Makes audio ts parser default. Suspend fixes.

Experimental ac3 audio drift correction support.
Removes LPCM detection from TS parser.
2023-10-10 17:16:51 +00:00 · 2012-03-03 18:47:07 +01:00 · 2012-03-03 16:45:59 +01:00 · 2012-03-03 16:11:38 +01:00 · 2012-03-02 18:17:51 +01:00 · 2012-03-02 18:16:20 +01:00
14 changed files with 4818 additions and 1653 deletions
--- a/110
+++ b/110
@@ -1,6 +1,116 @@
 User johns
 Date:

+    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
+
+    Release Version 0.4.6
+    Warn only on the first duplicated frame in sequence.
+    Increase audio buffer, if bigger audio delay is used.
+    Makes SkipLines configure in setup menu.
+    Auto-crop only enabled with normal 4:3 display mode.
+    Vaapi updates OSD when cropping changes.
+    Add A-V info output and compile time option.
+    Fix bug: VA-API intel software decoder broken by aspect commit.
+    Add support for 4:3 output modes.
+    Quicker auto-crop after channel switch.
+    Add auto-crop support for Intel VA-API backend.
+    Fix bug: Auto-Crop logo skip didn't use displayed width.
+    Workaround for mpeg2 FFMpeg + VA-API + Intel GPU hung.
+    Fix bug: Missing vaSyncSurface and vaDestroyImage.
+    Fix bug: Only black picture with VA-API hw decoder.
+
+User HelAu
+Date: Mon Jan 30 16:54:47 CET 2012
+
+    Add support to start the plugin in suspended mode.
+
+User johns
+Date: Mon Jan 30 15:58:21 CET 2012
+
+    Finished rewrite of video code, to support output modules.
+    Add aspect change support to software decoder path.
+    Repair software decoder with vaapi vdpau backend.
+    Add workaround for Intel VA-API MPEG GPU hung.
+
+User johns
+Date: Sat Jan 28 13:32:12 CET 2012
+
    Release Version 0.4.5
    Add configurable skip lines at video top and bottom.
    Add auto-crop tolerance configuration.
--- a/43
+++ b/43
@@ -19,19 +19,24 @@ GIT_REV = $(shell git describe --always 2>/dev/null)
 ### Configuration (edit this for your needs)

 CONFIG := #-DDEBUG
-#CONFIG += -DHAVE_PTHREAD_NAME
+#CONFIG += -DUSE_AUDIO_DRIFT_CORRECTION	# build new audio drift code
+#CONFIG += -DUSE_AC3_DRIFT_CORRECTION	# build new ac-3 drift code
+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:

-CC       ?= gcc
-CXX      ?= g++
-CFLAGS   ?=	-g -O2 -W -Wall -Wextra -Winit-self \
-		-Wdeclaration-after-statement
+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 -Woverloaded-virtual

 ### The directory environment:
@@ -65,34 +70,32 @@ 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` \
 	$(if $(findstring USE_VDPAU,$(CONFIG)), \
-	            `pkg-config --cflags vdpau`) \
+		    `pkg-config --cflags vdpau`) \
 	$(if $(findstring USE_VAAPI,$(CONFIG)), \
-	            `pkg-config --cflags libva-x11 libva-glx libva`) \
+		    `pkg-config --cflags libva-x11 libva-glx libva`) \
 	$(if $(findstring USE_ALSA,$(CONFIG)), \
-	            `pkg-config --cflags alsa`)
+		    `pkg-config --cflags alsa`)

 #override _CFLAGS  += -Werror
 override CXXFLAGS += $(_CFLAGS)
-override CFLAGS   += $(_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` \
 	$(if $(findstring USE_VDPAU,$(CONFIG)), \
-	            `pkg-config --libs vdpau`) \
+		    `pkg-config --libs vdpau`) \
 	$(if $(findstring USE_VAAPI,$(CONFIG)), \
-	            `pkg-config --libs libva-x11 libva-glx libva`) \
+		    `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):

@@ -121,11 +124,11 @@ $(OBJS): Makefile

 ### Internationalization (I18N):

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

 %.mo: %.po
 	msgfmt -c -o $@ $<
--- a/README.txt
+++ b/README.txt
@@ -76,6 +76,13 @@ 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 +92,7 @@ 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
@@ -105,7 +113,7 @@ Setup: /etc/vdr/setup.conf
 	softhddevice.HideMainMenuEntry = 0
 	0 = show softhddevice main menu entry, 1 = hide entry

-	<res> of the next parameters is 567i, 720p, 1080i_fake or 1080i.
+	<res> of the next parameters is 576i, 720p, 1080i_fake or 1080i.
 	1080i_fake is 1280x1080 or 1440x1080
 	1080i is "real" 1920x1080

@@ -119,6 +127,9 @@ 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)

@@ -128,12 +139,17 @@ Setup: /etc/vdr/setup.conf

 	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.AutoCrop.Interval = 0
 	0 disables auto-crop
 	n each 'n' frames auto-crop is checked.
@@ -146,6 +162,18 @@ Setup: /etc/vdr/setup.conf
 	if detected crop area is too small, cut max 'n' pixels at top and
 	bottom.

+	softhddevice.Background = 0
+	32bit RGBA background color
+	(Red * 16777216 +  Green * 65536 + Blue * 256 + Alpha)
+	or hex RRGGBBAA
+	grey = 2155905279
+
+	softhddevice.SkipLines = 0
+	skip 'n' lines at top and bottom of the video picture.
+
+	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.
@@ -154,6 +182,15 @@ 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
+
+	VideoDisplayFormat = ?
+	0 pan and scan
+	1 letter box
+	2 center cut-out
+
 Setup: /etc/vdr/remote.conf
 ------

@@ -187,8 +224,20 @@ 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

 Running:
 --------
--- a/51
+++ b/51
@@ -21,11 +21,12 @@ $Id: $
 missing:
    software deinterlace (yadif, ...)
    software decoder with software deinterlace
-    zoom/fit-zoom 4:3 (SetVideoDisplayFormat, SetVideoFormat?)
-    ITU BT601, ITU BT709 (HD), RGB studio levels (16-235)?
    suspend output / energie saver: stop and restart X11
+    suspend plugin didn't restore full-screen (is this wanted?)
    Option deinterlace off / deinterlace force!
-    Make output drivers better modular (under construction).
+    ColorSpace aren't configurable with the gui.
+    Replay of old vdr 1.6 recordings.
+    svdrp support for hot-keys.

 crash:
    AudioPlayHandlerThread -> pthread_cond_wait
@@ -35,18 +36,21 @@ video:
    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 line not configurable from setup menu.
+    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

 vdpau:
+    software decoder path not working

 libva:
    yaepghd (VaapiSetOutputPosition) support
    can associate only displayed part of osd
    grab image for va-api
-    still many:
+    remove stderr output of libva init
+    still many: (workaround export NO_MPEG_HW=1)
    [drm:i915_hangcheck_elapsed] *ERROR* Hangcheck timer elapsed... GPU hung
    [drm:i915_wait_request] *ERROR* i915_wait_request returns -11 ...

@@ -54,28 +58,36 @@ libva: branch vaapi-ext
    add support for vaapi-ext

 libva-intel-driver:
+    deinterlace only supported with vaapi-ext
    1080i does no v-sync (sometimes correct working with vaapi-ext)
    OSD has sometimes wrong size (workaround written)
-    software decoder needs UV swab
+    sometimes software decoder deinterlace isn't working and 1080i channels
+    show artefacts

 libva-vdpau-driver:
    G210/GT520 OSD update too slow (needs hardware problem workaround)
    hangup on exit (VaapiDelDecoder -> VaapiCleanup
 	-> vaDestroyContext -> pthread_rwlock_wrlock)
-    with auto-crop OSD has wrong position
+    OSD still has some problems with auto-crop and 4:3 zoom.

 libva-xvba-driver:
-    with auto-crop OSD has wrong position

 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
+    software volume support (could be done with asound.conf)
+    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.

 audio/alsa:
    better downmix of >2 channels on 2 channel hardware
@@ -85,16 +97,18 @@ audio/alsa:

 audio/oss:
    alsa oss emulation mixer "pcm" not working
+    oss4 mixer channel not working
    ring buffer overflow with alsa oss emulation

 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 (OSD updates?)

 setup:
    Setup of decoder type.
@@ -105,8 +119,6 @@ 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.

 future features (not planed for 1.0 - 1.5)
@@ -118,5 +130,6 @@ future features (not planed for 1.0 - 1.5)
    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)
--- a/audio.c
+++ b/audio.c
@@ -107,12 +107,16 @@ typedef struct _audio_module_

    void (*Thread) (void);		///< module thread handler
    void (*Enqueue) (const void *, int);	///< enqueue samples for output
+    void (*VideoReady) (void);		///< video ready, start audio
    void (*FlushBuffers) (void);	///< flush sample buffers
    void (*Poller) (void);		///< output poller
    int (*FreeBytes) (void);		///< number of bytes free in buffer
+    int (*UsedBytes) (void);		///< number of bytes used in buffer
     uint64_t(*GetDelay) (void);	///< get current audio delay
    void (*SetVolume) (int);		///< set output volume
    int (*Setup) (int *, int *, int);	///< setup channels, samplerate
+    void (*Play) (void);		///< play
+    void (*Pause) (void);		///< pause
    void (*Init) (void);		///< initialize audio output module
    void (*Exit) (void);		///< cleanup audio output module
 } AudioModule;
@@ -123,6 +127,8 @@ static const AudioModule NoopModule;	///< forward definition of noop module
 //	Variables
 //----------------------------------------------------------------------------

+char AudioAlsaDriverBroken;		///< disable broken driver message
+
 static const char *AudioModuleName;	///< which audio module to use

    /// Selected audio module.
@@ -132,12 +138,13 @@ static const char *AudioAC3Device;	///< alsa/OSS AC3 device name
 static const char *AudioMixerDevice;	///< alsa/OSS mixer device name
 static const char *AudioMixerChannel;	///< alsa/OSS mixer channel name
 static volatile char AudioRunning;	///< thread running / stopped
-static int AudioPaused;			///< audio paused
+static volatile char AudioPaused;	///< audio paused
+static volatile char AudioVideoIsReady;	///< video ready start early
 static unsigned AudioSampleRate;	///< audio sample rate in hz
 static unsigned AudioChannels;		///< number of audio channels
 static const int AudioBytesProSample = 2;	///< number of bytes per sample
 static int64_t AudioPTS;		///< audio pts clock
-static const int AudioBufferTime = 450;	///< audio buffer time in ms
+static int AudioBufferTime = 336;	///< audio buffer time in ms

 #ifdef USE_AUDIO_THREAD
 static pthread_t AudioThread;		///< audio play thread
@@ -147,6 +154,8 @@ static pthread_cond_t AudioStartCond;	///< condition variable
 static const int AudioThread;		///< dummy audio thread
 #endif

+extern int VideoAudioDelay;		///< import audio/video delay
+
 #ifdef USE_AUDIORING

 //----------------------------------------------------------------------------
@@ -283,13 +292,19 @@ static int AlsaAddToRingbuffer(const void *samples, int count)
 	// too many bytes are lost
 	// FIXME: should skip more, longer skip, but less often?
    }
-    // Update audio clock
-    AudioPTS +=
-	((int64_t) count * 90000) / (AudioSampleRate * AudioChannels *
-	AudioBytesProSample);

    if (!AudioRunning) {
-	if (AlsaStartThreshold < RingBufferUsedBytes(AlsaRingBuffer)) {
+	Debug(4, "audio/alsa: start %4zdms\n",
+	    (RingBufferUsedBytes(AlsaRingBuffer) * 1000)
+	    / (AudioSampleRate * AudioChannels * AudioBytesProSample));
+
+	// forced start
+	if (AlsaStartThreshold * 2 < RingBufferUsedBytes(AlsaRingBuffer)) {
+	    return 1;
+	}
+	// enough video + audio buffered
+	if (AudioVideoIsReady
+	    && AlsaStartThreshold < RingBufferUsedBytes(AlsaRingBuffer)) {
 	    // restart play-back
 	    return 1;
 	}
@@ -318,7 +333,8 @@ static int AlsaPlayRingbuffer(void)
 	    if (n == -EAGAIN) {
 		continue;
 	    }
-	    Error(_("audio/alsa: underrun error?\n"));
+	    Error(_("audio/alsa: avail underrun error? '%s'\n"),
+		snd_strerror(n));
 	    err = snd_pcm_recover(AlsaPCMHandle, n, 0);
 	    if (err >= 0) {
 		continue;
@@ -332,7 +348,18 @@ static int AlsaPlayRingbuffer(void)
 	    if (first) {
 		// happens with broken alsa drivers
 		if (AudioThread) {
-		    Error(_("audio/alsa: broken driver %d\n"), avail);
+		    if (!AudioAlsaDriverBroken) {
+			Error(_("audio/alsa: broken driver %d\n"), avail);
+			Error("audio/alsa: state %s\n",
+			    snd_pcm_state_name(snd_pcm_state(AlsaPCMHandle)));
+		    }
+		    if (snd_pcm_state(AlsaPCMHandle)
+			== SND_PCM_STATE_PREPARED) {
+			if ((err = snd_pcm_start(AlsaPCMHandle)) < 0) {
+			    Error(_("audio/alsa: snd_pcm_start(): %s\n"),
+				snd_strerror(err));
+			}
+		    }
 		    usleep(5 * 1000);
 		}
 	    }
@@ -343,6 +370,9 @@ static int AlsaPlayRingbuffer(void)
 	n = RingBufferGetReadPointer(AlsaRingBuffer, &p);
 	if (!n) {			// ring buffer empty
 	    if (first) {		// only error on first loop
+		Debug(4, "audio/alsa: empty buffers %d\n", avail);
+		// ring buffer empty
+		// AlsaLowWaterMark = 1;
 		return 1;
 	    }
 	    return 0;
@@ -372,7 +402,8 @@ static int AlsaPlayRingbuffer(void)
 		   goto again;
 		   }
 		 */
-		Error(_("audio/alsa: underrun error?\n"));
+		Error(_("audio/alsa: writei underrun error? '%s'\n"),
+		    snd_strerror(err));
 		err = snd_pcm_recover(AlsaPCMHandle, err, 0);
 		if (err >= 0) {
 		    goto again;
@@ -404,7 +435,7 @@ static void AlsaFlushBuffers(void)
 	RingBufferReadAdvance(AlsaRingBuffer,
 	    RingBufferUsedBytes(AlsaRingBuffer));
 	state = snd_pcm_state(AlsaPCMHandle);
-	Debug(3, "audio/alsa: state %d - %s\n", state,
+	Debug(3, "audio/alsa: flush state %d - %s\n", state,
 	    snd_pcm_state_name(state));
 	if (state != SND_PCM_STATE_OPEN) {
 	    if ((err = snd_pcm_drop(AlsaPCMHandle)) < 0) {
@@ -417,6 +448,7 @@ static void AlsaFlushBuffers(void)
 	}
    }
    AudioRunning = 0;
+    AudioVideoIsReady = 0;
    AudioPTS = INT64_C(0x8000000000000000);
 }

@@ -441,6 +473,14 @@ static int AlsaFreeBytes(void)
    return AlsaRingBuffer ? RingBufferFreeBytes(AlsaRingBuffer) : INT32_MAX;
 }

+/**
+**	Get used bytes in audio output.
+*/
+static int AlsaUsedBytes(void)
+{
+    return AlsaRingBuffer ? RingBufferUsedBytes(AlsaRingBuffer) : 0;
+}
+
 #if 0

 //----------------------------------------------------------------------------
@@ -561,11 +601,8 @@ static void AlsaEnqueue(const void *samples, int count)
 	    state = snd_pcm_state(AlsaPCMHandle);
 	    Debug(3, "audio/alsa: state %s\n", snd_pcm_state_name(state));
 	    Debug(3, "audio/alsa: unpaused\n");
-	    AudioPaused = 0;
 	}
    }
-    // Update audio clock
-    // AudioPTS += (size * 90000) / (AudioSampleRate * AudioChannels * AudioBytesProSample);
 }

 #endif
@@ -620,18 +657,22 @@ static void AlsaThread(void)
 	    AlsaFlushBuffer = 0;
 	    break;
 	}
+	if (AudioPaused) {
+	    break;
+	}
 	// wait for space in kernel buffers
-	if ((err = snd_pcm_wait(AlsaPCMHandle, 100)) < 0) {
-	    Error(_("audio/alsa: wait underrun error?\n"));
+	if ((err = snd_pcm_wait(AlsaPCMHandle, 24)) < 0) {
+	    Error(_("audio/alsa: wait underrun error? '%s'\n"),
+		snd_strerror(err));
 	    err = snd_pcm_recover(AlsaPCMHandle, err, 0);
 	    if (err >= 0) {
 		continue;
 	    }
 	    Error(_("audio/alsa: snd_pcm_wait(): %s\n"), snd_strerror(err));
-	    usleep(100 * 1000);
+	    usleep(24 * 1000);
 	    continue;
 	}
-	if (AlsaFlushBuffer) {
+	if (AlsaFlushBuffer || AudioPaused) {
 	    continue;
 	}
 	if ((err = AlsaPlayRingbuffer())) {	// empty / error
@@ -642,11 +683,12 @@ static void AlsaThread(void)
 	    }
 	    state = snd_pcm_state(AlsaPCMHandle);
 	    if (state != SND_PCM_STATE_RUNNING) {
-		Debug(3, "audio/alsa: stopping play\n");
+		Debug(3, "audio/alsa: stopping play '%s'\n",
+		    snd_pcm_state_name(state));
 		break;
 	    }
 	    pthread_yield();
-	    usleep(20 * 1000);		// let fill/empty the buffers
+	    usleep(24 * 1000);		// let fill/empty the buffers
 	}
    }
 }
@@ -675,6 +717,26 @@ static void AlsaThreadEnqueue(const void *samples, int count)
    }
 }

+/**
+**	Video is ready, start audio if possible,
+*/
+static void AlsaVideoReady(void)
+{
+    if (AudioSampleRate && AudioChannels) {
+	Debug(3, "audio/alsa: start %4zdms video start\n",
+	    (RingBufferUsedBytes(AlsaRingBuffer) * 1000)
+	    / (AudioSampleRate * AudioChannels * AudioBytesProSample));
+    }
+
+    if (!AudioRunning) {
+	// enough video + audio buffered
+	if (AlsaStartThreshold < RingBufferUsedBytes(AlsaRingBuffer)) {
+	    AudioRunning = 1;
+	    pthread_cond_signal(&AudioStartCond);
+	}
+    }
+}
+
 /**
 **	Flush alsa buffers with thread.
 */
@@ -708,12 +770,12 @@ static snd_pcm_t *AlsaOpenPCM(int use_ac3)

    // &&|| hell
    if (!(use_ac3 && ((device = AudioAC3Device)
-		|| (device = getenv("ALSA_AC3_DEVICE"))
-		|| (device = getenv("ALSA_PASSTHROUGH_DEVICE"))))
+		|| (device = getenv("ALSA_AC3_DEVICE"))))
 	&& !(device = AudioPCMDevice) && !(device = getenv("ALSA_DEVICE"))) {
 	device = "default";
    }
-    Debug(3, "audio/alsa: &&|| hell '%s'\n", device);
+    Info(_("audio/alsa: using %sdevice '%s'\n"), use_ac3 ? "ac3 " : "",
+	device);

    // open none blocking; if device is already used, we don't want wait
    if ((err =
@@ -740,7 +802,8 @@ static void AlsaInitPCM(void)
    snd_pcm_t *handle;
    snd_pcm_hw_params_t *hw_params;
    int err;
-    snd_pcm_uframes_t buffer_size;
+
+    //snd_pcm_uframes_t buffer_size;

    if (!(handle = AlsaOpenPCM(0))) {
 	return;
@@ -755,8 +818,9 @@ static void AlsaInitPCM(void)
    }
    AlsaCanPause = snd_pcm_hw_params_can_pause(hw_params);
    Info(_("audio/alsa: supports pause: %s\n"), AlsaCanPause ? "yes" : "no");
-    snd_pcm_hw_params_get_buffer_size_max(hw_params, &buffer_size);
-    Info(_("audio/alsa: max buffer size %lu\n"), buffer_size);
+    // needs audio setup
+    //snd_pcm_hw_params_get_buffer_size_max(hw_params, &buffer_size);
+    //Info(_("audio/alsa: max buffer size %lu\n"), buffer_size);

    AlsaPCMHandle = handle;
 }
@@ -816,7 +880,7 @@ static void AlsaInitMixer(void)
 	    const char *name;

 	    name = snd_mixer_selem_get_name(alsa_mixer_elem);
-	    if (strcasecmp(name, alsa_mixer_elem_name) == 0) {
+	    if (!strcasecmp(name, alsa_mixer_elem_name)) {
 		snd_mixer_selem_get_playback_volume_range(alsa_mixer_elem,
 		    &alsa_mixer_elem_min, &alsa_mixer_elem_max);
 		AlsaRatio =
@@ -856,6 +920,9 @@ static uint64_t AlsaGetDelay(void)
    if (!AlsaPCMHandle || !AudioSampleRate) {
 	return 0UL;
    }
+    if (!AudioRunning) {		// audio not running
+	return 0UL;
+    }
    // FIXME: thread safe? __assert_fail_base in snd_pcm_delay

    // delay in frames in alsa + kernel buffers
@@ -900,6 +967,7 @@ static int AlsaSetup(int *freq, int *channels, int use_ac3)
    snd_pcm_uframes_t period_size;
    int err;
    int ret;
+    int delay;
    snd_pcm_t *handle;

    if (!AlsaPCMHandle) {		// alsa not running yet
@@ -928,7 +996,7 @@ static int AlsaSetup(int *freq, int *channels, int use_ac3)
 	    snd_pcm_set_params(AlsaPCMHandle, SND_PCM_FORMAT_S16,
 		AlsaUseMmap ? SND_PCM_ACCESS_MMAP_INTERLEAVED :
 		SND_PCM_ACCESS_RW_INTERLEAVED, *channels, *freq, 1,
-		125 * 1000))) {
+		96 * 1000))) {
 	Error(_("audio/alsa: set params error: %s\n"), snd_strerror(err));

 	/*
@@ -1078,18 +1146,26 @@ static int AlsaSetup(int *freq, int *channels, int use_ac3)
    // update buffer

    snd_pcm_get_params(AlsaPCMHandle, &buffer_size, &period_size);
-    Info(_("audio/alsa: buffer size %lu, period size %lu\n"), buffer_size,
-	period_size);
+    Info(_("audio/alsa: buffer size %lu %zdms, period size %lu %zdms\n"),
+	buffer_size, snd_pcm_frames_to_bytes(AlsaPCMHandle,
+	    buffer_size) * 1000 / (AudioSampleRate * AudioChannels *
+	    AudioBytesProSample), period_size,
+	snd_pcm_frames_to_bytes(AlsaPCMHandle,
+	    period_size) * 1000 / (AudioSampleRate * AudioChannels *
+	    AudioBytesProSample));
    Debug(3, "audio/alsa: state %s\n",
 	snd_pcm_state_name(snd_pcm_state(AlsaPCMHandle)));

    AlsaStartThreshold = snd_pcm_frames_to_bytes(AlsaPCMHandle, period_size);
    // buffer time/delay in ms
+    delay = AudioBufferTime;
+    if (VideoAudioDelay > 0) {
+	delay += VideoAudioDelay / 90;
+    }
    if (AlsaStartThreshold <
-	(*freq * *channels * AudioBytesProSample * AudioBufferTime) / 1000U) {
+	(*freq * *channels * AudioBytesProSample * delay) / 1000U) {
 	AlsaStartThreshold =
-	    (*freq * *channels * AudioBytesProSample * AudioBufferTime) /
-	    1000U;
+	    (*freq * *channels * AudioBytesProSample * delay) / 1000U;
    }
    // no bigger, than the buffer
    if (AlsaStartThreshold > RingBufferFreeBytes(AlsaRingBuffer)) {
@@ -1101,6 +1177,47 @@ static int AlsaSetup(int *freq, int *channels, int use_ac3)
    return ret;
 }

+/**
+**	Play audio.
+*/
+void AlsaPlay(void)
+{
+    int err;
+
+    if (AlsaCanPause) {
+	if ((err = snd_pcm_pause(AlsaPCMHandle, 0))) {
+	    Error(_("audio/alsa: snd_pcm_pause(): %s\n"), snd_strerror(err));
+	}
+    } else {
+	if ((err = snd_pcm_prepare(AlsaPCMHandle)) < 0) {
+	    Error(_("audio/alsa: snd_pcm_prepare(): %s\n"), snd_strerror(err));
+	}
+    }
+#ifdef DEBUG
+    if (snd_pcm_state(AlsaPCMHandle) == SND_PCM_STATE_PAUSED) {
+	Error(_("audio/alsa: still paused\n"));
+    }
+#endif
+}
+
+/**
+**	Pause audio.
+*/
+void AlsaPause(void)
+{
+    int err;
+
+    if (AlsaCanPause) {
+	if ((err = snd_pcm_pause(AlsaPCMHandle, 1))) {
+	    Error(_("snd_pcm_pause(): %s\n"), snd_strerror(err));
+	}
+    } else {
+	if ((err = snd_pcm_drop(AlsaPCMHandle)) < 0) {
+	    Error(_("snd_pcm_drop(): %s\n"), snd_strerror(err));
+	}
+    }
+}
+
 /**
 **	Empty log callback
 */
@@ -1159,16 +1276,21 @@ static const AudioModule AlsaModule = {
 #ifdef USE_AUDIO_THREAD
    .Thread = AlsaThread,
    .Enqueue = AlsaThreadEnqueue,
+    .VideoReady = AlsaVideoReady,
    .FlushBuffers = AlsaThreadFlushBuffers,
 #else
    .Enqueue = AlsaEnqueue,
+    .VideoReady = AlsaVideoReady,
    .FlushBuffers = AlsaFlushBuffers,
 #endif
    .Poller = AlsaPoller,
    .FreeBytes = AlsaFreeBytes,
+    .UsedBytes = AlsaUsedBytes,
    .GetDelay = AlsaGetDelay,
    .SetVolume = AlsaSetVolume,
    .Setup = AlsaSetup,
+    .Play = AlsaPlay,
+    .Pause = AlsaPause,
    .Init = AlsaInit,
    .Exit = AlsaExit,
 };
@@ -1189,6 +1311,7 @@ static int OssPcmFildes = -1;		///< pcm file descriptor
 static int OssMixerFildes = -1;		///< mixer file descriptor
 static int OssMixerChannel;		///< mixer channel index
 static RingBuffer *OssRingBuffer;	///< audio ring buffer
+static int OssFragmentTime;		///< fragment time in ms
 static unsigned OssStartThreshold;	///< start play, if filled

 #ifdef USE_AUDIO_THREAD
@@ -1217,13 +1340,19 @@ static int OssAddToRingbuffer(const void *samples, int count)
 	// too many bytes are lost
 	// FIXME: should skip more, longer skip, but less often?
    }
-    // Update audio clock
-    AudioPTS +=
-	((int64_t) count * 90000) / (AudioSampleRate * AudioChannels *
-	AudioBytesProSample);

    if (!AudioRunning) {
-	if (OssStartThreshold < RingBufferUsedBytes(OssRingBuffer)) {
+	Debug(4, "audio/oss: start %4zdms\n",
+	    (RingBufferUsedBytes(OssRingBuffer) * 1000)
+	    / (AudioSampleRate * AudioChannels * AudioBytesProSample));
+
+	// forced start
+	if (OssStartThreshold * 2 < RingBufferUsedBytes(OssRingBuffer)) {
+	    return 1;
+	}
+	// enough video + audio buffered
+	if (AudioVideoIsReady
+	    && OssStartThreshold < RingBufferUsedBytes(OssRingBuffer)) {
 	    // restart play-back
 	    return 1;
 	}
@@ -1272,7 +1401,7 @@ static int OssPlayRingbuffer(void)
 		Error(_("audio/oss: write error: %s\n"), strerror(errno));
 		return 1;
 	    }
-	    Error(_("audio/oss: error not all bytes written\n"));
+	    Warning(_("audio/oss: error not all bytes written\n"));
 	}
 	// advance how many could written
 	RingBufferReadAdvance(OssRingBuffer, n);
@@ -1297,6 +1426,7 @@ static void OssFlushBuffers(void)
 	}
    }
    AudioRunning = 0;
+    AudioVideoIsReady = 0;
    AudioPTS = INT64_C(0x8000000000000000);
 }

@@ -1355,6 +1485,14 @@ static int OssFreeBytes(void)
    return OssRingBuffer ? RingBufferFreeBytes(OssRingBuffer) : INT32_MAX;
 }

+/**
+**	Get used bytes in audio output.
+*/
+static int OssUsedBytes(void)
+{
+    return OssRingBuffer ? RingBufferUsedBytes(OssRingBuffer) : 0;
+}
+
 #ifdef USE_AUDIO_THREAD

 //----------------------------------------------------------------------------
@@ -1378,18 +1516,21 @@ static void OssThread(void)
 	    OssFlushBuffer = 0;
 	    break;
 	}
+	if (AudioPaused) {
+	    break;
+	}

 	fds[0].fd = OssPcmFildes;
 	fds[0].events = POLLOUT | POLLERR;
 	// wait for space in kernel buffers
-	err = poll(fds, 1, 100);
+	err = poll(fds, 1, OssFragmentTime);
 	if (err < 0) {
 	    Error(_("audio/oss: error poll %s\n"), strerror(errno));
-	    usleep(100 * 1000);
+	    usleep(OssFragmentTime * 1000);
 	    continue;
 	}

-	if (OssFlushBuffer) {
+	if (OssFlushBuffer || AudioPaused) {
 	    continue;
 	}

@@ -1398,7 +1539,7 @@ static void OssThread(void)
 		break;
 	    }
 	    pthread_yield();
-	    usleep(20 * 1000);		// let fill/empty the buffers
+	    usleep(OssFragmentTime * 1000);	// let fill/empty the buffers
 	}
    }
 }
@@ -1422,6 +1563,26 @@ static void OssThreadEnqueue(const void *samples, int count)
    }
 }

+/**
+**	Video is ready, start audio if possible,
+*/
+static void OssVideoReady(void)
+{
+    if (AudioSampleRate && AudioChannels) {
+	Debug(3, "audio/oss: start %4zdms video start\n",
+	    (RingBufferUsedBytes(OssRingBuffer) * 1000)
+	    / (AudioSampleRate * AudioChannels * AudioBytesProSample));
+    }
+
+    if (!AudioRunning) {
+	// enough video + audio buffered
+	if (OssStartThreshold < RingBufferUsedBytes(OssRingBuffer)) {
+	    AudioRunning = 1;
+	    pthread_cond_signal(&AudioStartCond);
+	}
+    }
+}
+
 /**
 **	Flush OSS buffers with thread.
 */
@@ -1458,7 +1619,7 @@ static int OssOpenPCM(int use_ac3)
 	&& !(device = AudioPCMDevice) && !(device = getenv("OSS_AUDIODEV"))) {
 	device = "/dev/dsp";
    }
-    Debug(3, "audio/oss: &&|| hell '%s'\n", device);
+    Info(_("audio/oss: using %sdevice '%s'\n"), use_ac3 ? "ac3" : "", device);

    if ((fildes = open(device, O_WRONLY)) < 0) {
 	Error(_("audio/oss: can't open dsp device '%s': %s\n"), device,
@@ -1578,8 +1739,7 @@ static uint64_t OssGetDelay(void)
    if (OssPcmFildes == -1) {		// setup failure
 	return 0UL;
    }
-
-    if (!AudioRunning) {
+    if (!AudioRunning) {		// audio not running
 	return 0UL;
    }
    // delay in bytes in kernel buffers
@@ -1589,18 +1749,14 @@ static uint64_t OssGetDelay(void)
 	    strerror(errno));
 	return 0UL;
    }
-    if (delay == -1) {
-	delay = 0UL;
+    if (delay < 0) {
+	delay = 0;
    }

-    pts = ((uint64_t) delay * 90 * 1000)
+    pts = ((uint64_t) (delay + RingBufferUsedBytes(OssRingBuffer)) * 90 * 1000)
 	/ (AudioSampleRate * AudioChannels * AudioBytesProSample);
-    pts += ((uint64_t) RingBufferUsedBytes(OssRingBuffer) * 90 * 1000)
-	/ (AudioSampleRate * AudioChannels * AudioBytesProSample);
-    if (pts > 600 * 90) {
-	Debug(4, "audio/oss: hw+sw delay %zd %" PRId64 " ms\n",
-	    RingBufferUsedBytes(OssRingBuffer), pts / 90);
-    }
+    Debug(4, "audio/oss: hw+sw delay %zd %" PRId64 " ms\n",
+	RingBufferUsedBytes(OssRingBuffer), pts / 90);

    return pts;
 }
@@ -1622,6 +1778,8 @@ static int OssSetup(int *freq, int *channels, int use_ac3)
 {
    int ret;
    int tmp;
+    int delay;
+    audio_buf_info bi;

    if (OssPcmFildes == -1) {		// OSS not ready
 	return -1;
@@ -1685,48 +1843,72 @@ static int OssSetup(int *freq, int *channels, int use_ac3)

    // FIXME: setup buffers

-    if (1) {
-	audio_buf_info bi;
+#ifdef SNDCTL_DSP_POLICY
+    tmp = 3;
+    if (ioctl(OssPcmFildes, SNDCTL_DSP_POLICY, &tmp) == -1) {
+	Error(_("audio/oss: ioctl(SNDCTL_DSP_POLICY): %s\n"), strerror(errno));
+    } else {
+	Info("audio/oss: set policy to %d\n", tmp);
+    }
+#endif

-	if (ioctl(OssPcmFildes, SNDCTL_DSP_GETOSPACE, &bi) == -1) {
-	    Error(_("audio/oss: ioctl(SNDCTL_DSP_GETOSPACE): %s\n"),
-		strerror(errno));
-	} else {
-	    Debug(3, "audio/oss: %d bytes buffered\n", bi.bytes);
-	}
-
-	tmp = -1;
-	if (ioctl(OssPcmFildes, SNDCTL_DSP_GETODELAY, &tmp) == -1) {
-	    Error(_("audio/oss: ioctl(SNDCTL_DSP_GETODELAY): %s\n"),
-		strerror(errno));
-	    // FIXME: stop player, set setup failed flag
-	    return -1;
-	}
-	if (tmp == -1) {
-	    tmp = 0;
-	}
-	// start when enough bytes for initial write
-	OssStartThreshold = bi.bytes + tmp;
-	// buffer time/delay in ms
-	if (OssStartThreshold <
-	    (*freq * *channels * AudioBytesProSample * AudioBufferTime) /
-	    1000U) {
-	    OssStartThreshold =
-		(*freq * *channels * AudioBytesProSample * AudioBufferTime) /
-		1000U;
-	}
-	// no bigger, than the buffer
-	if (OssStartThreshold > RingBufferFreeBytes(OssRingBuffer)) {
-	    OssStartThreshold = RingBufferFreeBytes(OssRingBuffer);
-	}
-
-	Info(_("audio/oss: delay %u ms\n"), (OssStartThreshold * 1000)
-	    / (AudioSampleRate * AudioChannels * AudioBytesProSample));
+    if (ioctl(OssPcmFildes, SNDCTL_DSP_GETOSPACE, &bi) == -1) {
+	Error(_("audio/oss: ioctl(SNDCTL_DSP_GETOSPACE): %s\n"),
+	    strerror(errno));
+	bi.fragsize = 4096;
+	bi.fragstotal = 16;
+    } else {
+	Debug(3, "audio/oss: %d bytes buffered\n", bi.bytes);
    }

+    OssFragmentTime = (bi.fragsize * 1000)
+	/ (AudioSampleRate * AudioChannels * AudioBytesProSample);
+
+    Info(_("audio/oss: buffer size %d %dms, fragment size %d %dms\n"),
+	bi.fragsize * bi.fragstotal, (bi.fragsize * bi.fragstotal * 1000)
+	/ (AudioSampleRate * AudioChannels * AudioBytesProSample), bi.fragsize,
+	OssFragmentTime);
+
+    // start when enough bytes for initial write
+    OssStartThreshold = (bi.fragsize - 1) * bi.fragstotal;
+
+    // buffer time/delay in ms
+    delay = AudioBufferTime + 300;
+    if (VideoAudioDelay > 0) {
+	delay += VideoAudioDelay / 90;
+    }
+    if (OssStartThreshold <
+	(AudioSampleRate * AudioChannels * AudioBytesProSample * delay) /
+	1000U) {
+	OssStartThreshold =
+	    (AudioSampleRate * AudioChannels * AudioBytesProSample * delay) /
+	    1000U;
+    }
+    // no bigger, than the buffer
+    if (OssStartThreshold > RingBufferFreeBytes(OssRingBuffer)) {
+	OssStartThreshold = RingBufferFreeBytes(OssRingBuffer);
+    }
+
+    Info(_("audio/oss: delay %u ms\n"), (OssStartThreshold * 1000)
+	/ (AudioSampleRate * AudioChannels * AudioBytesProSample));
+
    return ret;
 }

+/**
+**	Play audio.
+*/
+void OssPlay(void)
+{
+}
+
+/**
+**	Pause audio.
+*/
+void OssPause(void)
+{
+}
+
 /**
 **	Initialize OSS audio output module.
 */
@@ -1762,16 +1944,21 @@ static const AudioModule OssModule = {
 #ifdef USE_AUDIO_THREAD
    .Thread = OssThread,
    .Enqueue = OssThreadEnqueue,
+    .VideoReady = OssVideoReady,
    .FlushBuffers = OssThreadFlushBuffers,
 #else
    .Enqueue = OssEnqueue,
+    .VideoReady = OssVideoReady,
    .FlushBuffers = OssFlushBuffers,
 #endif
    .Poller = OssPoller,
    .FreeBytes = OssFreeBytes,
+    .UsedBytes = OssUsedBytes,
    .GetDelay = OssGetDelay,
    .SetVolume = OssSetVolume,
    .Setup = OssSetup,
+    .Play = OssPlay,
+    .Pause = OssPause,
    .Init = OssInit,
    .Exit = OssExit,
 };
@@ -1802,6 +1989,14 @@ static int NoopFreeBytes(void)
    return INT32_MAX;			// no driver, much space
 }

+/**
+**	Get used bytes in audio output.
+*/
+static int NoopUsedBytes(void)
+{
+    return 0;				// no driver, nothing used
+}
+
 /**
 **	Get audio delay in time stamps.
 **
@@ -1849,12 +2044,16 @@ static void NoopVoid(void)
 static const AudioModule NoopModule = {
    .Name = "noop",
    .Enqueue = NoopEnqueue,
+    .VideoReady = NoopVoid,
    .FlushBuffers = NoopVoid,
    .Poller = NoopVoid,
    .FreeBytes = NoopFreeBytes,
+    .UsedBytes = NoopUsedBytes,
    .GetDelay = NoopGetDelay,
    .SetVolume = NoopSetVolume,
    .Setup = NoopSetup,
+    .Play = NoopVoid,
+    .Pause = NoopVoid,
    .Init = NoopVoid,
    .Exit = NoopVoid,
 };
@@ -1879,6 +2078,10 @@ static void *AudioPlayHandlerThread(void *dummy)
 	    pthread_cond_wait(&AudioStartCond, &AudioMutex);
 	    // cond_wait can return, without signal!
 	} while (!AudioRunning);
+
+	Debug(3, "audio: ----> %d ms\n", (AudioUsedBytes() * 1000)
+	    / (AudioSampleRate * AudioChannels * AudioBytesProSample));
+
 	pthread_mutex_unlock(&AudioMutex);

 #ifdef USE_AUDIORING
@@ -1979,7 +2182,38 @@ static const AudioModule *AudioModules[] = {
 */
 void AudioEnqueue(const void *samples, int count)
 {
+    if (0) {
+	static uint32_t last;
+	static uint32_t tick;
+	static uint32_t max = 101;
+	uint64_t delay;
+
+	delay = AudioGetDelay();
+	tick = GetMsTicks();
+	if ((last && tick - last > max) && AudioRunning) {
+
+	    //max = tick - last;
+	    Debug(3, "audio: packet delta %d %lu\n", tick - last, delay / 90);
+	}
+	last = tick;
+    }
    AudioUsedModule->Enqueue(samples, count);
+
+    // Update audio clock (stupid gcc developers thinks INT64_C is unsigned)
+    if (AudioPTS != (int64_t) INT64_C(0x8000000000000000)) {
+	AudioPTS +=
+	    ((int64_t) count * 90000) / (AudioSampleRate * AudioChannels *
+	    AudioBytesProSample);
+    }
+}
+
+/**
+**	Video is ready.
+*/
+void AudioVideoReady(void)
+{
+    AudioVideoIsReady = 1;
+    AudioUsedModule->VideoReady();
 }

 /**
@@ -2006,6 +2240,14 @@ int AudioFreeBytes(void)
    return AudioUsedModule->FreeBytes();
 }

+/**
+**	Get used bytes in audio output.
+*/
+int AudioUsedBytes(void)
+{
+    return AudioUsedModule->UsedBytes();
+}
+
 /**
 **	Get audio delay in time stamps.
 **
@@ -2040,7 +2282,8 @@ void AudioSetClock(int64_t pts)
 */
 int64_t AudioGetClock(void)
 {
-    if ((uint64_t) AudioPTS != INT64_C(0x8000000000000000)) {
+    // (cast) needed for the evil gcc
+    if (AudioPTS != (int64_t) INT64_C(0x8000000000000000)) {
 	int64_t delay;

 	if ((delay = AudioGetDelay())) {
@@ -2057,13 +2300,7 @@ int64_t AudioGetClock(void)
 */
 void AudioSetVolume(int volume)
 {
-#ifdef USE_ALSA
-    AlsaSetVolume(volume);
-#endif
-#ifdef USE_OSS
-    OssSetVolume(volume);
-#endif
-    (void)volume;
+    return AudioUsedModule->SetVolume(volume);
 }

 /**
@@ -2097,6 +2334,48 @@ int AudioSetup(int *freq, int *channels, int use_ac3)
    return AudioUsedModule->Setup(freq, channels, use_ac3);
 }

+/**
+**	Play audio.
+*/
+void AudioPlay(void)
+{
+    if (!AudioPaused) {
+	Debug(3, "audio: not paused, check the code\n");
+	return;
+    }
+    Debug(3, "audio: resumed\n");
+    AudioPaused = 0;
+    AudioEnqueue(NULL, 0);		// wakeup thread
+}
+
+/**
+**	Pause audio.
+*/
+void AudioPause(void)
+{
+    if (AudioPaused) {
+	Debug(3, "audio: already paused, check the code\n");
+	return;
+    }
+    Debug(3, "audio: paused\n");
+    AudioPaused = 1;
+}
+
+/**
+**	Set audio buffer time.
+**
+**	PES audio packets have a max distance of 300 ms.
+**	TS audio packet have a max distance of 100 ms.
+**	The period size of the audio buffer is 24 ms.
+*/
+void AudioSetBufferTime(int delay)
+{
+    if (!delay) {
+	delay = 336;
+    }
+    AudioBufferTime = delay;
+}
+
 /**
 **	Set pcm audio device.
 **
@@ -2137,6 +2416,18 @@ void AudioSetDeviceAC3(const char *device)
    AudioAC3Device = device;
 }

+/**
+**	Set pcm audio mixer channel.
+**
+**	@param channel	name of the mixer channel (fe. PCM or Master)
+**
+**	@note this is currently used to select alsa/OSS output module.
+*/
+void AudioSetChannel(const char *channel)
+{
+    AudioMixerChannel = channel;
+}
+
 /**
 **	Initialize audio output module.
 **
@@ -2188,8 +2479,7 @@ void AudioInit(void)
 	AudioInitThread();
    }
 #endif
-
-    AudioPaused = 1;
+    AudioPaused = 0;
 }

 /**
--- a/audio.h
+++ b/audio.h
@@ -31,20 +31,28 @@ 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 int AudioUsedBytes(void);	///< used bytes in audio output
 extern uint64_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 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
+
 /// @}
--- a/codec.c
+++ b/codec.c
@@ -30,13 +30,10 @@
 ///		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 noUSE_AUDIO_DRIFT_CORRECTION

 #include <stdio.h>
 #include <unistd.h>
@@ -355,7 +352,7 @@ 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 +377,7 @@ 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;
@@ -603,8 +600,7 @@ 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

@@ -613,6 +609,21 @@ 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_PASSTHROUGH
@@ -621,7 +632,11 @@ 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 +676,12 @@ 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 +689,12 @@ 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)
@@ -680,9 +703,19 @@ 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) {
-	pthread_mutex_unlock(&CodecLockMutex);
-	Fatal(_("codec: can't open audio codec\n"));
+    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);
@@ -693,14 +726,11 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name,
 	// 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"));
-    }
    audio_decoder->SampleRate = 0;
    audio_decoder->Channels = 0;
    audio_decoder->HwSampleRate = 0;
    audio_decoder->HwChannels = 0;
+    audio_decoder->LastDelay = 0;
 }

 /**
@@ -711,14 +741,25 @@ 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);
@@ -735,332 +776,518 @@ 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
-
 /**
-**	Decode an audio packet.
+**	Set audio downmix.
 **
-**	PTS must be handled self.
-**
-**	@param audio_decoder	audio decoder data
-**	@param avpkt		audio packet
+**	@param onoff	enable/disable downmix.
 */
-void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
+void CodecSetAudioDownmix(int onoff)
 {
-    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					// 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);
-	    if (l < 0) {		// no audio frame could be decompressed
-		Error(_("codec: error audio data at %d\n"), index);
-		break;
-	    }
-#ifdef notyetFF_API_OLD_DECODE_AUDIO
-	    // FIXME: ffmpeg git comeing
-	    int got_frame;
-
-	    avcodec_decode_audio4(audio_ctx, frame, &got_frame, dpkt);
-#else
-#endif
-	    // Update audio clock
-	    if ((uint64_t) dpkt->pts != AV_NOPTS_VALUE) {
-		AudioSetClock(dpkt->pts);
-	    }
-	    // FIXME: must first play remainings bytes, than change and play new.
-	    if (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;
-		    }
-		}
-	    }
-
-	    if (audio_decoder->HwSampleRate && audio_decoder->HwChannels) {
-		// need to resample audio
-		if (audio_decoder->ReSample) {
-		    int16_t outbuf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 +
-			FF_INPUT_BUFFER_PADDING_SIZE]
-			__attribute__ ((aligned(16)));
-		    int outlen;
-
-		    // FIXME: libav-0.7.2 crash here
-		    outlen =
-			audio_resample(audio_decoder->ReSample, outbuf, buf,
-			buf_sz);
-#ifdef DEBUG
-		    if (outlen != buf_sz) {
-			Debug(3, "codec/audio: possible fixed ffmpeg\n");
-		    }
-#endif
-		    if (outlen) {
-			// outlen seems to be wrong in ffmpeg-0.9
-			outlen /= audio_decoder->Channels *
-			    av_get_bytes_per_sample(audio_ctx->sample_fmt);
-			outlen *=
-			    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);
-		    }
-		} 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
-			// dpkt is the original data
-			buf_sz = 6144;
-			if (buf_sz < dpkt->size + 8) {
-			    Error(_
-				("codec/audio: decoded data smaller than encoded\n"));
-			    break;
-			}
-			// 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);
-		    }
-#if 0
-		    //
-		    //	old experimental code
-		    //
-		    if (1) {
-			// FIXME: need to detect dts
-			// copy original data for output
-			// FIXME: buf is sint
-			buf[0] = 0x72;
-			buf[1] = 0xF8;
-			buf[2] = 0x1F;
-			buf[3] = 0x4E;
-			buf[4] = 0x00;
-			switch (dpkt->size) {
-			    case 512:
-				buf[5] = 0x0B;
-				break;
-			    case 1024:
-				buf[5] = 0x0C;
-				break;
-			    case 2048:
-				buf[5] = 0x0D;
-				break;
-			    default:
-				Debug(3,
-				    "codec/audio: dts sample burst not supported\n");
-				buf[5] = 0x00;
-				break;
-			}
-			buf[6] = (dpkt->size * 8);
-			buf[7] = (dpkt->size * 8) >> 8;
-			//buf[8] = 0x0B;
-			//buf[9] = 0x77;
-			//printf("%x %x\n", dpkt->data[0],dpkt->data[1]);
-			// swab?
-			memcpy(buf + 8, dpkt->data, dpkt->size);
-			memset(buf + 8 + dpkt->size, 0,
-			    buf_sz - 8 - dpkt->size);
-		    } else if (1) {
-			// FIXME: need to detect mp2
-			// FIXME: mp2 passthrough
-			// see softhddev.c version/layer
-			// 0x04 mpeg1 layer1
-			// 0x05 mpeg1 layer23
-			// 0x06 mpeg2 ext
-			// 0x07 mpeg2.5 layer 1
-			// 0x08 mpeg2.5 layer 2
-			// 0x09 mpeg2.5 layer 3
-		    }
-		    // DTS HD?
-		    // True HD?
-#endif
-#endif
-		    AudioEnqueue(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
+    CodecDownmix = onoff;
 }

-#else
-
 /**
-**	Decode an audio packet.
+**	Reorder audio frame.
 **
-**	PTS must be handled self.
-**
-**	@param audio_decoder	audio decoder data
-**	@param avpkt		audio packet
+**	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
 */
-void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
+static void CodecReorderAudioFrame(int16_t * buf, int size, int channels)
 {
-    int16_t buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 +
-	FF_INPUT_BUFFER_PADDING_SIZE] __attribute__ ((aligned(16)));
-    AVCodecContext *audio_ctx;
-    int index;
+    int i;
+    int c;
+    int ls;
+    int rs;
+    int lfe;

-//#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);
+    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;
-	}
-#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;
+	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;
+    }
+}

-	avcodec_decode_audio4(audio_ctx, frame, &got_frame, dpkt);
-#else
-#endif
-	// FIXME: see above, old code removed
+/**
+**	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;

-	index += n;
+    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 %zd ms\n", delay / 90);
+	return;
+    }
+    // collect over some time
+    pts_diff = pts - audio_decoder->LastPTS;
+    if (pts_diff < 10 * 1000 * 90) {
+	return;
    }

-#if 1
-    // or av_free_packet, make no difference here
-    av_destruct_packet(spkt);
+    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:%5zdms D:%4zdms %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;
+    } else {
+
+	drift += audio_decoder->Drift;
+	audio_decoder->Drift = drift;
+	corr = (10 * audio_decoder->HwSampleRate * drift) / (90 * 1000);
+#if defined(USE_PASSTHROUGH) && !defined(USE_AC3_DRIFT_CORRECTION)
+	// SPDIF/HDMI passthrough
+	if (!CodecPassthroughAC3
+	    || audio_decoder->AudioCtx->codec_id != CODEC_ID_AC3)
+#endif
+	{
+	    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;
+
+	distance = (pts_diff * audio_decoder->HwSampleRate) / (90 * 1000);
+	av_resample_compensate(audio_decoder->AvResample,
+	    audio_decoder->DriftCorr / 10, distance);
+    }
+    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;
+
+    audio_ctx = audio_decoder->AudioCtx;
+
+    audio_decoder->PassthroughAC3 = CodecPassthroughAC3;
+
+    // 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_decoder->SampleRate = audio_ctx->sample_rate;
+    audio_decoder->HwSampleRate = audio_ctx->sample_rate;
+    audio_decoder->Channels = audio_ctx->channels;
+    // 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 (!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 samples
+**
+*/
+void CodecAudioEnqueue(AudioDecoder * audio_decoder, int16_t * data, int count)
+{
+#ifdef USE_AUDIO_DRIFT_CORRECTION
+    if (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.
+**
+**	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)));
+    int buf_sz;
+    int l;
+    AVCodecContext *audio_ctx;
+
+    audio_ctx = audio_decoder->AudioCtx;
+
+    buf_sz = sizeof(buf);
+    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\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, 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
+	|| audio_decoder->SampleRate != audio_ctx->sample_rate
+	|| audio_decoder->Channels != audio_ctx->channels) {
+	CodecAudioUpdateFormat(audio_decoder);
+    }
+
+    if (audio_decoder->HwSampleRate && audio_decoder->HwChannels) {
+	// need to resample audio
+	if (audio_decoder->ReSample) {
+	    int16_t outbuf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 +
+		FF_INPUT_BUFFER_PADDING_SIZE]
+		__attribute__ ((aligned(16)));
+	    int outlen;
+
+	    // FIXME: libav-0.7.2 crash here
+	    outlen =
+		audio_resample(audio_decoder->ReSample, outbuf, buf, buf_sz);
+#ifdef DEBUG
+	    if (outlen != buf_sz) {
+		Debug(3, "codec/audio: possible fixed ffmpeg\n");
+	    }
+#endif
+	    if (outlen) {
+		// outlen seems to be wrong in ffmpeg-0.9
+		outlen /= audio_decoder->Channels *
+		    av_get_bytes_per_sample(audio_ctx->sample_fmt);
+		outlen *=
+		    audio_decoder->HwChannels *
+		    av_get_bytes_per_sample(audio_ctx->sample_fmt);
+		Debug(4, "codec/audio: %d -> %d\n", buf_sz, outlen);
+		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 (1) {
+		    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);
+		return;
+	    }
+#if 0
+	    //
+	    //	old experimental code
+	    //
+	    if (1) {
+		// FIXME: need to detect dts
+		// copy original data for output
+		// FIXME: buf is sint
+		buf[0] = 0x72;
+		buf[1] = 0xF8;
+		buf[2] = 0x1F;
+		buf[3] = 0x4E;
+		buf[4] = 0x00;
+		switch (avpkt->size) {
+		    case 512:
+			buf[5] = 0x0B;
+			break;
+		    case 1024:
+			buf[5] = 0x0C;
+			break;
+		    case 2048:
+			buf[5] = 0x0D;
+			break;
+		    default:
+			Debug(3,
+			    "codec/audio: dts sample burst not supported\n");
+			buf[5] = 0x00;
+			break;
+		}
+		buf[6] = (avpkt->size * 8);
+		buf[7] = (avpkt->size * 8) >> 8;
+		//buf[8] = 0x0B;
+		//buf[9] = 0x77;
+		//printf("%x %x\n", avpkt->data[0],avpkt->data[1]);
+		// swab?
+		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
+		// see softhddev.c version/layer
+		// 0x04 mpeg1 layer1
+		// 0x05 mpeg1 layer23
+		// 0x06 mpeg2 ext
+		// 0x07 mpeg2.5 layer 1
+		// 0x08 mpeg2.5 layer 2
+		// 0x09 mpeg2.5 layer 3
+	    }
+	    // DTS HD?
+	    // True HD?
+#endif
+#endif
+	    CodecAudioEnqueue(audio_decoder, buf, buf_sz);
+	}
+    }
+}

 /**
 **	Flush the audio decoder.
@@ -1069,7 +1296,6 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
 */
 void CodecAudioFlushBuffers(AudioDecoder * decoder)
 {
-    // FIXME: reset audio parser
    avcodec_flush_buffers(decoder->AudioCtx);
 }

--- a/codec.h
+++ b/codec.h
@@ -67,6 +67,9 @@ extern void CodecAudioOpen(AudioDecoder *, const char *, int);
    /// Close audio codec.
 extern void CodecAudioClose(AudioDecoder *);

+    /// Decode an audio packet.
+extern void CodecAudioDecodeOld(AudioDecoder *, const AVPacket *);
+
    /// Decode an audio packet.
 extern void CodecAudioDecode(AudioDecoder *, const AVPacket *);

--- a/softhddev.c
+++ b/softhddev.c
--- a/softhddev.h
+++ b/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 +50,17 @@ 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 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
@@ -72,7 +76,7 @@ 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 main thread hook
--- a/softhddevice.cpp
+++ b/softhddevice.cpp
--- a/vdr-softhddevice-9999.ebuild
+++ b/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
--- a/video.c
+++ b/video.c
--- a/video.h
+++ b/video.h
@@ -30,6 +30,12 @@
    /// Video hardware decoder typedef
 typedef struct _video_hw_decoder_ VideoHwDecoder;

+//----------------------------------------------------------------------------
+//	Variables
+//----------------------------------------------------------------------------
+
+extern char VideoIgnoreRepeatPict;	///< disable repeat pict warning
+
 //----------------------------------------------------------------------------
 //	Prototypes
 //----------------------------------------------------------------------------
@@ -47,16 +53,17 @@ extern unsigned VideoGetSurface(VideoHwDecoder *);
 extern void VideoReleaseSurface(VideoHwDecoder *, unsigned);

 #ifdef LIBAVCODEC_VERSION
-    /// Render a ffmpeg frame.
-extern void VideoRenderFrame(VideoHwDecoder *, AVCodecContext *, AVFrame *);
-
-    /// Get ffmpeg vaapi context.
-extern struct vaapi_context *VideoGetVaapiContext(VideoHwDecoder *);
-
    /// Callback to negotiate the PixelFormat.
 extern enum PixelFormat Video_get_format(VideoHwDecoder *, AVCodecContext *,
    const enum PixelFormat *);

+    /// Render a ffmpeg frame.
+extern void VideoRenderFrame(VideoHwDecoder *, const AVCodecContext *,
+    const AVFrame *);
+
+    /// Get ffmpeg vaapi context.
+extern struct vaapi_context *VideoGetVaapiContext(VideoHwDecoder *);
+
 #ifdef AVCODEC_VDPAU_H
    /// Draw vdpau render state.
 extern void VideoDrawRenderState(VideoHwDecoder *,
@@ -73,12 +80,18 @@ extern void VideoDisplayWakeup(void);
    /// Set video geometry.
 extern int VideoSetGeometry(const char *);

+    /// Set 60Hz display mode.
+extern void VideoSet60HzMode(int);
+
    /// Set video output position.
 extern void VideoSetOutputPosition(int, int, int, int);

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

+    /// Set display format.
+extern void VideoSetDisplayFormat(int);
+
    /// Set video fullscreen mode.
 extern void VideoSetFullscreen(int);

@@ -88,6 +101,9 @@ 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[]);

@@ -100,6 +116,12 @@ extern void VideoSetSharpen(int[]);
    /// Set skip lines.
 extern void VideoSetSkipLines(int);

+    /// Set studio levels.
+extern void VideoSetStudioLevels(int);
+
+    /// Set background.
+extern void VideoSetBackground(uint32_t);
+
    /// Set audio delay.
 extern void VideoSetAudioDelay(int);

@@ -128,5 +150,6 @@ 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 VideoGetBuffers(void);	///< Get number of input buffers.

 /// @}
Author	SHA1	Message	Date
Johns	74a62e3649	Makes audio ts parser default. Suspend fixes.	2012-03-03 18:47:07 +01:00
Johns	7e1a42f7ed	Experimental ac3 audio drift correction support.	2012-03-03 16:45:59 +01:00
Johns	dda9011abc	Removes LPCM detection from TS parser.	2012-03-03 16:11:38 +01:00
Johns	de79e9211f	Disabled audio drift correction as default.	2012-03-02 18:17:51 +01:00
Johns	b0d9f41020	Rewrote video/audio start code.	2012-03-02 18:16:20 +01:00
Johns	4d1a516c80	Fix Bug: PES audio buffer not correct reset.	2012-03-02 16:06:45 +01:00
Johns	995f1286bd	Fix attach.	2012-03-02 00:38:52 +01:00
Johns	fd0ae12f24	Fix warning.	2012-03-02 00:28:53 +01:00
Johns	db258a0fbd	Detach/Attach on MakePrimaryDevice.	2012-03-02 00:22:08 +01:00
Johns	0df8e8a5fc	Handle initial suspend mode like normal suspend.	2012-03-02 00:05:03 +01:00
Johns	6a28064dce	Add support for attach/detach plugin.	2012-03-01 22:12:22 +01:00
Johns	b5e9077c74	Increase AudioBufferTime for OSS.	2012-03-01 17:50:57 +01:00
Johns	3b4ace14cf	Add ac3 to info message for pass-through.	2012-02-29 18:21:28 +01:00
Johns	5aa868c296	Don't change correction value during pass-through.	2012-02-29 17:40:58 +01:00
Johns	43b48224b5	Improved audio drift correction support.	2012-02-29 16:35:49 +01:00
Johns	144f22314f	Experimental audio drift correction support.	2012-02-27 23:13:53 +01:00
Johns	51eb720265	VideoSetFullscreen needs X11 connection.	2012-02-26 20:54:31 +01:00
Johns	e977007dd3	Removed old cruft.	2012-02-26 14:30:46 +01:00
Johns	769f00b4f6	Try to restart alsa after underrun.	2012-02-25 18:10:19 +01:00
Johns	aa426cd8b2	Add SVDRP HOTK command support and cleanup.	2012-02-25 13:02:15 +01:00
Johns	b2cab00599	Remove AVDictionary.	2012-02-24 18:16:24 +01:00
Johns	b54d62ef35	Video background color documentation.	2012-02-24 15:42:32 +01:00
Johns	9b68248a3e	Increased audio buffer time for PES packets.	2012-02-24 15:41:17 +01:00
Johns	762959fbb4	Only a single frame is supported.	2012-02-24 15:38:04 +01:00
Johns	07b426f2b5	Fix bug in new audio ts parser: hangup.	2012-02-24 15:22:26 +01:00
Johns	668a6ec277	Support configuration and set of video background.	2012-02-24 15:15:50 +01:00
Johns	82f61de117	Include GIT version, when build from git.	2012-02-23 23:33:00 +01:00
Johns	67e571f02b	Survive lost X11 display.	2012-02-23 17:57:21 +01:00
Johns	c17af0e958	Fix bug: 100% cpu use with plugins like mp3.	2012-02-23 15:32:43 +01:00
Johns	2561214c3e	Info time should be 1 minute and not ~1 second.	2012-02-22 18:37:50 +01:00
Johns	7382bd60ff	VA-API branch staging support.	2012-02-22 16:50:35 +01:00
Johns	73b93f1aba	Makes A/V sync info time configurable.	2012-02-22 16:32:40 +01:00
Johns	0243b1c8a7	Fix bug: No OSD until valid video stream shown.	2012-02-22 15:10:47 +01:00
Johns	6ce760ccd8	60Hz display mode configurable with setup.conf.	2012-02-22 15:06:05 +01:00
Johns	2f869884ba	Support downmix of AC-3 to stero.	2012-02-21 22:36:10 +01:00
Johns	5d8dea1b6b	New audio PES handling. New easier and more flexible audio PES packet parser, which includes own codec parser. Removed av_parser use. Reduced audio buffer time, faster channel switch. New audio transport stream parser (not enabled as default).	2012-02-21 20:55:28 +01:00
Johns	1f232db5b4	Nicer debug output when clock out of range.	2012-02-19 22:45:29 +01:00
Johns	c4ad13c53f	Fix bug: Grabbing JPG image fails while suspended.	2012-02-19 20:52:57 +01:00
Johns	98f73f2199	Add support for hot keys.	2012-02-19 19:22:03 +01:00
Johns	89ca44206c	Add support to use characters input in edit mode.	2012-02-17 16:37:38 +01:00
Johns	5c9b85b69b	Use SetVideoFormat to call SetVideoDisplayFormat.	2012-02-17 15:10:24 +01:00
Johns	09cfab3856	Add posibility to disable repeat pict warning.	2012-02-16 21:55:14 +01:00
Johns	30e903d90a	Wakeup audio thread after pause.	2012-02-16 18:41:46 +01:00
Johns	852d367225	Adds trick speed support.	2012-02-16 15:31:53 +01:00
Johns	a7f0cf6d6f	Version 0.4.8 released.	2012-02-16 09:59:40 +01:00
Johns	346953d209	Fix message, if no hq scaling is supported.	2012-02-16 09:58:13 +01:00
Johns	97af9c6de2	Fix bug: wrong start of video packet.	2012-02-15 22:19:50 +01:00
Johns	8dd95dab5e	VDPAU: Enable inverse telecine configuration.	2012-02-14 22:29:17 +01:00
Johns	6775173e4f	Fix bug: local id variable overwrites argument.	2012-02-14 21:51:13 +01:00
Johns	9170fcf485	Removed stupid gcc warnings.	2012-02-14 21:48:42 +01:00
Johns	919428cb80	Find AC3 (Dolby Digital) inside PES packet.	2012-02-14 21:18:24 +01:00
Johns	4331692ee5	Fix bug: audio increments invalid audio PTS.	2012-02-14 16:03:08 +01:00
Johns	5aa826bdb0	Fix bug: dvd plugin not working.	2012-02-14 15:12:48 +01:00
Johns	6736db082e	Fix bug: used frame-> instead of video_ctx->.	2012-02-14 14:46:49 +01:00
Johns	807b4df381	Release Version 0.4.7.	2012-02-13 23:21:11 +01:00
FireFly	56edfd4f54	Fix bug: unscaled jpeg includes PNG header.	2012-02-13 20:15:25 +01:00
Johns	0a1a258d2a	Update dependencies and install README.	2012-02-13 14:58:26 +01:00
Johns	09a0880d07	Update readme for new featurs.	2012-02-13 14:25:38 +01:00
Johns	a98a4adc7e	Studio levels could be configured in setup menu.	2012-02-13 14:13:24 +01:00
Johns	f872f54e2a	Window defaults to fullscreen without geometry.	2012-02-13 14:00:53 +01:00
m.Rcu	33c638d538	Jpeg screengrab use VDR RgbToJpeg function.	2012-02-12 20:30:50 +01:00
Johns	0a2a221fa9	Add play/pause audio support.	2012-02-12 20:14:43 +01:00
Johns	24a065e5de	Fix bug: audible glitch toggling AC-3 pass-through	2012-02-12 17:50:10 +01:00
Johns	6df970ca9e	Fix bug: mpeg stills not displayed.	2012-02-12 16:57:32 +01:00
Johns	616cd9e133	Forgot to reenable FindAudioSync.	2012-02-11 18:29:20 +01:00
Johns	a91533f6d1	Detect audio stream type only after stream switch.	2012-02-11 18:22:48 +01:00
Johns	baa4500a2c	Detect more h264 streams with leading zeros.	2012-02-11 17:18:44 +01:00
Johns	f28a737a9a	VDPAU: support for studio levels added.	2012-02-10 15:47:52 +01:00
Johns	19cec561ba	Fix bug #876 : Keypad not working.	2012-02-10 10:43:31 +01:00
Johns	d8f63adaad	Software deinterlacer (config/skip chroma deint): Add support for skip chroma deinterlace to software deinterlacer. Type of software deinterlacer now configurable from setup menu.	2012-02-09 21:22:42 +01:00
Johns	8c16466d31	Set mixer channel through command line option	2012-02-09 16:01:36 +01:00
Johns	ced54a5cf1	Fix bug: LFE moved to wrong position.	2012-02-09 00:46:02 +01:00
Johns	08246b5ac3	Guard suspend/resume against multiple calls.	2012-02-08 23:26:49 +01:00
Johns	c3a1de8c7b	jpeg_mem_dest only supported by jpeg 8.0.	2012-02-08 22:32:47 +01:00
Johns	918170d00b	Add support for AAC LATM audio streams.	2012-02-08 15:19:18 +01:00
Johns	bc50f37c4d	Spatial deinterlacer for VA-API.	2012-02-07 18:18:13 +01:00
Johns	09cf1f5c85	Fix bug: alsa+ffmpeg use different channel layout.	2012-02-07 17:08:59 +01:00
Johns	947f6b312e	Support more LPCM sample rates and channels.	2012-02-06 23:54:22 +01:00
Johns	99728258f1	Quick&dirty support for mpeg LPCM streams.	2012-02-06 22:58:42 +01:00
Johns	c972f8c4dd	Workaround for text2skin undrawn OSD areas.	2012-02-06 20:54:20 +01:00
Johns	7d38dff5bf	Detect dvb LPCM stream and ignore it.	2012-02-05 14:17:46 +01:00
Johns	8db8b68edd	Makes Workarounds command line configurable.	2012-02-04 16:38:10 +01:00
Johns	00cafd18ed	Release Version 0.4.6.	2012-02-02 23:32:51 +01:00
Johns	ab4e89132e	Auto-crop improvement and nicer a-v sync display.	2012-02-02 16:01:38 +01:00
Johns	3585f1df19	Increase audio buffer, if bigger audio delay used.	2012-02-02 16:01:08 +01:00
Johns	e258c35537	Makes SkipLines configure in setup menu.	2012-02-02 16:00:26 +01:00
Johns	91dbe46786	Add A-V info output and compile time option.	2012-02-01 23:12:45 +01:00
Johns	a7389111ff	Fix bug: VA-API intel software decoder broken.	2012-02-01 18:36:24 +01:00
Johns	27e9a88e2f	Video updates and bug fix. Check if surface is ready in VaapiGetSurface. Set PTS/DTS only in the first split video packet. Add support for 4:3 output modes. Quicker auto-crop after channel switch. Add auto-crop support for Intel VA-API backend. Fix bug: Auto-Crop logo skip didn't use displayed width.	2012-02-01 16:50:48 +01:00
Johns	33e9c71aea	Removed debug printf.	2012-01-31 20:48:47 +01:00
Johns	bd84e3f3b9	Workaround for mpeg2 FFMpeg+VA-API+Intel GPU hung.	2012-01-31 20:45:09 +01:00
Johns	364cc04736	Fix bug: Only black picture with VA-API hw decoder.	2012-01-30 23:09:53 +01:00
HelAu	ec4a899bb8	Add support to start the plugin in suspended mode.	2012-01-30 17:03:15 +01:00
Johns	dab31e2367	Finished rewrite of video code, to support modules.	2012-01-30 15:58:21 +01:00
Johns	e613ff1f7e	Add aspect change support to software decoder path	2012-01-29 23:57:22 +01:00
Johns	1886b745e5	Repair software decoder with vaapi vdpau backend.	2012-01-29 19:28:46 +01:00
Johns	422c378a5e	Add workaround for Intel VA-API MPEG GPU hung.	2012-01-29 11:28:10 +01:00