Release Version 0.5.0.

Use config value to change audio/video delay.
Change audio/video delay with hot-key.
2023-10-10 17:16:51 +00:00 · 2012-04-07 20:21:55 +02:00 · 2012-04-07 13:43:51 +02:00 · 2012-04-07 00:15:57 +02:00 · 2012-04-06 14:38:51 +02:00 · 2012-04-05 22:51:20 +02:00
14 changed files with 4260 additions and 1768 deletions
--- a/65
+++ b/65
@@ -1,3 +1,68 @@
+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

--- a/14
+++ b/14
@@ -19,8 +19,12 @@ 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 += -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")
@@ -33,7 +37,7 @@ 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
+CXXFLAGS ?= -g -O2 -W -Wall -Wextra -Werror=overloaded-virtual

 ### The directory environment:

@@ -66,7 +70,7 @@ 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 +86,7 @@ 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` \
--- a/README.txt
+++ b/README.txt
@@ -20,23 +20,24 @@ $Id$

 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 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 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 pass-through
+    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 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: software volume, software channel resample
    o planned: atmo light support

 To compile you must have the 'requires' installed.
@@ -137,14 +138,25 @@ 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.AutoCrop.Interval = 0
 	0 disables auto-crop
 	n each 'n' frames auto-crop is checked.
@@ -157,8 +169,13 @@ 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.
@@ -171,6 +188,14 @@ 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
+
 	VideoDisplayFormat = ?
 	0 pan and scan
 	1 letter box
@@ -209,8 +234,20 @@ 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:
 --------
@@ -248,6 +285,11 @@ 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
--- a/35
+++ b/35
@@ -19,26 +19,24 @@ GNU Affero General Public License for more details.
 $Id: $

 missing:
-    software deinterlace (yadif, ...)
-    software decoder with software deinterlace
+    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!
    ColorSpace aren't configurable with the gui.
-    Inverse telecine isn't configurable with the gui.
-
-crash:
-    AudioPlayHandlerThread -> pthread_cond_wait
+    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
-    hard channel switch
-    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.

 vdpau:
    software decoder path not working
@@ -52,8 +50,8 @@ 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
@@ -71,18 +69,21 @@ 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 (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.
+    multiple open of audio device, reduce them.

 audio/alsa:
    better downmix of >2 channels on 2 channel hardware
@@ -99,8 +100,11 @@ HDMI/SPDIF Passthrough:
    only AC-3 written

 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.
@@ -112,6 +116,7 @@ setup:

 unsorted:
    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)

--- a/audio.c
+++ b/audio.c
@@ -105,18 +105,20 @@ typedef struct _audio_module_
 {
    const char *Name;			///< audio output module name

-    void (*Thread) (void);		///< module thread handler
-    void (*Enqueue) (const void *, int);	///< enqueue samples for output
-    void (*FlushBuffers) (void);	///< flush sample buffers
-    void (*Poller) (void);		///< output poller
-    int (*FreeBytes) (void);		///< number of bytes free 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
+    void (*const Thread) (void);	///< module thread handler
+    void (*const Enqueue) (const void *, int);	///< enqueue samples for output
+    void (*const VideoReady) (void);	///< video ready, start audio
+    void (*const FlushBuffers) (void);	///< flush sample buffers
+    void (*const Poller) (void);	///< output poller
+    int (*const FreeBytes) (void);	///< number of bytes free in buffer
+    int (*const UsedBytes) (void);	///< number of bytes used in buffer
+     int64_t(*const GetDelay) (void);	///< get current audio delay
+    void (*const SetVolume) (int);	///< set output volume
+    int (*const Setup) (int *, int *, int);	///< setup channels, samplerate
+    void (*const Play) (void);		///< play
+    void (*const Pause) (void);		///< pause
+    void (*const Init) (void);		///< initialize audio output module
+    void (*const Exit) (void);		///< cleanup audio output module
 } AudioModule;

 static const AudioModule NoopModule;	///< forward definition of noop module
@@ -137,11 +139,12 @@ 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 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 = 350;	///< 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
@@ -151,7 +154,7 @@ static pthread_cond_t AudioStartCond;	///< condition variable
 static const int AudioThread;		///< dummy audio thread
 #endif

-extern int VideoAudioDelay;		/// import audio/video delay
+extern int VideoAudioDelay;		///< import audio/video delay

 #ifdef USE_AUDIORING

@@ -220,7 +223,7 @@ static void AudioRingInit(void)

    for (i = 0; i < AUDIO_RING_MAX; ++i) {
 	// FIXME:
-	//AlsaRingBuffer = RingBufferNew(48000 * 8 * 2);	// ~1s 8ch 16bit
+	//AlsaRingBuffer = RingBufferNew(2 * 48000 * 8 * 2);	// ~2s 8ch 16bit
    }
    // one slot always reservered
    AudioRingWrite = 1;
@@ -289,15 +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 (stupid gcc developers thinks INT64_C is unsigned)
-    if (AudioPTS != (int64_t) INT64_C(0x8000000000000000)) {
-	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;
 	}
@@ -326,7 +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;
@@ -342,6 +350,15 @@ static int AlsaPlayRingbuffer(void)
 		if (AudioThread) {
 		    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);
 		}
@@ -353,6 +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;
@@ -382,7 +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;
@@ -411,11 +432,26 @@ static void AlsaFlushBuffers(void)
    snd_pcm_state_t state;

    if (AlsaRingBuffer && AlsaPCMHandle) {
+#ifdef DEBUG
+	const void *r;
+	void *w;
+#endif
+
 	RingBufferReadAdvance(AlsaRingBuffer,
 	    RingBufferUsedBytes(AlsaRingBuffer));
+#ifdef DEBUG
+	RingBufferGetWritePointer(AlsaRingBuffer, &w);
+	RingBufferGetReadPointer(AlsaRingBuffer, &r);
+	if (r != w) {
+	    Fatal(_("audio/alsa: ringbuffer out of sync %zd-%zd\n"),
+		RingBufferGetWritePointer(AlsaRingBuffer, &w),
+		RingBufferGetReadPointer(AlsaRingBuffer, &r));
+	    abort();
+	}
+#endif
+
 	state = snd_pcm_state(AlsaPCMHandle);
-	Debug(3, "audio/alsa: state %d - %s\n", state,
-	    snd_pcm_state_name(state));
+	Debug(3, "audio/alsa: flush state %s\n", snd_pcm_state_name(state));
 	if (state != SND_PCM_STATE_OPEN) {
 	    if ((err = snd_pcm_drop(AlsaPCMHandle)) < 0) {
 		Error(_("audio: snd_pcm_drop(): %s\n"), snd_strerror(err));
@@ -427,6 +463,7 @@ static void AlsaFlushBuffers(void)
 	}
    }
    AudioRunning = 0;
+    AudioVideoIsReady = 0;
    AudioPTS = INT64_C(0x8000000000000000);
 }

@@ -451,6 +488,14 @@ 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

 //----------------------------------------------------------------------------
@@ -573,8 +618,6 @@ static void AlsaEnqueue(const void *samples, int count)
 	    Debug(3, "audio/alsa: unpaused\n");
 	}
    }
-    // Update audio clock
-    // AudioPTS += (size * 90000) / (AudioSampleRate * AudioChannels * AudioBytesProSample);
 }

 #endif
@@ -633,17 +676,19 @@ static void AlsaThread(void)
 	    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 || AudioPaused) {
+	// timeout or some commands
+	if (!err || AlsaFlushBuffer || AudioPaused) {
 	    continue;
 	}
 	if ((err = AlsaPlayRingbuffer())) {	// empty / error
@@ -654,11 +699,12 @@ 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
 	}
    }
 }
@@ -671,7 +717,7 @@ static void AlsaThread(void)
 */
 static void AlsaThreadEnqueue(const void *samples, int count)
 {
-    if (!AlsaRingBuffer || !AlsaPCMHandle || !AudioSampleRate) {
+    if (!AlsaRingBuffer || !AlsaPCMHandle) {
 	Debug(3, "audio/alsa: enqueue not ready\n");
 	return;
    }
@@ -687,6 +733,38 @@ static void AlsaThreadEnqueue(const void *samples, int count)
    }
 }

+/**
+**	Video is ready, start audio if possible,
+*/
+static void AlsaVideoReady(void)
+{
+    if (!AudioRunning) {
+	size_t used;
+
+	used = RingBufferUsedBytes(AlsaRingBuffer);
+	// enough video + audio buffered
+	if (AlsaStartThreshold < used) {
+	    // too much audio buffered, skip it
+	    if (AlsaStartThreshold * 2 < used) {
+		Debug(3, "audio/alsa: start %4zdms skip ready\n",
+		    ((used - AlsaStartThreshold * 2) * 1000)
+		    / (AudioSampleRate * AudioChannels * AudioBytesProSample));
+		RingBufferReadAdvance(AlsaRingBuffer,
+		    used - AlsaStartThreshold * 2);
+	    }
+	    AudioRunning = 1;
+	    pthread_cond_signal(&AudioStartCond);
+	}
+    }
+
+    if (AudioSampleRate && AudioChannels) {
+	Debug(3, "audio/alsa: start %4zdms video ready\n",
+	    (RingBufferUsedBytes(AlsaRingBuffer) * 1000)
+	    / (AudioSampleRate * AudioChannels * AudioBytesProSample));
+    }
+
+}
+
 /**
 **	Flush alsa buffers with thread.
 */
@@ -720,12 +798,12 @@ 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 =
@@ -752,7 +830,8 @@ 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;
@@ -767,8 +846,9 @@ 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;
 }
@@ -859,14 +939,17 @@ static void AlsaInitMixer(void)
 **
 **	@todo FIXME: handle the case no audio running
 */
-static uint64_t AlsaGetDelay(void)
+static int64_t AlsaGetDelay(void)
 {
    int err;
    snd_pcm_sframes_t delay;
-    uint64_t pts;
+    int64_t pts;

    if (!AlsaPCMHandle || !AudioSampleRate) {
-	return 0UL;
+	return 0L;
+    }
+    if (!AudioRunning) {		// audio not running
+	return 0L;
    }
    // FIXME: thread safe? __assert_fail_base in snd_pcm_delay

@@ -884,10 +967,10 @@ static uint64_t AlsaGetDelay(void)
 	delay = 0L;
    }

-    pts = ((uint64_t) delay * 90 * 1000) / AudioSampleRate;
-    pts += ((uint64_t) RingBufferUsedBytes(AlsaRingBuffer) * 90 * 1000)
+    pts = ((int64_t) delay * 90 * 1000) / AudioSampleRate;
+    pts += ((int64_t) RingBufferUsedBytes(AlsaRingBuffer) * 90 * 1000)
 	/ (AudioSampleRate * AudioChannels * AudioBytesProSample);
-    Debug(4, "audio/alsa: hw+sw delay %zd %" PRId64 " ms\n",
+    Debug(4, "audio/alsa: hw+sw delay %zd %" PRId64 "ms\n",
 	RingBufferUsedBytes(AlsaRingBuffer), pts / 90);

    return pts;
@@ -921,6 +1004,9 @@ static int AlsaSetup(int *freq, int *channels, int use_ac3)
 #if 1					// easy alsa hw setup way
    // flush any buffered data
    AudioFlushBuffers();
+    Debug(3, "audio: %dms flush\n", (AudioUsedBytes() * 1000)
+	/ (!AudioSampleRate + !AudioChannels +
+	    AudioSampleRate * AudioChannels * AudioBytesProSample));

    if (1) {				// close+open to fix hdmi no sound bugs
 	handle = AlsaPCMHandle;
@@ -941,7 +1027,7 @@ 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));

 	/*
@@ -1091,16 +1177,21 @@ 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);
+    Debug(3, "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 > -100) {
-	delay += 100 + VideoAudioDelay / 90;
+    if (VideoAudioDelay > 0) {
+	delay += VideoAudioDelay / 90;
    }
    if (AlsaStartThreshold <
 	(*freq * *channels * AudioBytesProSample * delay) / 1000U) {
@@ -1111,7 +1202,7 @@ static int AlsaSetup(int *freq, int *channels, int use_ac3)
    if (AlsaStartThreshold > RingBufferFreeBytes(AlsaRingBuffer)) {
 	AlsaStartThreshold = RingBufferFreeBytes(AlsaRingBuffer);
    }
-    Info(_("audio/alsa: delay %u ms\n"), (AlsaStartThreshold * 1000)
+    Info(_("audio/alsa: delay %ums\n"), (AlsaStartThreshold * 1000)
 	/ (AudioSampleRate * AudioChannels * AudioBytesProSample));

    return ret;
@@ -1181,7 +1272,7 @@ static void AlsaInit(void)
 #else
    (void)AlsaNoopCallback;
 #endif
-    AlsaRingBuffer = RingBufferNew(48000 * 8 * 2);	// ~1s 8ch 16bit
+    AlsaRingBuffer = RingBufferNew(2 * 48000 * 8 * 2);	// ~2s 8ch 16bit

    AlsaInitPCM();
    AlsaInitMixer();
@@ -1216,13 +1307,16 @@ 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,
@@ -1248,6 +1342,7 @@ 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
@@ -1276,15 +1371,19 @@ 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 (stupid gcc developers thinks INT64_C is unsigned)
-    if (AudioPTS != (int64_t) INT64_C(0x8000000000000000)) {
-	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;
 	}
@@ -1333,7 +1432,7 @@ 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);
@@ -1358,6 +1457,7 @@ static void OssFlushBuffers(void)
 	}
    }
    AudioRunning = 0;
+    AudioVideoIsReady = 0;
    AudioPTS = INT64_C(0x8000000000000000);
 }

@@ -1380,7 +1480,7 @@ static void OssEnqueue(const void *samples, int count)
    uint32_t tick;

    tick = GetMsTicks();
-    Debug(4, "audio/oss: %4d %d ms\n", count, tick - last_tick);
+    Debug(4, "audio/oss: %4d %dms\n", count, tick - last_tick);
    last_tick = tick;
 #endif

@@ -1416,6 +1516,14 @@ 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

 //----------------------------------------------------------------------------
@@ -1446,10 +1554,10 @@ static void OssThread(void)
 	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;
 	}

@@ -1462,7 +1570,7 @@ static void OssThread(void)
 		break;
 	    }
 	    pthread_yield();
-	    usleep(20 * 1000);		// let fill/empty the buffers
+	    usleep(OssFragmentTime * 1000);	// let fill/empty the buffers
 	}
    }
 }
@@ -1475,7 +1583,7 @@ static void OssThread(void)
 */
 static void OssThreadEnqueue(const void *samples, int count)
 {
-    if (!OssRingBuffer || OssPcmFildes == -1 || !AudioSampleRate) {
+    if (!OssRingBuffer || OssPcmFildes == -1) {
 	Debug(3, "audio/oss: enqueue not ready\n");
 	return;
    }
@@ -1486,6 +1594,26 @@ 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.
 */
@@ -1522,7 +1650,7 @@ 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,
@@ -1634,17 +1762,16 @@ static void OssInitMixer(void)
 **
 **	@returns audio delay in time stamps.
 */
-static uint64_t OssGetDelay(void)
+static int64_t OssGetDelay(void)
 {
    int delay;
-    uint64_t pts;
+    int64_t pts;

    if (OssPcmFildes == -1) {		// setup failure
-	return 0UL;
+	return 0L;
    }
-
-    if (!AudioRunning) {
-	return 0UL;
+    if (!AudioRunning) {		// audio not running
+	return 0L;
    }
    // delay in bytes in kernel buffers
    delay = -1;
@@ -1653,18 +1780,14 @@ 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 = ((int64_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;
 }
@@ -1687,6 +1810,7 @@ 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;
@@ -1750,46 +1874,54 @@ static int OssSetup(int *freq, int *channels, int use_ac3)

    // FIXME: setup buffers

-    if (1) {
-	audio_buf_info bi;
-
-	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
-	delay = AudioBufferTime;
-	if (VideoAudioDelay > -100) {
-	    delay += 100 + VideoAudioDelay / 90;
-	}
-	if (OssStartThreshold <
-	    (*freq * *channels * AudioBytesProSample * delay) / 1000U) {
-	    OssStartThreshold =
-		(*freq * *channels * 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));
+#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));
+	bi.fragsize = 4096;
+	bi.fragstotal = 16;
+    } else {
+	Debug(3, "audio/oss: %d bytes buffered\n", bi.bytes);
+    }
+
+    OssFragmentTime = (bi.fragsize * 1000)
+	/ (AudioSampleRate * AudioChannels * AudioBytesProSample);
+
+    Debug(3, "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 %ums\n"), (OssStartThreshold * 1000)
+	/ (AudioSampleRate * AudioChannels * AudioBytesProSample));

    return ret;
 }
@@ -1813,7 +1945,7 @@ void OssPause(void)
 */
 static void OssInit(void)
 {
-    OssRingBuffer = RingBufferNew(48000 * 8 * 2);	// ~1s 8ch 16bit
+    OssRingBuffer = RingBufferNew(2 * 48000 * 8 * 2);	// ~2s 8ch 16bit

    OssInitPCM();
    OssInitMixer();
@@ -1843,13 +1975,16 @@ 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,
@@ -1885,14 +2020,22 @@ 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.
 **
 **	@returns audio delay in time stamps.
 */
-static uint64_t NoopGetDelay(void)
+static int64_t NoopGetDelay(void)
 {
-    return 0UL;
+    return 0L;
 }

 /**
@@ -1932,9 +2075,11 @@ 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,
@@ -1964,6 +2109,11 @@ static void *AudioPlayHandlerThread(void *dummy)
 	    pthread_cond_wait(&AudioStartCond, &AudioMutex);
 	    // cond_wait can return, without signal!
 	} while (!AudioRunning);
+
+	Debug(3, "audio: ----> %dms start\n", (AudioUsedBytes() * 1000)
+	    / (!AudioSampleRate + !AudioChannels +
+		AudioSampleRate * AudioChannels * AudioBytesProSample));
+
 	pthread_mutex_unlock(&AudioMutex);

 #ifdef USE_AUDIORING
@@ -1997,7 +2147,6 @@ static void *AudioPlayHandlerThread(void *dummy)
 	}
 #endif

-	Debug(3, "audio: play start\n");
 	AudioUsedModule->Thread();
    }

@@ -2064,7 +2213,41 @@ static const AudioModule *AudioModules[] = {
 */
 void AudioEnqueue(const void *samples, int count)
 {
+    if (!AudioSampleRate || !AudioChannels) {
+	return;				// not setup
+    }
+    if (0) {
+	static uint32_t last;
+	static uint32_t tick;
+	static uint32_t max = 101;
+	int64_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 * 90 * 1000) / (AudioSampleRate * AudioChannels *
+	    AudioBytesProSample);
+    }
+}
+
+/**
+**	Video is ready.
+*/
+void AudioVideoReady(void)
+{
+    AudioVideoIsReady = 1;
+    AudioUsedModule->VideoReady();
 }

 /**
@@ -2091,12 +2274,20 @@ int AudioFreeBytes(void)
    return AudioUsedModule->FreeBytes();
 }

+/**
+**	Get used bytes in audio output.
+*/
+int AudioUsedBytes(void)
+{
+    return AudioUsedModule->UsedBytes();
+}
+
 /**
 **	Get audio delay in time stamps.
 **
 **	@returns audio delay in time stamps.
 */
-uint64_t AudioGetDelay(void)
+int64_t AudioGetDelay(void)
 {
    return AudioUsedModule->GetDelay();
 }
@@ -2110,9 +2301,8 @@ void AudioSetClock(int64_t pts)
 {
 #ifdef DEBUG
    if (AudioPTS != pts) {
-	Debug(4, "audio: set clock to %#012" PRIx64 " %#012" PRIx64 " pts\n",
-	    AudioPTS, pts);
-
+	Debug(4, "audio: set clock %s -> %s pts\n", Timestamp2String(AudioPTS),
+	    Timestamp2String(pts));
    }
 #endif
    AudioPTS = pts;
@@ -2183,11 +2373,12 @@ int AudioSetup(int *freq, int *channels, int use_ac3)
 void AudioPlay(void)
 {
    if (!AudioPaused) {
-	Warning("audio: not paused, check the code\n");
+	Debug(3, "audio: not paused, check the code\n");
 	return;
    }
    Debug(3, "audio: resumed\n");
    AudioPaused = 0;
+    AudioEnqueue(NULL, 0);		// wakeup thread
 }

 /**
@@ -2196,13 +2387,28 @@ void AudioPlay(void)
 void AudioPause(void)
 {
    if (AudioPaused) {
-	Warning("audio: already paused, check the code\n");
+	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.
 **
--- a/audio.h
+++ b/audio.h
@@ -31,9 +31,8 @@ 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
@@ -42,6 +41,8 @@ extern int AudioSetup(int *, int *, int);	///< setup audio output
 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 pass-through device
 extern void AudioSetChannel(const char *);	///< set mixer channel
--- 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;
@@ -561,6 +558,11 @@ 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,
@@ -603,8 +605,6 @@ 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
@@ -614,6 +614,21 @@ 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
@@ -626,6 +641,7 @@ static char CodecPassthroughAC3;	///< pass ac3 through

 static const int CodecPassthroughAC3 = 0;
 #endif
+static char CodecDownmix;		///< enable ac-3 downmix

 /**
 **	Allocate a new audio decoder context.
@@ -665,10 +681,12 @@ 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;
@@ -676,6 +694,12 @@ 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)
@@ -684,9 +708,19 @@ 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);
@@ -694,17 +728,14 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *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;
 }

 /**
@@ -715,14 +746,25 @@ 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);
@@ -742,12 +784,26 @@ void CodecSetAudioPassthrough(int mask)
    (void)mask;
 }

+/**
+**	Set audio downmix.
+**
+**	@param onoff	enable/disable downmix.
+*/
+void CodecSetAudioDownmix(int onoff)
+{
+    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)
 {
@@ -798,7 +854,277 @@ static void CodecReorderAudioFrame(int16_t * buf, int size, int channels)
    }
 }

-#ifdef USE_AVPARSER
+/**
+**	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 %zd 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:%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;
+
+	// 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);
+    }
+    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 (!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 (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.
@@ -812,324 +1138,172 @@ 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);
-	    if (l == AVERROR(EAGAIN)) {
-		index += n;		// this is needed for aac latm
-		continue;
-	    }
-	    if (l < 0) {		// no audio frame could be decompressed
-		Error(_("codec: error audio data at %d\n"), index);
-		break;
-	    }
+    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;
+    // 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);
-	    }
-	    // 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) {
-		int err;
-		int isAC3;

-		audio_decoder->PassthroughAC3 = CodecPassthroughAC3;
-		// FIXME: use swr_convert from swresample (only in ffmpeg!)
-		// FIXME: tell ac3 decoder to use downmix
-		if (audio_decoder->ReSample) {
-		    audio_resample_close(audio_decoder->ReSample);
-		    audio_decoder->ReSample = NULL;
-		}
+    // 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);
+    }

-		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;
-		}
+    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;

-		// 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);
+	    // 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");
-		    }
+	    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);
-			CodecReorderAudioFrame(outbuf, outlen,
-			    audio_decoder->HwChannels);
-			AudioEnqueue(outbuf, outlen);
-		    }
-		} else {
+	    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
-			// 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);
+	    // 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;
 		    }
-#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
-		    CodecReorderAudioFrame(buf, buf_sz,
-			audio_decoder->HwChannels);
-		    AudioEnqueue(buf, buf_sz);
+		    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 (dpkt->size > l) {
-		Error(_("codec: error more than one frame data\n"));
+#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
 	    }
-	}
-
-	index += n;
-    }
-
-#if 1
-    // or av_free_packet, make no difference here
-    av_destruct_packet(spkt);
+	    // DTS HD?
+	    // True HD?
 #endif
+#endif
+	    CodecAudioEnqueue(audio_decoder, buf, buf_sz);
+	}
+    }
 }

-#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.
 **
@@ -1137,7 +1311,6 @@ 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/misc.h
+++ b/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.
 **
--- 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 +50,19 @@ 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);

--- a/softhddevice.cpp
+++ b/softhddevice.cpp
--- a/video.c
+++ b/video.c
--- a/video.h
+++ b/video.h
@@ -30,6 +30,13 @@
    /// Video hardware decoder typedef
 typedef struct _video_hw_decoder_ VideoHwDecoder;

+//----------------------------------------------------------------------------
+//	Variables
+//----------------------------------------------------------------------------
+
+extern char VideoIgnoreRepeatPict;	///< disable repeat pict warning
+extern int VideoAudioDelay;		///< audio/video delay
+
 //----------------------------------------------------------------------------
 //	Prototypes
 //----------------------------------------------------------------------------
@@ -71,9 +78,18 @@ 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 video output position.
 extern void VideoSetOutputPosition(int, int, int, int);

@@ -104,12 +120,18 @@ 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);

@@ -125,7 +147,14 @@ 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 video clock.
+extern void VideoSetClock(VideoHwDecoder *, int64_t);
+
+    /// Get video clock.
+extern int64_t VideoGetClock(const VideoHwDecoder *);
+
+    /// Set trick play speed.
+extern void VideoSetTrickSpeed(VideoHwDecoder *, int);

    /// Grab screen.
 extern uint8_t *VideoGrab(int *, int *, int *, int);
@@ -138,5 +167,6 @@ 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	e10e62dcf7	Release Version 0.5.0.	2012-04-07 20:21:55 +02:00
Johns	2a1793c98e	Use config value to change audio/video delay.	2012-04-07 13:43:51 +02:00
Johns	30d4586448	Change audio/video delay with hot-key.	2012-04-07 00:15:57 +02:00
Johns	aa4debc9c8	Enable/disable/toggle fullscreen with hot-key.	2012-04-06 14:38:51 +02:00
CafeDelMar	ac2e10a308	Cutting pixels configured for each resolution.	2012-04-05 22:51:20 +02:00
Johns	c986d285ea	Buffer less video and audio.	2012-04-05 15:47:59 +02:00
Johns	8612044b9b	Move suspend on inactivity to houesekeeping.	2012-04-05 15:43:32 +02:00
Johns	c19b86411a	Update documents.	2012-04-05 15:42:48 +02:00
Johns	9165052d5e	Fix gcc error bug (also for VA-API).	2012-04-03 16:36:06 +02:00
Johns	413983a666	Calling TrickSpeed without decoder can happen.	2012-04-01 17:04:16 +02:00
Johns	f86fa4edd7	VDR 1.7.27 suggested change.	2012-04-01 17:02:15 +02:00
Johns	7f8110557f	Fix gcc error bug.	2012-03-31 23:20:06 +02:00
Johns	c9b344a3fd	Audio/Video sync rewrite. Trick-speed support moved to video module. Reduce video messages.	2012-03-31 21:27:54 +02:00
Johns	b41f934c37	Faster VdpauBlackSurface version.	2012-03-30 17:19:31 +02:00
Johns	6058f3da56	Fix bug: VideoSetPts wrong position.	2012-03-30 16:04:25 +02:00
CafeDelMar	689d75b808	Add VideoSkipPixels support.	2012-03-26 20:49:18 +02:00
Johns	bd4503f30b	More debug for flush buffers. Bigger audio buffer.	2012-03-23 18:43:20 +01:00
Johns	24ba8175a3	Disable suspend on inactivity until player fixed.	2012-03-22 16:06:32 +01:00
Johns	fe24cbb182	mp3 needs 100% cpu again!	2012-03-20 16:36:42 +01:00
Johns	6eff8fa818	Forgot VDPAU in requires.	2012-03-19 17:15:21 +01:00
Johns	552a994db3	Add optional argument to ATTA svdrp commmand.	2012-03-15 15:42:51 +01:00
Johns	d24f19bc2d	More SVDRP commands help.	2012-03-14 15:07:08 +01:00
Johns	7b570c507c	Cleanups.	2012-03-12 17:58:19 +01:00
Johns	09ba3e2993	Let inactivity suspend wakeup with remote keys.	2012-03-11 14:12:49 +01:00
Johns	d0f825f831	Comments added.	2012-03-10 17:46:00 +01:00
Johns	47d2896468	Better Poll(), flush video buffers after replay.	2012-03-10 17:05:41 +01:00
Johns	f59425ac57	AudioGetDelay returns signed value and cleanups.	2012-03-10 15:00:58 +01:00
Johns	1acdeee913	Adds ffmpeg 0.8.7 bug workaround: Single nal end seq aren't consumed and an endless loop entered.	2012-03-09 21:47:06 +01:00
Johns	c2938c7ef3	Wakeup display to show OSD for remote learning.	2012-03-09 12:08:56 +01:00
Johns	d65fe88c83	Support switching the primary device with svdrp.	2012-03-08 15:28:10 +01:00
Johns	7d3f4f4434	Disable and reenable screen saver and DPMS.	2012-03-08 15:25:10 +01:00
Johns	acc35fe30c	Video cleanup. Add noop video output module. Move VideoThread check into lock/unlock functions. Add support for choosing video output module.	2012-03-07 15:31:43 +01:00
Johns	ee5804fed7	Handle snd_pcm_wait timeouts.	2012-03-07 15:13:07 +01:00
Johns	1cbaddf75c	Need extra space in ring buffer for sequence end.	2012-03-06 18:37:40 +01:00
Johns	226760490b	VADisplayAttribDirectSurface removed.	2012-03-06 16:56:26 +01:00
Johns	7931909e28	Workaround should be for abs.	2012-03-06 15:39:29 +01:00
Johns	129c139ed7	Fix fast backward with some h264 streams.	2012-03-06 15:38:30 +01:00
Johns	340816d763	Make soft start sync setup menu configurable.	2012-03-06 12:16:47 +01:00
Johns	d6c6818ecf	Workaround for av_resample_compensate ffmpeg bug. FFmpeg commit a67cb012e6947fb238193afc0f18114f6e20818c or 1b9ca38d9d06d319fffd61d27e4eb385d6572ba8 breaks av_resample_compensate. Only big sample_delta compensation_distance ratios are now working.	2012-03-05 20:38:43 +01:00
Johns	181a0bb372	Move grab unsupported warning to low-level.	2012-03-05 20:10:23 +01:00
Johns	f2d4163899	Fix bug: NAL end of sequence is 10 and not 0x10. Cleanup, remove old cruft. Add support for pes recordings.	2012-03-05 20:05:56 +01:00
Johns	4cc98d7937	Move time-stamp printing to misc.h.	2012-03-05 17:34:10 +01:00
Johns	3812fa8d38	Fix bug: AudioEnqueue crash without sound card.	2012-03-05 15:06:46 +01:00
Johns	da5c5cd5fd	Version 0.4.9 released.	2012-03-04 22:36:14 +01:00
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