with plugins like mp3.
+    Wakeup display thread on channel switch, osd can now be shown without
+    video.
+    Makes 60Hz display mode configurable with setup.conf.
+    Support downmix of AC-3 to stero.
+    New audio PES packet parser.
+    Fix bug: Grabbing a JPG image fails while suspended.
+    Add support for hot keys.
+    Add support to use characters input in edit mode.
+    Adds trick speed support.
+
+User johns
+Date: Thu Feb 16 09:59:14 CET 2012
+
+    Release Version 0.4.8
+    Fix bug: wrong start of video packet.
+    VDPAU: Enables inverse telecine configuration.
+    Find AC3 (Dolby Digital) inside PES packet.
+    Fix bug: audio increments invalid audio PTS.
+    Fix bug: dvd plugin not working.
+    Fix bug: used frame-> instead of video_ctx-> for old libav/ffmpeg.
+
+User johns
+Date: Mon Feb 13 23:20:26 CET 2012
+
+    Release Version 0.4.7
+
+User FireFly
+Date: Mon Feb 13 20:14:11 CET 2012
+
+    Fix bug: unscaled jpeg includes PNG header.
+
+User johns
+Date: Mon Feb 13 14:58:26 CET 2012
+
+    VDPAU: Studio levels could be configured in the setup menu.
+    Window size defaults to fullscreen, if no geometry is given.
+
+User m.Rcu
+Date: Sun Feb 12 20:28:22 CET 2012
+
+    Jpeg screengrab use VDR RgbToJpeg function.
+
+User johns
+Date: Sun Feb 12 20:14:43 CET 2012
+
+    Add play/pause audio support.
+    Fix bug: audible glitch when switching AC-3 pass-through <-> none.
+    Fix bug: mpeg stills not displayed.
+    Detect audio stream type only after stream switch.
+    Detect more h264 streams with leading zeros.
+    VDPAU: support for studio levels added.
+    Add support for skip chroma deinterlace to software deinterlacer.
+    Type of software deinterlacer now configurable from setup menu.
+    Mixer channel could be set through command line option.
+    Fix bug: LFE moved to wrong position.
+    Guard suspend/resume against multiple calls.
+    Add support for AAC LATM audio streams.
+    Fix bug: alsa and ffmpeg use different channel layout.
+    Support more LPCM sample rates and number of channels.
+    Quick&dirty support for mpeg LPCM streams.
+    Workaround for text2skin undrawn OSD areas.
+    Detect dvb LPCM stream and ignore it.
+
 User johns
 Date: Thu Feb  2 23:29:35 CET 2012
 

Makefile

@@ -19,12 +19,13 @@ GIT_REV = $(shell git describe --always 2>/dev/null)
 ### Configuration (edit this for your needs)
 
 CONFIG := #-DDEBUG
-CONFIG += -DAV_INFO
-#CONFIG += -DHAVE_PTHREAD_NAME
+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 += $(shell ls /usr/lib/libjpeg* >/dev/null 2>&1 && echo "-DUSE_JPEG")
 CONFIG += -DUSE_OSS
 
 ### The C++ compiler and options:
@@ -32,8 +33,9 @@ CONFIG += -DUSE_OSS
 CC	 ?= gcc
 CXX	 ?= g++
 CFLAGS	 ?=	-g -O2 -W -Wall -Wextra -Winit-self \
-		-Wdeclaration-after-statement
-CXXFLAGS ?= -g -O2 -W -Wall -Wextra -Woverloaded-virtual
+		-Wdeclaration-after-statement \
+		-ftree-vectorize -msse3 -flax-vector-conversions
+CXXFLAGS ?= -g -O2 -W -Wall -Wextra -Werror=overloaded-virtual
 
 ### The directory environment:
 
@@ -43,7 +45,7 @@ TMPDIR ?= /tmp
 
 ### Make sure that necessary options are included:
 
-include $(VDRDIR)/Make.global
+-include $(VDRDIR)/Make.global
 
 ### Allow user defined options to overwrite defaults:
 
@@ -66,7 +68,7 @@ DEFINES += $(CONFIG) -D_GNU_SOURCE -DPLUGIN_NAME_I18N='"$(PLUGIN)"' \
 	$(if $(GIT_REV), -DGIT_REV='"$(GIT_REV)"')
 
 _CFLAGS = $(DEFINES) $(INCLUDES) \
-	$(shell pkg-config --cflags libavcodec libavformat) \
+	$(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` \
@@ -82,7 +84,7 @@ override CXXFLAGS += $(_CFLAGS)
 override CFLAGS	  += $(_CFLAGS)
 
 LIBS += -lrt \
-	$(shell pkg-config --libs libavcodec libavformat) \
+	$(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` \
@@ -91,9 +93,7 @@ LIBS += -lrt \
 	$(if $(findstring USE_VAAPI,$(CONFIG)), \
 		    `pkg-config --libs libva-x11 libva-glx libva`) \
 	$(if $(findstring USE_ALSA,$(CONFIG)), \
-	            `pkg-config --libs alsa`) \
-	$(if $(findstring USE_JPEG,$(CONFIG)), \
-	            -ljpeg)
+		    `pkg-config --libs alsa`)
 
 ### The object files (add further files here):
 

README.txt

@@ -20,24 +20,25 @@ $Id$
 
 A software and GPU emulated HD output device plugin for VDR.
 
-    o Video VA-API/VA-API (with intel, nvidia and amd backend supported)
-    o Video CPU/VA-API
-    o Video VDPAU/VDPAU
-    o Video CPU/VDPAU
+    o Video decoder CPU / VA-API / VDPAU
+    o Video output VA-API / VDPAU
     o Audio FFMpeg / Alsa / Analog
     o Audio FFMpeg / Alsa / Digital
     o Audio FFMpeg / OSS / Analog
-    o HDMI/SPDIF Passthrough
-    o VA-API bob software deinterlace
-    o Auto-crop
+    o HDMI/SPDIF pass-through
+    o Software volume, compression, normalize and channel resample
+    o YaepgHD support
+    o Software deinterlacer Bob (VA-API only)
+    o Autocrop
+    o Grab image (VDPAU only)
+    o Suspend
+    o Letterbox, Stretch and Center cut-out video display modes
+    o atmo light support with plugin http://github.com/durchflieger/DFAtmo
 
-    o planned: Video VA-API/Opengl
-    o planned: Video VDPAU/Opengl
-    o planned: Video CPU/Xv
-    o planned: Video CPU/Opengl
+    o planned: Video decoder VA-API Branch: vaapi-ext/staging
+    o planned: Video output XvBA / Opengl / Xv
+    o planned: VA-API grab image
     o planned: Improved Software Deinterlacer (yadif or/and ffmpeg filters)
-    o planned: Video XvBA/XvBA
-    o planned: atmo light support
 
 To compile you must have the 'requires' installed.
 
@@ -76,6 +77,13 @@ Setup:	environment
 
 	DISPLAY=:0.0
 		x11 display name
+	NO_HW=1
+		if set don't use the hardware decoders
+	NO_MPEG_HW=1
+		if set don't use the hardware decoder for mpeg1/2
+	STUDIO_LEVELS=1
+		if set use studio levels with vdpau (deprecated use setup)
+
     only if alsa is configured
 	ALSA_DEVICE=default
 		alsa PCM device name
@@ -85,6 +93,7 @@ Setup:	environment
 		alsa control device name
 	ALSA_MIXER_CHANNEL=PCM
 		alsa control channel name
+
     only if oss is configured
 	OSS_AUDIODEV=/dev/dsp
 		oss dsp device name
@@ -99,13 +108,18 @@ Setup: /etc/vdr/setup.conf
 ------
 	Following is supported:
 
-	softhddevice.MakePrimary = 1
+	softhddevice.MakePrimary = 0
 	0 = no change, 1 make softhddevice primary at start
 
 	softhddevice.HideMainMenuEntry = 0
 	0 = show softhddevice main menu entry, 1 = hide entry
 
-	<res> of the next parameters is 567i, 720p, 1080i_fake or 1080i.
+	softhddevice.Osd.Width = 0
+	0 = auto (=display, unscaled) n = fixed osd size scaled for display
+	softhddevice.Osd.Height = 0
+	0 = auto (=display, unscaled) n = fixed osd size scaled for display
+
+	<res> of the next parameters is 576i, 720p, 1080i_fake or 1080i.
 	1080i_fake is 1280x1080 or 1440x1080
 	1080i is "real" 1920x1080
 
@@ -119,6 +133,9 @@ Setup: /etc/vdr/setup.conf
 	softhddevice.<res>.SkipChromaDeinterlace = 0
 	0 = disabled, 1 = enabled (for slower cards, poor qualit<69>t)
 
+	softhddevice.<res>.InverseTelecine = 0
+	0 = disabled, 1 = enabled
+
 	softhddevice.<res>.Denoise = 0
 	0 .. 1000 noise reduction level (0 off, 1000 max)
 
@@ -126,14 +143,48 @@ Setup: /etc/vdr/setup.conf
 	-1000 .. 1000 noise reduction level (0 off, -1000 max blur,
 	    1000 max sharp)
 
+	softhddevice.<res>.CutTopBottom = 0
+	Cut 'n' pixels at at top and bottom of the video picture.
+
+	softhddevice.<res>.CutLeftRight = 0
+	Cut 'n' pixels at at left and right of the video picture.
+
 	softhddevice.AudioDelay = 0
 	+n or -n ms
+	delay audio or delay video
 
 	softhddevice.AudioPassthrough = 0
 	0 = none, 1 = AC-3
 
 	for AC-3 the pass-through device is used.
 
+	softhddevice.AudioDownmix = 0
+	0 = none, 1 = downmix
+	downmix AC-3 to stero.
+
+	softhddevice.AudioSoftvol = 0
+	0 = off, use hardware volume control
+	1 = on, use software volume control
+
+	softhddevice.AudioNormalize = 0
+	0 = off, 1 = enable audio normalize
+
+	softhddevice.AudioMaxNormalize = 0
+	maximal volume factor/1000 of the normalize filter
+
+	softhddevice.AudioCompression = 0
+	0 = off, 1 = enable audio compression
+
+	softhddevice.AudioMaxCompression = 0
+	maximal volume factor/1000 of the compression filter
+
+	softhddevice.AudioStereoDescent = 0
+	reduce volume level (/1000) for stereo sources
+
+	softhddevice.AudioBufferTime = 0
+	0 = default (336 ms)
+	1 - 1000 = size of the buffer in ms
+
 	softhddevice.AutoCrop.Interval = 0
 	0 disables auto-crop
 	n each 'n' frames auto-crop is checked.
@@ -146,8 +197,17 @@ Setup: /etc/vdr/setup.conf
 	if detected crop area is too small, cut max 'n' pixels at top and
 	bottom.
 
-	softhddevice.SkipLines = 0
-	skip 'n' lines at top and bottom of the video picture.
+	softhddevice.Background = 0
+	32bit RGBA background color
+	(Red * 16777216 +  Green * 65536 + Blue * 256 + Alpha)
+	or hex RRGGBBAA
+	grey 127 * 16777216 + 127 * 65536 + 127 * 256 => 2139062016
+	in the setup menu this is entered as (24bit RGB and 8bit Alpha)
+	(Red * 65536 +  Green * 256 + Blue)
+
+	softhddevice.StudioLevels = 0
+		0 use PC levels (0-255) with vdpau.
+		1 use studio levels (16-235) with vdpau.
 
 	softhddevice.Suspend.Close = 0
 	1 suspend closes x11 window, connection and audio device.
@@ -156,6 +216,23 @@ Setup: /etc/vdr/setup.conf
 	softhddevice.Suspend.X11 = 0
 	1 suspend stops X11 server (not working yet)
 
+	softhddevice.60HzMode = 0
+	0 disable 60Hz display mode
+	1 enable 60Hz display mode
+
+	softhddevice.SoftStartSync = 0
+	0 disable soft start of audio/video sync
+	1 enable soft start of audio/video sync
+
+	softhddevice.BlackPicture = 0
+	0 disable black picture during channel switch
+	1 enable black picture during channel switch
+
+	VideoDisplayFormat = ?
+	0 pan and scan
+	1 letter box
+	2 center cut-out
+
 Setup: /etc/vdr/remote.conf
 ------
 
@@ -189,8 +266,32 @@ Commandline:
 SVDRP:
 ------
 
-	Use 'svdrpsend.pl plug softhddevice HELP' to see the SVDRP commands
-	help and which are supported by the plugin.
+	Use 'svdrpsend.pl plug softhddevice HELP'
+	or 'svdrpsend plug softhddevice HELP' to see the SVDRP commands help
+	and which are supported by the plugin.
+
+Keymacros:
+----------
+
+	See keymacros.conf how to setup the macros.
+
+	This are the supported key sequences:
+
+	@softhddevice Blue 1 0		disable pass-through
+	@softhddevice Blue 1 1		enable pass-through
+	@softhddevice Blue 1 2		toggle pass-through
+	@softhddevice Blue 1 3		decrease audio delay by 10ms
+	@softhddevice Blue 1 4		increase audio delay by 10ms
+	@softhddevice Blue 2 0		disable fullscreen
+	@softhddevice Blue 2 1		enable fullscreen
+	@softhddevice Blue 2 2		toggle fullscreen
+	@softhddevice Blue 2 3		disable auto-crop
+	@softhddevice Blue 2 4		enable auto-crop
+	@softhddevice Blue 2 5		toggle auto-crop
+	@softhddevice Blue 3 0		stretch 4:3 to 16:9
+	@softhddevice Blue 3 1		letter box 4:3 in 16:9
+	@softhddevice Blue 3 2		center cut-out 4:3 to 16:9
+	@softhddevice Blue 3 9		rotate 4:3 to 16:9 zoom mode
 
 Running:
 --------
@@ -202,6 +303,11 @@ Warning:
 --------
 	libav is not supported, expect many bugs with it.
 
+Known Bugs:
+-----------
+	VA-API doesn't v-sync 1080i streams
+	vdr-image not working
+
 Requires:
 ---------
 	media-video/ffmpeg (version >=0.7)
@@ -228,6 +334,11 @@ Requires:
 	x11-libs/xvba-video
 		XVBA Backend for Video Acceleration (VA) API
 		http://www.freedesktop.org/wiki/Software/vaapi
+
+	x11-libs/libvdpau
+		VDPAU wrapper and trace libraries
+		http://www.freedesktop.org/wiki/Software/VDPAU
+
 	x11-libs/libxcb,
 		X C-language Bindings library
 		http://xcb.freedesktop.org

Todo

@@ -19,26 +19,26 @@ GNU Affero General Public License for more details.
 $Id: $
 
 missing:
-    software deinterlace (yadif, ...)
-    software decoder with software deinterlace
-    ITU BT601, ITU BT709 (HD), RGB studio levels (16-235)?
+    more software deinterlace (yadif, ...)
+    more software decoder with software deinterlace
     suspend output / energie saver: stop and restart X11
     suspend plugin didn't restore full-screen (is this wanted?)
     Option deinterlace off / deinterlace force!
-
-crash:
-    AudioPlayHandlerThread -> pthread_cond_wait
+    ColorSpace aren't configurable with the gui.
+    works for me: restart vdr not working, when started x11 was killed.
 
 video:
     subtitle not cleared
     subtitle could be asyncron
-    reduce warnings after channel switch
     grab image with hardware and better scaling support
-    suspendoutput didn't show logo or black pictures
-	(must detect video format to show image)
-    hard channel switch
-    skip lines not configurable from setup menu.
-    OSD can only be shown after some stream could be shown
+    yaepghd changed position is lost on channel switch
+    pause (live tv) has sometime problems with SAT1 HD Pro7 HD
+    radio show black background
+    radio no need to wait on video buffers
+    starting with radio and own X11 server, shows no video
+    some low-bandwidth tv channels have hiccups.
+    check start with 24Hz display rate
+    crash with ffmpeg without vaapi and vdpau.
 
 vdpau:
     software decoder path not working
@@ -52,8 +52,8 @@ libva:
     [drm:i915_hangcheck_elapsed] *ERROR* Hangcheck timer elapsed... GPU hung
     [drm:i915_wait_request] *ERROR* i915_wait_request returns -11 ...
 
-libva: branch vaapi-ext
-    add support for vaapi-ext
+libva: branch vaapi-ext / staging
+    add support for vaapi-ext / staging
 
 libva-intel-driver:
     deinterlace only supported with vaapi-ext
@@ -71,22 +71,23 @@ libva-vdpau-driver:
 libva-xvba-driver:
 
 x11:
-    disable screensaver
     skip multiple configure-notify, handle only the last one.
     support embedded mode
 
 audio:
-    write TS -> PES parser, which feeds audio before the next start packet
-    Combine alsa+oss ringbuffer code.
     Make alsa thread/polled and oss thread/polled output module runtime
     selectable.
-    software volume support
+    Mute should do a real mute and not only set volume to zero.
+    Starting suspended and muted, didn't register the mute.
+    Relaxed audio sync checks at end of packet and already in sync
+    samplerate problem resume/suspend.
+    only wait for video start, if video is running.
+    Not primary device, don't use and block audio/video.
+    multiple open of audio device, reduce them.
+    Not all channel conversions are written (f.e. 2->3 ... 5->6 ...)
 
 audio/alsa:
-    better downmix of >2 channels on 2 channel hardware
     remix support of unsupported sample rates
-    libav supports only resample of mono to 2 channels
-    ffmpeg didn't support resample of 5 to 2 channels
 
 audio/oss:
     alsa oss emulation mixer "pcm" not working
@@ -95,12 +96,13 @@ audio/oss:
 
 HDMI/SPDIF Passthrough:
     only AC-3 written
-    Channels are wrong setup, if changing setting during operation.
-    support oss pass-through
 
 playback of recording
-    pause is not reset, when replay exit
-    replay/pause need 100% cpu
+    pause is not reset, when replay exit (fixed?)
+    replay/pause need 100% cpu (fixed?)
+
+plugins:
+    mp3 plugin needs 100% cpu (bad ::Poll)
 
 setup:
     Setup of decoder type.
@@ -111,9 +113,8 @@ setup:
 	Can a notice be added to the setup menu?
 
 unsorted:
-    Menu -> Setup -> Plugins -> skingenigmang -> General
-	-> Try 8bpp single area: no, has missing parts.
     stoping vdr while plugin is suspended opens and closes a window.
+    svdrp prim: support plugin names for device numbers.
 
 future features (not planed for 1.0 - 1.5)
 
@@ -121,9 +122,8 @@ future features (not planed for 1.0 - 1.5)
     video out with opengl
     video out with xvba
     software decoder for xv / opengl
-    atmolight support
     multistream handling
     pip support
     save and use auto-crop with channel zapping
 
-    upmix stereo to AC-3
+    upmix stereo to AC-3 (supported by alsa plugin)

audio.h

@@ -31,20 +31,32 @@ extern void AudioEnqueue(const void *, int);	///< buffer audio samples
 extern void AudioFlushBuffers(void);	///< flush audio buffers
 extern void AudioPoller(void);		///< poll audio events/handling
 extern int AudioFreeBytes(void);	///< free bytes in audio output
-
-//extern int AudioUsedBytes(void);	///< used bytes in audio output
-extern uint64_t AudioGetDelay(void);	///< get current audio delay
+extern int AudioUsedBytes(void);	///< used bytes in audio output
+extern int64_t AudioGetDelay(void);	///< get current audio delay
 extern void AudioSetClock(int64_t);	///< set audio clock base
 extern int64_t AudioGetClock();		///< get current audio clock
 extern void AudioSetVolume(int);	///< set volume
 extern int AudioSetup(int *, int *, int);	///< setup audio output
 
-//extern void AudioPlay(void);		///< play audio
-//extern void AudioPause(void);		///< pause audio
+extern void AudioPlay(void);		///< play audio
+extern void AudioPause(void);		///< pause audio
+
+extern void AudioSetBufferTime(int);	///< set audio buffer time
+extern void AudioSetSoftvol(int);	///< enable/disable softvol
+extern void AudioSetNormalize(int, int);	///< set normalize parameters
+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 Passthrough device
+extern void AudioSetDeviceAC3(const char *);	///< set pass-through device
+extern void AudioSetChannel(const char *);	///< set mixer channel
 extern void AudioInit(void);		///< setup audio module
 extern void AudioExit(void);		///< cleanup and exit audio module
 
+//----------------------------------------------------------------------------
+//	Variables
+//----------------------------------------------------------------------------
+
+extern char AudioAlsaDriverBroken;	///< disable broken driver message
+
 /// @}

codec.c

@@ -30,13 +30,12 @@
 ///		many bugs and incompatiblity in it.  Don't use this shit.
 ///
 
-    /**
-    **	use av_parser to support insane dvb audio streams.
-    */
-#define USE_AVPARSER
-
-    /// compile with passthrough support (experimental)
+    /// compile with passthrough support (stable, ac3 only)
 #define USE_PASSTHROUGH
+    /// compile audio drift correction support (experimental)
+#define USE_AUDIO_DRIFT_CORRECTION
+    /// compile AC3 audio drift correction support (experimental)
+#define USE_AC3_DRIFT_CORRECTION
 
 #include <stdio.h>
 #include <unistd.h>
@@ -355,7 +354,7 @@ void CodecVideoOpen(VideoDecoder * decoder, const char *name, int codec_id)
 {
     AVCodec *video_codec;
 
-    Debug(3, "codec: using codec %s or ID %#04x\n", name, codec_id);
+    Debug(3, "codec: using video codec %s or ID %#06x\n", name, codec_id);
 
     if (decoder->VideoCtx) {
 	Error(_("codec: missing close\n"));
@@ -380,7 +379,7 @@ void CodecVideoOpen(VideoDecoder * decoder, const char *name, int codec_id)
     if (name && (video_codec = avcodec_find_decoder_by_name(name))) {
 	Debug(3, "codec: vdpau decoder found\n");
     } else if (!(video_codec = avcodec_find_decoder(codec_id))) {
-	Fatal(_("codec: codec ID %#04x not found\n"), codec_id);
+	Fatal(_("codec: codec ID %#06x not found\n"), codec_id);
 	// FIXME: none fatal
     }
     decoder->VideoCodec = video_codec;
@@ -561,6 +560,11 @@ void CodecVideoDecode(VideoDecoder * decoder, const AVPacket * avpkt)
 	    video_ctx->frame_number, used);
     }
     if (used != pkt->size) {
+	// ffmpeg 0.8.7 dislikes our seq_end_h264 and enters endless loop here
+	if (used == 0 && pkt->size == 5 && pkt->data[4] == 0x0A) {
+	    Warning("codec: ffmpeg 0.8.x workaround used\n");
+	    return;
+	}
 	if (used >= 0 && used < pkt->size) {
 	    // some tv channels, produce this
 	    Debug(4,
@@ -581,8 +585,10 @@ void CodecVideoDecode(VideoDecoder * decoder, const AVPacket * avpkt)
 */
 void CodecVideoFlushBuffers(VideoDecoder * decoder)
 {
+    if (decoder->VideoCtx) {
 	avcodec_flush_buffers(decoder->VideoCtx);
     }
+}
 
 //----------------------------------------------------------------------------
 //	Audio
@@ -603,8 +609,7 @@ struct _audio_decoder_
     AVCodec *AudioCodec;		///< audio codec
     AVCodecContext *AudioCtx;		///< audio codec context
 
-    /// audio parser to support insane dvb streaks
-    AVCodecParserContext *AudioParser;
+    int PassthroughAC3;			///< current ac-3 pass-through
     int SampleRate;			///< current stream sample rate
     int Channels;			///< current stream channels
 
@@ -613,15 +618,39 @@ struct _audio_decoder_
 
     ReSampleContext *ReSample;		///< audio resampling context
 
+    int64_t LastDelay;			///< last delay
+    struct timespec LastTime;		///< last time
+    int64_t LastPTS;			///< last PTS
+
+    int Drift;				///< accumulated audio drift
+    int DriftCorr;			///< audio drift correction value
+    int DriftFrac;			///< audio drift fraction for ac3
+
+    struct AVResampleContext *AvResample;	///< second audio resample context
+#define MAX_CHANNELS 8			///< max number of channels supported
+    int16_t *Buffer[MAX_CHANNELS];	///< deinterleave sample buffers
+    int BufferSize;			///< size of sample buffer
+    int16_t *Remain[MAX_CHANNELS];	///< filter remaining samples
+    int RemainSize;			///< size of remain buffer
+    int RemainCount;			///< number of remaining samples
 };
 
+#ifdef USE_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)
+#else
+
+static const int CodecPassthroughAC3 = 0;
 #endif
+static char CodecDownmix;		///< enable ac-3 downmix
 
 /**
 **	Allocate a new audio decoder context.
@@ -661,10 +690,12 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name,
 {
     AVCodec *audio_codec;
 
+    Debug(3, "codec: using audio codec %s or ID %#06x\n", name, codec_id);
+
     if (name && (audio_codec = avcodec_find_decoder_by_name(name))) {
 	Debug(3, "codec: audio decoder '%s' found\n", name);
     } else if (!(audio_codec = avcodec_find_decoder(codec_id))) {
-	Fatal(_("codec: codec ID %#04x not found\n"), codec_id);
+	Fatal(_("codec: codec ID %#06x not found\n"), codec_id);
 	// FIXME: errors aren't fatal
     }
     audio_decoder->AudioCodec = audio_codec;
@@ -672,6 +703,12 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name,
     if (!(audio_decoder->AudioCtx = avcodec_alloc_context3(audio_codec))) {
 	Fatal(_("codec: can't allocate audio codec context\n"));
     }
+
+    if (CodecDownmix) {
+	audio_decoder->AudioCtx->request_channels = 2;
+	audio_decoder->AudioCtx->request_channel_layout =
+	    AV_CH_LAYOUT_STEREO_DOWNMIX;
+    }
     pthread_mutex_lock(&CodecLockMutex);
     // open codec
 #if LIBAVCODEC_VERSION_INT <= AV_VERSION_INT(53,5,0)
@@ -680,27 +717,34 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name,
 	Fatal(_("codec: can't open audio codec\n"));
     }
 #else
-    if (avcodec_open2(audio_decoder->AudioCtx, audio_codec, NULL) < 0) {
+    if (1) {
+	AVDictionary *av_dict;
+
+	av_dict = NULL;
+	// FIXME: import settings
+	//av_dict_set(&av_dict, "dmix_mode", "0", 0);
+	//av_dict_set(&av_dict, "ltrt_cmixlev", "1.414", 0);
+	//av_dict_set(&av_dict, "loro_cmixlev", "1.414", 0);
+	if (avcodec_open2(audio_decoder->AudioCtx, audio_codec, &av_dict) < 0) {
 	    pthread_mutex_unlock(&CodecLockMutex);
 	    Fatal(_("codec: can't open audio codec\n"));
 	}
+	av_dict_free(&av_dict);
+    }
 #endif
     pthread_mutex_unlock(&CodecLockMutex);
     Debug(3, "codec: audio '%s'\n", audio_decoder->AudioCtx->codec_name);
 
     if (audio_codec->capabilities & CODEC_CAP_TRUNCATED) {
 	Debug(3, "codec: audio can use truncated packets\n");
-	// we do not send complete frames
-	audio_decoder->AudioCtx->flags |= CODEC_FLAG_TRUNCATED;
-    }
-    if (!(audio_decoder->AudioParser =
-	    av_parser_init(audio_decoder->AudioCtx->codec_id))) {
-	Fatal(_("codec: can't init audio parser\n"));
+	// we send only complete frames
+	// audio_decoder->AudioCtx->flags |= CODEC_FLAG_TRUNCATED;
     }
     audio_decoder->SampleRate = 0;
     audio_decoder->Channels = 0;
     audio_decoder->HwSampleRate = 0;
     audio_decoder->HwChannels = 0;
+    audio_decoder->LastDelay = 0;
 }
 
 /**
@@ -711,14 +755,25 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name,
 void CodecAudioClose(AudioDecoder * audio_decoder)
 {
     // FIXME: output any buffered data
+    if (audio_decoder->AvResample) {
+	int ch;
+
+	av_resample_close(audio_decoder->AvResample);
+	audio_decoder->AvResample = NULL;
+	audio_decoder->RemainCount = 0;
+	audio_decoder->BufferSize = 0;
+	audio_decoder->RemainSize = 0;
+	for (ch = 0; ch < MAX_CHANNELS; ++ch) {
+	    free(audio_decoder->Buffer[ch]);
+	    audio_decoder->Buffer[ch] = NULL;
+	    free(audio_decoder->Remain[ch]);
+	    audio_decoder->Remain[ch] = NULL;
+	}
+    }
     if (audio_decoder->ReSample) {
 	audio_resample_close(audio_decoder->ReSample);
 	audio_decoder->ReSample = NULL;
     }
-    if (audio_decoder->AudioParser) {
-	av_parser_close(audio_decoder->AudioParser);
-	audio_decoder->AudioParser = NULL;
-    }
     if (audio_decoder->AudioCtx) {
 	pthread_mutex_lock(&CodecLockMutex);
 	avcodec_close(audio_decoder->AudioCtx);
@@ -727,19 +782,384 @@ void CodecAudioClose(AudioDecoder * audio_decoder)
     }
 }
 
+/**
+**	Set audio drift correction.
+**
+**	@param mask	enable mask (PCM, AC3)
+*/
+void CodecSetAudioDrift(int mask)
+{
+#ifdef USE_AUDIO_DRIFT_CORRECTION
+    CodecAudioDrift = mask & 3;
+#endif
+    (void)mask;
+}
+
 /**
 **	Set audio pass-through.
+**
+**	@param mask	enable mask (PCM, AC3)
 */
 void CodecSetAudioPassthrough(int mask)
 {
 #ifdef USE_PASSTHROUGH
     CodecPassthroughAC3 = mask & 1 ? 1 : 0;
 #endif
-    // FIXME: must update audio decoder (nr. of channels wrong)
     (void)mask;
 }
 
-#ifdef USE_AVPARSER
+/**
+**	Set audio downmix.
+**
+**	@param onoff	enable/disable downmix.
+*/
+void CodecSetAudioDownmix(int onoff)
+{
+    if (onoff == -1) {
+	CodecDownmix ^= 1;
+	return;
+    }
+    CodecDownmix = onoff;
+}
+
+/**
+**	Reorder audio frame.
+**
+**	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 buf[IN,OUT]	sample buffer
+**	@param size		size of sample buffer in bytes
+**	@param channels		number of channels interleaved in sample buffer
+*/
+static void CodecReorderAudioFrame(int16_t * buf, int size, int channels)
+{
+    int i;
+    int c;
+    int ls;
+    int rs;
+    int lfe;
+
+    switch (channels) {
+	case 5:
+	    size /= 2;
+	    for (i = 0; i < size; i += 5) {
+		c = buf[i + 2];
+		ls = buf[i + 3];
+		rs = buf[i + 4];
+		buf[i + 2] = ls;
+		buf[i + 3] = rs;
+		buf[i + 4] = c;
+	    }
+	    break;
+	case 6:
+	    size /= 2;
+	    for (i = 0; i < size; i += 6) {
+		c = buf[i + 2];
+		lfe = buf[i + 3];
+		ls = buf[i + 4];
+		rs = buf[i + 5];
+		buf[i + 2] = ls;
+		buf[i + 3] = rs;
+		buf[i + 4] = c;
+		buf[i + 5] = lfe;
+	    }
+	    break;
+	case 8:
+	    size /= 2;
+	    for (i = 0; i < size; i += 8) {
+		c = buf[i + 2];
+		lfe = buf[i + 3];
+		ls = buf[i + 4];
+		rs = buf[i + 5];
+		buf[i + 2] = ls;
+		buf[i + 3] = rs;
+		buf[i + 4] = c;
+		buf[i + 5] = lfe;
+	    }
+	    break;
+    }
+}
+
+/**
+**	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_REALTIME, &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);
+	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:%5zdms 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 & 2) && (!CodecPassthroughAC3
+		|| audio_decoder->AudioCtx->codec_id != CODEC_ID_AC3)) {
+	    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;
+	}
+    }
+    // FIXME: this works with libav 0.8, and only with >10ms with ffmpeg 0.10
+    if (audio_decoder->AvResample && 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);
+	}
+	av_resample_compensate(audio_decoder->AvResample,
+	    audio_decoder->DriftCorr / 10, distance);
+    }
+    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)
+{
+    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;
+    }
+    if (audio_decoder->AvResample) {
+	av_resample_close(audio_decoder->AvResample);
+	audio_decoder->AvResample = NULL;
+	audio_decoder->RemainCount = 0;
+    }
+
+    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" : "");
+
+    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);
+
+	if (err == 1) {
+	    audio_decoder->ReSample =
+		av_audio_resample_init(audio_decoder->HwChannels,
+		audio_ctx->channels, audio_decoder->HwSampleRate,
+		audio_ctx->sample_rate, audio_ctx->sample_fmt,
+		audio_ctx->sample_fmt, 16, 10, 0, 0.8);
+	    // libav-0.8_pre didn't support 6 -> 2 channels
+	    if (!audio_decoder->ReSample) {
+		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;
+	}
+    }
+    // prepare audio drift resample
+#ifdef USE_AUDIO_DRIFT_CORRECTION
+    if ((CodecAudioDrift & 1) && !isAC3) {
+	if (audio_decoder->AvResample) {
+	    Error(_("codec/audio: overwrite resample\n"));
+	}
+	audio_decoder->AvResample =
+	    av_resample_init(audio_decoder->HwSampleRate,
+	    audio_decoder->HwSampleRate, 16, 10, 0, 0.8);
+	if (!audio_decoder->AvResample) {
+	    Error(_("codec/audio: AvResample setup error\n"));
+	} else {
+	    // reset drift to some default value
+	    audio_decoder->DriftCorr /= 2;
+	    audio_decoder->DriftFrac = 0;
+	    av_resample_compensate(audio_decoder->AvResample,
+		audio_decoder->DriftCorr / 10,
+		10 * audio_decoder->HwSampleRate);
+	}
+    }
+#endif
+}
+
+/**
+**	Codec enqueue audio samples.
+**
+**	@param audio_decoder	audio decoder data
+**	@param data		samples data
+**	@param count		number of bytes in sample data
+*/
+void CodecAudioEnqueue(AudioDecoder * audio_decoder, int16_t * data, int count)
+{
+#ifdef USE_AUDIO_DRIFT_CORRECTION
+    if ((CodecAudioDrift & 1) && 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];
+	int consumed;
+	int i;
+	int n;
+	int ch;
+	int bytes_n;
+
+	bytes_n = count / audio_decoder->HwChannels;
+	// resize sample buffer, if needed
+	if (audio_decoder->RemainCount + bytes_n > audio_decoder->BufferSize) {
+	    audio_decoder->BufferSize = audio_decoder->RemainCount + bytes_n;
+	    for (ch = 0; ch < MAX_CHANNELS; ++ch) {
+		audio_decoder->Buffer[ch] =
+		    realloc(audio_decoder->Buffer[ch],
+		    audio_decoder->BufferSize);
+	    }
+	}
+	// copy remaining bytes into sample buffer
+	for (ch = 0; ch < audio_decoder->HwChannels; ++ch) {
+	    memcpy(audio_decoder->Buffer[ch], audio_decoder->Remain[ch],
+		audio_decoder->RemainCount);
+	}
+	// deinterleave samples into sample buffer
+	for (i = 0; i < bytes_n / 2; i++) {
+	    for (ch = 0; ch < audio_decoder->HwChannels; ++ch) {
+		audio_decoder->Buffer[ch][audio_decoder->RemainCount / 2 + i]
+		    = data[i * audio_decoder->HwChannels + ch];
+	    }
+	}
+
+	bytes_n += audio_decoder->RemainSize;
+	n = 0;				// keep gcc lucky
+	// resample the sample buffer into tmp buffer
+	for (ch = 0; ch < audio_decoder->HwChannels; ++ch) {
+	    n = av_resample(audio_decoder->AvResample, buftmp[ch],
+		audio_decoder->Buffer[ch], &consumed, bytes_n / 2,
+		sizeof(buftmp[ch]) / 2, ch == audio_decoder->HwChannels - 1);
+	    // fixme remaining channels
+	    if (bytes_n - consumed * 2 > audio_decoder->RemainSize) {
+		audio_decoder->RemainSize = bytes_n - consumed * 2;
+	    }
+	    audio_decoder->Remain[ch] =
+		realloc(audio_decoder->Remain[ch], audio_decoder->RemainSize);
+	    memcpy(audio_decoder->Remain[ch],
+		audio_decoder->Buffer[ch] + consumed,
+		audio_decoder->RemainSize);
+	    audio_decoder->RemainCount = audio_decoder->RemainSize;
+	}
+
+	// interleave samples from sample buffer
+	for (i = 0; i < n; i++) {
+	    for (ch = 0; ch < audio_decoder->HwChannels; ++ch) {
+		buf[i * audio_decoder->HwChannels + ch] = buftmp[ch][i];
+	    }
+	}
+	n *= 2;
+
+	n *= audio_decoder->HwChannels;
+	CodecReorderAudioFrame(buf, n, audio_decoder->HwChannels);
+	AudioEnqueue(buf, n);
+	return;
+    }
+#endif
+    CodecReorderAudioFrame(data, count, audio_decoder->HwChannels);
+    AudioEnqueue(data, count);
+}
 
 /**
 **	Decode an audio packet.
@@ -753,119 +1173,42 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
 {
     int16_t buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 +
 	FF_INPUT_BUFFER_PADDING_SIZE] __attribute__ ((aligned(16)));
+    int buf_sz;
+    int l;
     AVCodecContext *audio_ctx;
-    int index;
-
-//#define spkt avpkt
-#if 1					// didn't fix crash in av_parser_parse2
-    AVPacket spkt[1];
-
-    // av_new_packet reserves FF_INPUT_BUFFER_PADDING_SIZE and clears it
-    if (av_new_packet(spkt, avpkt->size)) {
-	Error(_("codec: out of memory\n"));
-	return;
-    }
-    memcpy(spkt->data, avpkt->data, avpkt->size);
-    spkt->pts = avpkt->pts;
-    spkt->dts = avpkt->dts;
-#endif
-#ifdef DEBUG
-    if (!audio_decoder->AudioParser) {
-	Fatal(_("codec: internal error parser freeded while running\n"));
-    }
-#endif
 
     audio_ctx = audio_decoder->AudioCtx;
-    index = 0;
-    while (spkt->size > index) {
-	int n;
-	int l;
-	AVPacket dpkt[1];
 
-	av_init_packet(dpkt);
-	n = av_parser_parse2(audio_decoder->AudioParser, audio_ctx,
-	    &dpkt->data, &dpkt->size, spkt->data + index, spkt->size - index,
-	    !index ? (uint64_t) spkt->pts : AV_NOPTS_VALUE,
-	    !index ? (uint64_t) spkt->dts : AV_NOPTS_VALUE, -1);
-
-	// FIXME: make this a function for both #ifdef cases
-	if (dpkt->size) {
-	    int buf_sz;
-
-	    dpkt->pts = audio_decoder->AudioParser->pts;
-	    dpkt->dts = audio_decoder->AudioParser->dts;
     buf_sz = sizeof(buf);
-	    l = avcodec_decode_audio3(audio_ctx, buf, &buf_sz, dpkt);
+    l = avcodec_decode_audio3(audio_ctx, buf, &buf_sz, (AVPacket *) avpkt);
+    if (avpkt->size != l) {
+	if (l == AVERROR(EAGAIN)) {
+	    Error(_("codec: latm\n"));
+	    return;
+	}
 	if (l < 0) {			// no audio frame could be decompressed
-		Error(_("codec: error audio data at %d\n"), index);
-		break;
+	    Error(_("codec: error audio data\n"));
+	    return;
+	}
+	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, dpkt);
+    avcodec_decode_audio4(audio_ctx, frame, &got_frame, avpkt);
 #else
 #endif
-	    // Update audio clock
-	    if ((uint64_t) dpkt->pts != AV_NOPTS_VALUE) {
-		AudioSetClock(dpkt->pts);
+
+    // 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->SampleRate != audio_ctx->sample_rate
+    if (audio_decoder->PassthroughAC3 != CodecPassthroughAC3
+	|| audio_decoder->SampleRate != audio_ctx->sample_rate
 	|| audio_decoder->Channels != audio_ctx->channels) {
-		int err;
-		int isAC3;
-
-		if (audio_decoder->ReSample) {
-		    audio_resample_close(audio_decoder->ReSample);
-		    audio_decoder->ReSample = NULL;
-		}
-
-		audio_decoder->SampleRate = audio_ctx->sample_rate;
-		audio_decoder->HwSampleRate = audio_ctx->sample_rate;
-		audio_decoder->Channels = audio_ctx->channels;
-#ifdef USE_PASSTHROUGH
-		// SPDIF/HDMI passthrough
-		if (CodecPassthroughAC3 && audio_ctx->codec_id == CODEC_ID_AC3) {
-		    audio_decoder->HwChannels = 2;
-		    isAC3 = 1;
-		} else
-#endif
-		{
-		    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);
-
-		    if (err == 1) {
-			audio_decoder->ReSample =
-			    av_audio_resample_init(audio_decoder->HwChannels,
-			    audio_ctx->channels, audio_decoder->HwSampleRate,
-			    audio_ctx->sample_rate, audio_ctx->sample_fmt,
-			    audio_ctx->sample_fmt, 16, 10, 0, 0.8);
-			// libav-0.8_pre didn't support 6 -> 2 channels
-			if (!audio_decoder->ReSample) {
-			    Error(_("codec/audio: resample setup error\n"));
-			    audio_decoder->HwChannels = 0;
-			    audio_decoder->HwSampleRate = 0;
-			}
-		    } else {
-			Debug(3, "codec/audio: audio setup error\n");
-			// FIXME: handle errors
-			audio_decoder->HwChannels = 0;
-			audio_decoder->HwSampleRate = 0;
-			break;
-		    }
-		}
+	CodecAudioUpdateFormat(audio_decoder);
     }
 
     if (audio_decoder->HwSampleRate && audio_decoder->HwChannels) {
@@ -878,8 +1221,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
 
 	    // FIXME: libav-0.7.2 crash here
 	    outlen =
-			audio_resample(audio_decoder->ReSample, outbuf, buf,
-			buf_sz);
+		audio_resample(audio_decoder->ReSample, outbuf, buf, buf_sz);
 #ifdef DEBUG
 	    if (outlen != buf_sz) {
 		Debug(3, "codec/audio: possible fixed ffmpeg\n");
@@ -893,30 +1235,52 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
 		    audio_decoder->HwChannels *
 		    av_get_bytes_per_sample(audio_ctx->sample_fmt);
 		Debug(4, "codec/audio: %d -> %d\n", buf_sz, outlen);
-			AudioEnqueue(outbuf, outlen);
+		CodecAudioEnqueue(audio_decoder, outbuf, outlen);
 	    }
 	} else {
 #ifdef USE_PASSTHROUGH
 	    // SPDIF/HDMI passthrough
-		    if (CodecPassthroughAC3
-			&& audio_ctx->codec_id == CODEC_ID_AC3) {
+	    if (CodecPassthroughAC3 && audio_ctx->codec_id == CODEC_ID_AC3) {
 		// build SPDIF header and append A52 audio to it
-			// dpkt is the original data
+		// avpkt is the original data
 		buf_sz = 6144;
-			if (buf_sz < dpkt->size + 8) {
+
+#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"));
-			    break;
+		    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 | (dpkt->data[5] & 0x07) << 8);
-			buf[3] = htole16(dpkt->size * 8);
-			swab(dpkt->data, buf + 4, dpkt->size);
-			memset(buf + 4 + dpkt->size / 2, 0,
-			    buf_sz - 8 - dpkt->size);
+		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);
+		return;
 	    }
 #if 0
 	    //
@@ -931,7 +1295,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
 		buf[2] = 0x1F;
 		buf[3] = 0x4E;
 		buf[4] = 0x00;
-			switch (dpkt->size) {
+		switch (avpkt->size) {
 		    case 512:
 			buf[5] = 0x0B;
 			break;
@@ -947,15 +1311,14 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
 			buf[5] = 0x00;
 			break;
 		}
-			buf[6] = (dpkt->size * 8);
-			buf[7] = (dpkt->size * 8) >> 8;
+		buf[6] = (avpkt->size * 8);
+		buf[7] = (avpkt->size * 8) >> 8;
 		//buf[8] = 0x0B;
 		//buf[9] = 0x77;
-			//printf("%x %x\n", dpkt->data[0],dpkt->data[1]);
+		//printf("%x %x\n", avpkt->data[0],avpkt->data[1]);
 		// swab?
-			memcpy(buf + 8, dpkt->data, dpkt->size);
-			memset(buf + 8 + dpkt->size, 0,
-			    buf_sz - 8 - dpkt->size);
+		memcpy(buf + 8, avpkt->data, avpkt->size);
+		memset(buf + 8 + avpkt->size, 0, buf_sz - 8 - avpkt->size);
 	    } else if (1) {
 		// FIXME: need to detect mp2
 		// FIXME: mp2 passthrough
@@ -971,96 +1334,10 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
 	    // True HD?
 #endif
 #endif
-		    AudioEnqueue(buf, buf_sz);
+	    CodecAudioEnqueue(audio_decoder, buf, buf_sz);
 	}
     }
-
-	    if (dpkt->size > l) {
-		Error(_("codec: error more than one frame data\n"));
 }
-	}
-
-	index += n;
-    }
-
-#if 1
-    // or av_free_packet, make no difference here
-    av_destruct_packet(spkt);
-#endif
-}
-
-#else
-
-/**
-**	Decode an audio packet.
-**
-**	PTS must be handled self.
-**
-**	@param audio_decoder	audio decoder data
-**	@param avpkt		audio packet
-*/
-void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
-{
-    int16_t buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 +
-	FF_INPUT_BUFFER_PADDING_SIZE] __attribute__ ((aligned(16)));
-    AVCodecContext *audio_ctx;
-    int index;
-
-//#define spkt avpkt
-#if 1
-    AVPacket spkt[1];
-
-    // av_new_packet reserves FF_INPUT_BUFFER_PADDING_SIZE and clears it
-    if (av_new_packet(spkt, avpkt->size)) {
-	Error(_("codec: out of memory\n"));
-	return;
-    }
-    memcpy(spkt->data, avpkt->data, avpkt->size);
-    spkt->pts = avpkt->pts;
-    spkt->dts = avpkt->dts;
-#endif
-    audio_ctx = audio_decoder->AudioCtx;
-    index = 0;
-    while (spkt->size > index) {
-	int n;
-	int buf_sz;
-	AVPacket dpkt[1];
-
-	av_init_packet(dpkt);
-	dpkt->data = spkt->data + index;
-	dpkt->size = spkt->size - index;
-
-	buf_sz = sizeof(buf);
-	n = avcodec_decode_audio3(audio_ctx, buf, &buf_sz, dpkt);
-	if (n < 0) {			// no audio frame could be decompressed
-	    Error(_("codec: error audio data at %d\n"), index);
-	    break;
-	}
-#ifdef DEBUG
-	Debug(4, "codec/audio: -> %d\n", buf_sz);
-	if ((unsigned)buf_sz > sizeof(buf)) {
-	    abort();
-	}
-#endif
-#ifdef notyetFF_API_OLD_DECODE_AUDIO
-	// FIXME: ffmpeg git comeing
-	int got_frame;
-
-	avcodec_decode_audio4(audio_ctx, frame, &got_frame, dpkt);
-#else
-#endif
-	// FIXME: see above, old code removed
-
-	index += n;
-    }
-
-#if 1
-    // or av_free_packet, make no difference here
-    av_destruct_packet(spkt);
-#endif
-}
-
-#endif
 
 /**
 **	Flush the audio decoder.
@@ -1069,7 +1346,6 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
 */
 void CodecAudioFlushBuffers(AudioDecoder * decoder)
 {
-    // FIXME: reset audio parser
     avcodec_flush_buffers(decoder->AudioCtx);
 }
 

codec.h

@@ -67,6 +67,15 @@ extern void CodecAudioOpen(AudioDecoder *, const char *, int);
     /// Close audio codec.
 extern void CodecAudioClose(AudioDecoder *);
 
+    /// Set audio drift correction.
+extern void CodecSetAudioDrift(int);
+
+    /// Set audio pass-through.
+extern void CodecSetAudioPassthrough(int);
+
+    /// Set audio downmix.
+extern void CodecSetAudioDownmix(int);
+
     /// Decode an audio packet.
 extern void CodecAudioDecode(AudioDecoder *, const AVPacket *);
 

misc.h

@@ -107,6 +107,31 @@ static inline void Syslog(const int level, const char *format, ...)
 #define Debug(level, fmt...)		/* disabled */
 #endif
 
+#ifndef AV_NOPTS_VALUE
+#define AV_NOPTS_VALUE INT64_C(0x8000000000000000)
+#endif
+
+/**
+**	Nice time-stamp string.
+**
+**	@param ts	dvb time stamp
+*/
+static inline const char *Timestamp2String(int64_t ts)
+{
+    static char buf[4][16];
+    static int idx;
+
+    if (ts == (int64_t) AV_NOPTS_VALUE) {
+	return "--:--:--.---";
+    }
+    idx = (idx + 1) % 3;
+    snprintf(buf[idx], sizeof(buf[idx]), "%2d:%02d:%02d.%03d",
+	(int)(ts / (90 * 3600000)), (int)((ts / (90 * 60000)) % 60),
+	(int)((ts / (90 * 1000)) % 60), (int)((ts / 90) % 1000));
+
+    return buf[idx];
+}
+
 /**
 **	Get ticks in ms.
 **

ringbuffer.c

@@ -48,6 +48,18 @@ struct _ring_buffer_
     atomic_t Filled;			///< how many of the buffer is used
 };
 
+/**
+**	Reset ring buffer pointers.
+**
+**	@param rb	Ring buffer to reset read/write pointers.
+*/
+void RingBufferReset(RingBuffer * rb)
+{
+    rb->ReadPointer = rb->Buffer;
+    rb->WritePointer = rb->Buffer;
+    atomic_set(&rb->Filled, 0);
+}
+
 /**
 **	Allocate a new ring buffer.
 **
@@ -69,10 +81,8 @@ RingBuffer *RingBufferNew(size_t size)
     }
 
     rb->Size = size;
-    rb->ReadPointer = rb->Buffer;
-    rb->WritePointer = rb->Buffer;
     rb->BufferEnd = rb->Buffer + size;
-    atomic_set(&rb->Filled, 0);
+    RingBufferReset(rb);
 
     return rb;
 }
@@ -89,7 +99,7 @@ void RingBufferDel(RingBuffer * rb)
 /**
 **	Advance write pointer in ring buffer.
 **
-**	@param rb	Ring buffer to adance write pointer.
+**	@param rb	Ring buffer to advance write pointer.
 **	@param cnt	Number of bytes to be adavanced.
 **
 **	@returns	Number of bytes that could be advanced in ring buffer.
@@ -198,7 +208,7 @@ size_t RingBufferGetWritePointer(RingBuffer * rb, void **wp)
 /**
 **	Advance read pointer in ring buffer.
 **
-**	@param rb	Ring buffer to adance read pointer.
+**	@param rb	Ring buffer to advance read pointer.
 **	@param cnt	Number of bytes to be advanced.
 **
 **	@returns	Number of bytes that could be advanced in ring buffer.

ringbuffer.h

@@ -23,13 +23,16 @@
 /// @addtogroup Ringbuffer
 /// @{
 
-    ///< ring buffer typedef
+    /// ring buffer typedef
 typedef struct _ring_buffer_ RingBuffer;
 
-    ///< create new ring buffer
+    /// reset ring buffer pointers
+extern void RingBufferReset(RingBuffer *);
+
+    /// create new ring buffer
 extern RingBuffer *RingBufferNew(size_t);
 
-    ///< free ring buffer
+    /// free ring buffer
 extern void RingBufferDel(RingBuffer *);
 
     /// write into ring buffer

softhddev.h

@@ -37,8 +37,8 @@ extern "C"
 
     /// C plugin play audio packet
     extern int PlayAudio(const uint8_t *, int, uint8_t);
-    /// C plugin mute audio
-    extern void Mute(void);
+    /// C plugin play TS audio packet
+    extern int PlayTsAudio(const uint8_t *, int);
     /// C plugin set audio volume
     extern void SetVolumeDevice(int);
 
@@ -50,13 +50,19 @@ extern "C"
     extern uint8_t *GrabImage(int *, int, int, int, int);
 
     /// C plugin set play mode
-    extern void SetPlayMode(void);
+    extern int SetPlayMode(int);
+    /// C plugin get current system time counter
+    extern int64_t GetSTC(void);
+    /// C plugin set trick speed
+    extern void TrickSpeed(int);
     /// C plugin clears all video and audio data from the device
     extern void Clear(void);
     /// C plugin sets the device into play mode
     extern void Play(void);
     /// C plugin sets the device into "freeze frame" mode
     extern void Freeze(void);
+    /// C plugin mute audio
+    extern void Mute(void);
     /// C plugin display I-frame as a still picture.
     extern void StillPicture(const uint8_t *, int);
     /// C plugin poll if ready
@@ -72,9 +78,11 @@ extern "C"
     /// C plugin exit + cleanup
     extern void SoftHdDeviceExit(void);
     /// C plugin start code
-    extern void Start(void);
+    extern int Start(void);
     /// C plugin stop code
     extern void Stop(void);
+    /// C plugin house keeping
+    extern void Housekeeping(void);
     /// C plugin main thread hook
     extern void MainThreadHook(void);
 
@@ -82,6 +90,9 @@ extern "C"
     extern void Suspend(int, int, int);
     /// Resume plugin
     extern void Resume(void);
+
+    /// Get decoder statistics
+    extern void GetStats(int *, int *, int *, int *);
 #ifdef __cplusplus
 }
 #endif

softhddevice_service.h

@@ -0,0 +1,44 @@
+///
+///	@file softhddev_service.h @brief software HD device service header file.
+///
+///	Copyright (c) 2012 by durchflieger.  All Rights Reserved.
+///
+///	Contributor(s):
+///
+///	License: AGPLv3
+///
+///	This program is free software: you can redistribute it and/or modify
+///	it under the terms of the GNU Affero General Public License as
+///	published by the Free Software Foundation, either version 3 of the
+///	License.
+///
+///	This program is distributed in the hope that it will be useful,
+///	but WITHOUT ANY WARRANTY; without even the implied warranty of
+///	MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+///	GNU Affero General Public License for more details.
+///
+///	$Id$
+//////////////////////////////////////////////////////////////////////////////
+
+#pragma once
+
+#define ATMO_GRAB_SERVICE	"SoftHDDevice-AtmoGrabService-v1.0"
+
+enum
+{ GRAB_IMG_RGBA_FORMAT_B8G8R8A8 };
+
+typedef struct
+{
+    int structSize;
+
+    // request data
+    int analyseSize;
+    int clippedOverscan;
+
+    // reply data
+    int imgType;
+    int imgSize;
+    int width;
+    int height;
+    void *img;
+} SoftHDDevice_AtmoGrabService_v1_0_t;

vdr-softhddevice-9999.ebuild

@@ -23,19 +23,19 @@ SLOT="0"
 KEYWORDS="~x86 ~amd64"
 IUSE="vaapi vdpau alsa oss yaepg opengl jpeg"
 
-DEPEND=">=x11-libs/libxcb-1.7
+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 )
-		>=media-video/ffmpeg-0.7
+		>=virtual/ffmpeg-0.7
 		sys-devel/gettext
 		sys-devel/make
 		dev-util/pkgconfig
-		yaepg? ( >=media-video/vdr-1.7[yaepg] )
-		!yaepg? ( >=media-video/vdr-1.7 )
+		yaepg? ( >=media-video/vdr-1.7.23[yaepg] )
+		!yaepg? ( >=media-video/vdr-1.7.23 )
 		vdpau? ( x11-libs/libvdpau )
 		vaapi? ( x11-libs/libva )
 		alsa? ( media-libs/alsa-lib )
@@ -64,10 +64,11 @@ src_compile() {
 src_install() {
 		vdr-plugin_src_install
 
-		dodir /etc/vdr/plugins || die
+		dodoc README.txt
 
-		insinto /etc/vdr/plugins
-		fowners -R vdr:vdr /etc/vdr || die
+		#dodir /etc/vdr/plugins || die
+		#insinto /etc/vdr/plugins
+		#fowners -R vdr:vdr /etc/vdr || die
 
 		#insinto /etc/conf.d
 		#doins vdr.softhddevice

video.h

@@ -30,6 +30,14 @@
     /// Video hardware decoder typedef
 typedef struct _video_hw_decoder_ VideoHwDecoder;
 
+//----------------------------------------------------------------------------
+//	Variables
+//----------------------------------------------------------------------------
+
+extern char VideoHardwareDecoder;	///< flag use hardware decoder
+extern char VideoIgnoreRepeatPict;	///< disable repeat pict warning
+extern int VideoAudioDelay;		///< audio/video delay
+
 //----------------------------------------------------------------------------
 //	Prototypes
 //----------------------------------------------------------------------------
@@ -71,9 +79,21 @@ extern void VideoPollEvent(void);
     /// Wakeup display handler.
 extern void VideoDisplayWakeup(void);
 
+    /// Set video device.
+extern void VideoSetDevice(const char *);
+
     /// Set video geometry.
 extern int VideoSetGeometry(const char *);
 
+    /// Set 60Hz display mode.
+extern void VideoSet60HzMode(int);
+
+    /// Set soft start audio/video sync.
+extern void VideoSetSoftStartSync(int);
+
+    /// Set show black picture during channel switch.
+extern void VideoSetBlackPicture(int);
+
     /// Set video output position.
 extern void VideoSetOutputPosition(int, int, int, int);
 
@@ -92,6 +112,9 @@ extern void VideoSetDeinterlace(int[]);
     /// Set skip chroma deinterlace.
 extern void VideoSetSkipChromaDeinterlace(int[]);
 
+    /// Set inverse telecine.
+extern void VideoSetInverseTelecine(int[]);
+
     /// Set scaling.
 extern void VideoSetScaling(int[]);
 
@@ -101,8 +124,17 @@ extern void VideoSetDenoise(int[]);
     /// Set sharpen.
 extern void VideoSetSharpen(int[]);
 
-    /// Set skip lines.
-extern void VideoSetSkipLines(int);
+    /// Set cut top and bottom.
+extern void VideoSetCutTopBottom(int[]);
+
+    /// Set cut left and right.
+extern void VideoSetCutLeftRight(int[]);
+
+    /// Set studio levels.
+extern void VideoSetStudioLevels(int);
+
+    /// Set background.
+extern void VideoSetBackground(uint32_t);
 
     /// Set audio delay.
 extern void VideoSetAudioDelay(int);
@@ -119,18 +151,41 @@ extern void VideoOsdDrawARGB(int, int, int, int, const uint8_t *);
     /// Get OSD size.
 extern void VideoGetOsdSize(int *, int *);
 
-extern int64_t VideoGetClock(void);	///< Get video clock.
+    /// Set OSD size.
+extern void VideoSetOsdSize(int, int);
+
+    /// Set video clock.
+extern void VideoSetClock(VideoHwDecoder *, int64_t);
+
+    /// Get video clock.
+extern int64_t VideoGetClock(const VideoHwDecoder *);
+
+    /// Set closing flag.
+extern void VideoSetClosing(VideoHwDecoder *);
+
+    /// Reset start of frame counter
+extern void VideoResetStart(VideoHwDecoder *);
+
+    /// Set trick play speed.
+extern void VideoSetTrickSpeed(VideoHwDecoder *, int);
 
     /// Grab screen.
 extern uint8_t *VideoGrab(int *, int *, int *, int);
 
+    /// Grab screen raw.
+extern uint8_t *VideoGrabService(int *, int *, int *);
+
+    /// Get decoder statistics.
+extern void VideoGetStats(VideoHwDecoder *, int *, int *, int *, int *);
+
 extern void VideoOsdInit(void);		///< Setup osd.
 extern void VideoOsdExit(void);		///< Cleanup osd.
 
 extern void VideoInit(const char *);	///< Setup video module.
 extern void VideoExit(void);		///< Cleanup and exit video module.
 
-extern void VideoFlushInput(void);	///< Flush video input buffers.
-extern int VideoDecode(void);		///< Decode video input buffers.
+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.
 
 /// @}