Improved pass-through (PCM+EAC3) support.

9 files changed, 449 insertions, 267 deletions

diff --git a/ChangeLog b/ChangeLog
index 20f0582..891fb78 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,6 +1,7 @@
 User johns
 Date:
 
+    Improved pass-through (PCM+EAC3) support.
     Support VDR 1.7.36 new build system.
     Improves VDPAU display preemption handling.
     Add modifiers to X11 remote key names (from Sibbi).
diff --git a/README.txt b/README.txt
index ac9562b..7b3f5fc 100644
--- a/README.txt
+++ b/README.txt
@@ -1,6 +1,6 @@
 @file README.txt		@brief A software HD output device for VDR
 
-Copyright (c) 2011, 2012 by Johns.  All Rights Reserved.
+Copyright (c) 2011 - 2013 by Johns.  All Rights Reserved.
 
 Contributor(s):
 
@@ -89,8 +89,8 @@ Setup:	environment
     only if alsa is configured
 	ALSA_DEVICE=default
 		alsa PCM device name
-	ALSA_AC3_DEVICE=
-		alsa AC3/pass-though device name
+	ALSA_PASSTHROUGH_DEVICE=
+		alsa pass-though (AC3,EAC3,DTS,...) device name
 	ALSA_MIXER=default
 		alsa control device name
 	ALSA_MIXER_CHANNEL=PCM
@@ -99,8 +99,8 @@ Setup:	environment
     only if oss is configured
 	OSS_AUDIODEV=/dev/dsp
 		oss dsp device name
-	OSS_AC3_AUDIODEV=
-		oss AC3/pass-though device name
+	OSS_PASSTHROUGHDEV=
+		oss pass-though (AC3,EAC3,DTS,...) device name
 	OSS_MIXERDEV=/dev/mixer
 		oss mixer device name
 	OSS_MIXER_CHANNEL=pcm
@@ -156,13 +156,16 @@ Setup: /etc/vdr/setup.conf
 	delay audio or delay video
 
 	softhddevice.AudioPassthrough = 0
-	0 = none, 1 = AC-3
+	0 = none, 1 = PCM, 2 = MPA, 4 = AC-3, 8 = EAC-3
 
-	for AC-3 the pass-through device is used.
+	for PCM/AC-3/EAC-3 the pass-through device is used and the audio
+	stream is passed undecoded to the output device.
+	z.b. 12 = AC-3+EAC-3, 13 = PCM+AC-3+EAC-3
+	note: MPA/DTS/TrueHD/... aren't supported yet
 
 	softhddevice.AudioDownmix = 0
 	0 = none, 1 = downmix
-	Use ffmpeg/libav downmix AC-3 to stereo.
+	Use ffmpeg/libav downmix of AC-3/EAC-3 audio to stereo.
 
 	softhddevice.AudioSoftvol = 0
 	0 = off, use hardware volume control
diff --git a/audio.c b/audio.c
index ed6198e..369eec9 100644
--- a/audio.c
+++ b/audio.c
@@ -129,7 +129,7 @@ static const char *AudioModuleName;	///< which audio module to use
     /// Selected audio module.
 static const AudioModule *AudioUsedModule = &NoopModule;
 static const char *AudioPCMDevice;	///< PCM device name
-static const char *AudioAC3Device;	///< AC3 device name
+static const char *AudioPassthroughDevice;	///< Passthrough device name
 static const char *AudioMixerDevice;	///< mixer device name
 static const char *AudioMixerChannel;	///< mixer channel name
 static char AudioDoingInit;		///> flag in init, reduce error
@@ -620,7 +620,7 @@ static void AudioResample(const int16_t * in, int in_chan, int frames,
 typedef struct _audio_ring_ring_
 {
     char FlushBuffers;			///< flag: flush buffers
-    char UseAc3;			///< flag: use ac3 pass-through
+    char Passthrough;			///< flag: use pass-through (AC3, ...)
     int16_t PacketSize;			///< packet size
     unsigned HwSampleRate;		///< hardware sample rate in Hz
     unsigned HwChannels;		///< hardware number of channels
@@ -642,14 +642,14 @@ static unsigned AudioStartThreshold;	///< start play, if filled
 **
 **	@param sample_rate	sample-rate frequency
 **	@param channels		number of channels
-**	@param use_ac3		use ac3/pass-through device
+**	@param passthrough	use /pass-through (AC3, ...) device
 **
 **	@retval -1	error
 **	@retval 0	okay
 **
 **	@note this function shouldn't fail.  Checks are done during AudoInit.
 */
-static int AudioRingAdd(unsigned sample_rate, int channels, int use_ac3)
+static int AudioRingAdd(unsigned sample_rate, int channels, int passthrough)
 {
     unsigned u;
 
@@ -680,7 +680,7 @@ static int AudioRingAdd(unsigned sample_rate, int channels, int use_ac3)
 
     // FIXME: don't flush buffers here
     AudioRing[AudioRingWrite].FlushBuffers = 1;
-    AudioRing[AudioRingWrite].UseAc3 = use_ac3;
+    AudioRing[AudioRingWrite].Passthrough = passthrough;
     AudioRing[AudioRingWrite].PacketSize = 0;
     AudioRing[AudioRingWrite].InSampleRate = sample_rate;
     AudioRing[AudioRingWrite].InChannels = channels;
@@ -836,8 +836,9 @@ static int AlsaPlayRingbuffer(void)
 	if (!avail) {			// full or buffer empty
 	    break;
 	}
-	// muting ac3, can produce disturbance
-	if (AudioMute || (AudioSoftVolume && !AudioRing[AudioRingRead].UseAc3)) {
+	// muting pass-through ac3, can produce disturbance
+	if (AudioMute || (AudioSoftVolume
+		&& !AudioRing[AudioRingRead].Passthrough)) {
 	    // FIXME: quick&dirty cast
 	    AudioSoftAmplifier((int16_t *) p, avail);
 	    // FIXME: if not all are written, we double amplify them
@@ -984,23 +985,23 @@ static int AlsaThread(void)
 /**
 **	Open alsa pcm device.
 **
-**	@param use_ac3	use ac3/pass-through device
+**	@param passthrough	use pass-through (AC3, ...) device
 */
-static snd_pcm_t *AlsaOpenPCM(int use_ac3)
+static snd_pcm_t *AlsaOpenPCM(int passthrough)
 {
     const char *device;
     snd_pcm_t *handle;
     int err;
 
     // &&|| hell
-    if (!(use_ac3 && ((device = AudioAC3Device)
-		|| (device = getenv("ALSA_AC3_DEVICE"))))
+    if (!(passthrough && ((device = AudioPassthroughDevice)
+		|| (device = getenv("ALSA_PASSTHROUGH_DEVICE"))))
 	&& !(device = AudioPCMDevice) && !(device = getenv("ALSA_DEVICE"))) {
 	device = "default";
     }
     if (!AudioDoingInit) {		// reduce blabla during init
-	Info(_("audio/alsa: using %sdevice '%s'\n"), use_ac3 ? "ac3 " : "",
-	    device);
+	Info(_("audio/alsa: using %sdevice '%s'\n"),
+	    passthrough ? "pass-through " : "", device);
     }
     // open none blocking; if device is already used, we don't want wait
     if ((err =
@@ -1167,7 +1168,7 @@ static int64_t AlsaGetDelay(void)
 **
 **	@param freq		sample frequency
 **	@param channels		number of channels
-**	@param use_ac3		use ac3/pass-through device
+**	@param passthrough	use pass-through (AC3, ...) device
 **
 **	@retval 0	everything ok
 **	@retval 1	didn't support frequency/channels combination
@@ -1175,7 +1176,7 @@ static int64_t AlsaGetDelay(void)
 **
 **	@todo FIXME: remove pointer for freq + channels
 */
-static int AlsaSetup(int *freq, int *channels, int use_ac3)
+static int AlsaSetup(int *freq, int *channels, int passthrough)
 {
     snd_pcm_uframes_t buffer_size;
     snd_pcm_uframes_t period_size;
@@ -1183,7 +1184,7 @@ static int AlsaSetup(int *freq, int *channels, int use_ac3)
     int delay;
 
     if (!AlsaPCMHandle) {		// alsa not running yet
-	// FIXME: if open fails for ac3, we never recover
+	// FIXME: if open fails for fe. pass-through, we never recover
 	return -1;
     }
     if (1) {				// close+open to fix HDMI no sound bug
@@ -1193,7 +1194,7 @@ static int AlsaSetup(int *freq, int *channels, int use_ac3)
 	// FIXME: need lock
 	AlsaPCMHandle = NULL;		// other threads should check handle
 	snd_pcm_close(handle);
-	if (!(handle = AlsaOpenPCM(use_ac3))) {
+	if (!(handle = AlsaOpenPCM(passthrough))) {
 	    return -1;
 	}
 	AlsaPCMHandle = handle;
@@ -1453,7 +1454,7 @@ static int OssPlayRingbuffer(void)
 	    break;			// bi.bytes could become negative!
 	}
 
-	if (AudioSoftVolume && !AudioRing[AudioRingRead].UseAc3) {
+	if (AudioSoftVolume && !AudioRing[AudioRingRead].Passthrough) {
 	    // FIXME: quick&dirty cast
 	    AudioSoftAmplifier((int16_t *) p, bi.bytes);
 	    // FIXME: if not all are written, we double amplify them
@@ -1560,22 +1561,22 @@ static int OssThread(void)
 /**
 **	Open OSS pcm device.
 **
-**	@param use_ac3	use ac3/pass-through device
+**	@param passthrough	use pass-through (AC3, ...) device
 */
-static int OssOpenPCM(int use_ac3)
+static int OssOpenPCM(int passthrough)
 {
     const char *device;
     int fildes;
 
     // &&|| hell
-    if (!(use_ac3 && ((device = AudioAC3Device)
-		|| (device = getenv("OSS_AC3_AUDIODEV"))))
+    if (!(passthrough && ((device = AudioPassthroughDevice)
+		|| (device = getenv("OSS_PASSTHROUGHDEV"))))
 	&& !(device = AudioPCMDevice) && !(device = getenv("OSS_AUDIODEV"))) {
 	device = "/dev/dsp";
     }
     if (!AudioDoingInit) {
-	Info(_("audio/oss: using %sdevice '%s'\n"), use_ac3 ? "ac3 " : "",
-	    device);
+	Info(_("audio/oss: using %sdevice '%s'\n"),
+	    passthrough ? "pass-through " : "", device);
     }
 
     if ((fildes = open(device, O_WRONLY)) < 0) {
@@ -1724,13 +1725,13 @@ static int64_t OssGetDelay(void)
 **
 **	@param sample_rate	sample rate/frequency
 **	@param channels		number of channels
-**	@param use_ac3		use ac3/pass-through device
+**	@param passthrough	use pass-through (AC3, ...) device
 **
 **	@retval 0	everything ok
 **	@retval 1	didn't support frequency/channels combination
 **	@retval -1	something gone wrong
 */
-static int OssSetup(int *sample_rate, int *channels, int use_ac3)
+static int OssSetup(int *sample_rate, int *channels, int passthrough)
 {
     int ret;
     int tmp;
@@ -1738,7 +1739,7 @@ static int OssSetup(int *sample_rate, int *channels, int use_ac3)
     audio_buf_info bi;
 
     if (OssPcmFildes == -1) {		// OSS not ready
-	// FIXME: if open fails for ac3, we never recover
+	// FIXME: if open fails for fe. pass-through, we never recover
 	return -1;
     }
 
@@ -1748,7 +1749,7 @@ static int OssSetup(int *sample_rate, int *channels, int use_ac3)
 	fildes = OssPcmFildes;
 	OssPcmFildes = -1;
 	close(fildes);
-	if (!(fildes = OssOpenPCM(use_ac3))) {
+	if (!(fildes = OssOpenPCM(passthrough))) {
 	    return -1;
 	}
 	OssPcmFildes = fildes;
@@ -1929,13 +1930,14 @@ static void NoopSetVolume( __attribute__ ((unused))
 /**
 **	Noop setup.
 **
-**	@param freq	sample frequency
-**	@param channels	number of channels
+**	@param freq		sample frequency
+**	@param channels		number of channels
+**	@param passthrough	use pass-through (AC3, ...) device
 */
 static int NoopSetup( __attribute__ ((unused))
     int *channels, __attribute__ ((unused))
     int *freq, __attribute__ ((unused))
-    int use_ac3)
+    int passthrough)
 {
     return -1;
 }
@@ -1973,17 +1975,17 @@ static const AudioModule NoopModule = {
 */
 static int AudioNextRing(void)
 {
-    int use_ac3;
+    int passthrough;
     int sample_rate;
     int channels;
     size_t used;
 
     // update audio format
     // not always needed, but check if needed is too complex
-    use_ac3 = AudioRing[AudioRingRead].UseAc3;
+    passthrough = AudioRing[AudioRingRead].Passthrough;
     sample_rate = AudioRing[AudioRingRead].HwSampleRate;
     channels = AudioRing[AudioRingRead].HwChannels;
-    if (AudioUsedModule->Setup(&sample_rate, &channels, use_ac3)) {
+    if (AudioUsedModule->Setup(&sample_rate, &channels, passthrough)) {
 	Error(_("audio: can't set channels %d sample-rate %dHz\n"), channels,
 	    sample_rate);
 	// FIXME: handle error
@@ -2068,10 +2070,10 @@ static void *AudioPlayHandlerThread(void *dummy)
 	    err = AudioUsedModule->Thread();
 	    // underrun, check if new ring buffer is available
 	    if (!err) {
-		int use_ac3;
+		int passthrough;
 		int sample_rate;
 		int channels;
-		int old_use_ac3;
+		int old_passthrough;
 		int old_sample_rate;
 		int old_channels;
 
@@ -2081,20 +2083,21 @@ static void *AudioPlayHandlerThread(void *dummy)
 		}
 
 		Debug(3, "audio: next ring buffer\n");
-		old_use_ac3 = AudioRing[AudioRingRead].UseAc3;
+		old_passthrough = AudioRing[AudioRingRead].Passthrough;
 		old_sample_rate = AudioRing[AudioRingRead].HwSampleRate;
 		old_channels = AudioRing[AudioRingRead].HwChannels;
 
 		atomic_dec(&AudioRingFilled);
 		AudioRingRead = (AudioRingRead + 1) % AUDIO_RING_MAX;
 
-		use_ac3 = AudioRing[AudioRingRead].UseAc3;
+		passthrough = AudioRing[AudioRingRead].Passthrough;
 		sample_rate = AudioRing[AudioRingRead].HwSampleRate;
 		channels = AudioRing[AudioRingRead].HwChannels;
 		Debug(3, "audio: thread channels %d frequency %dHz %s\n",
-		    channels, sample_rate, use_ac3 ? "ac3" : "pcm");
+		    channels, sample_rate, passthrough ? "pass-through" : "");
 		// audio config changed?
-		if (old_use_ac3 != use_ac3 || old_sample_rate != sample_rate
+		if (old_passthrough != passthrough
+		    || old_sample_rate != sample_rate
 		    || old_channels != channels) {
 		    // FIXME: wait for buffer drain
 		    if (AudioNextRing()) {
@@ -2196,7 +2199,7 @@ void AudioEnqueue(const void *samples, int count)
     }
     // audio sample modification allowed and needed?
     buffer = (void *)samples;
-    if (!AudioRing[AudioRingWrite].UseAc3 && (AudioCompression
+    if (!AudioRing[AudioRingWrite].Passthrough && (AudioCompression
 	    || AudioNormalize
 	    || AudioRing[AudioRingWrite].InChannels !=
 	    AudioRing[AudioRingWrite].HwChannels)) {
@@ -2416,7 +2419,7 @@ void AudioFlushBuffers(void)
     old = AudioRingWrite;
     AudioRingWrite = (AudioRingWrite + 1) % AUDIO_RING_MAX;
     AudioRing[AudioRingWrite].FlushBuffers = 1;
-    AudioRing[AudioRingWrite].UseAc3 = AudioRing[old].UseAc3;
+    AudioRing[AudioRingWrite].Passthrough = AudioRing[old].Passthrough;
     AudioRing[AudioRingWrite].HwSampleRate = AudioRing[old].HwSampleRate;
     AudioRing[AudioRingWrite].HwChannels = AudioRing[old].HwChannels;
     AudioRing[AudioRingWrite].InSampleRate = AudioRing[old].InSampleRate;
@@ -2528,7 +2531,7 @@ int64_t AudioGetClock(void)
 
 	// delay zero, if no valid time stamp
 	if ((delay = AudioGetDelay())) {
-	    if (AudioRing[AudioRingRead].UseAc3) {
+	    if (AudioRing[AudioRingRead].Passthrough) {
 		return AudioRing[AudioRingRead].PTS + 0 * 90 - delay;
 	    }
 	    return AudioRing[AudioRingRead].PTS + 0 * 90 - delay;
@@ -2548,7 +2551,7 @@ void AudioSetVolume(int volume)
     AudioMute = !volume;
     // reduce loudness for stereo output
     if (AudioStereoDescent && AudioRing[AudioRingRead].InChannels == 2
-	&& !AudioRing[AudioRingRead].UseAc3) {
+	&& !AudioRing[AudioRingRead].Passthrough) {
 	volume -= AudioStereoDescent;
 	if (volume < 0) {
 	    volume = 0;
@@ -2565,9 +2568,9 @@ void AudioSetVolume(int volume)
 /**
 **	Setup audio for requested format.
 **
-**	@param freq	sample frequency
-**	@param channels	number of channels
-**	@param use_ac3	use ac3/pass-through device
+**	@param freq		sample frequency
+**	@param channels		number of channels
+**	@param passthrough	use pass-through (AC3, ...) device
 **
 **	@retval 0	everything ok
 **	@retval 1	didn't support frequency/channels combination
@@ -2575,10 +2578,10 @@ void AudioSetVolume(int volume)
 **
 **	@todo add support to report best fitting format.
 */
-int AudioSetup(int *freq, int *channels, int use_ac3)
+int AudioSetup(int *freq, int *channels, int passthrough)
 {
     Debug(3, "audio: setup channels %d frequency %dHz %s\n", *channels, *freq,
-	use_ac3 ? "ac3" : "pcm");
+	passthrough ? "pass-through" : "");
 
     // invalid parameter
     if (!freq || !channels || !*freq || !*channels) {
@@ -2586,7 +2589,7 @@ int AudioSetup(int *freq, int *channels, int use_ac3)
 	// FIXME: set flag invalid setup
 	return -1;
     }
-    return AudioRingAdd(*freq, *channels, use_ac3);
+    return AudioRingAdd(*freq, *channels, passthrough);
 }
 
 /**
@@ -2722,7 +2725,7 @@ void AudioSetDevice(const char *device)
 **
 **	@note this is currently usable with alsa only.
 */
-void AudioSetDeviceAC3(const char *device)
+void AudioSetPassthroughDevice(const char *device)
 {
     if (!AudioModuleName) {
 	AudioModuleName = "alsa";	// detect alsa/OSS
@@ -2732,7 +2735,7 @@ void AudioSetDeviceAC3(const char *device)
 	    AudioModuleName = "oss";
 	}
     }
-    AudioAC3Device = device;
+    AudioPassthroughDevice = device;
 }
 
 /**
@@ -2961,7 +2964,7 @@ static void PrintVersion(void)
 #ifdef GIT_REV
 	"(GIT-" GIT_REV ")"
 #endif
-	",\n\t(c) 2009 - 2012 by Johns\n"
+	",\n\t(c) 2009 - 2013 by Johns\n"
 	"\tLicense AGPLv3: GNU Affero General Public License version 3\n");
 }
 
diff --git a/audio.h b/audio.h
index 808d385..4a0ac51 100644
--- a/audio.h
+++ b/audio.h
@@ -1,7 +1,7 @@
 ///
 ///	@file audio.h		@brief Audio module headerfile
 ///
-///	Copyright (c) 2009 - 2012 by Johns.  All Rights Reserved.
+///	Copyright (c) 2009 - 2013 by Johns.  All Rights Reserved.
 ///
 ///	Contributor(s):
 ///
@@ -48,7 +48,9 @@ extern void AudioSetCompression(int, int);	///< set compression parameters
 extern void AudioSetStereoDescent(int);	///< set stereo loudness descent
 
 extern void AudioSetDevice(const char *);	///< set PCM audio device
-extern void AudioSetDeviceAC3(const char *);	///< set pass-through device
+
+    /// set pass-through device
+extern void AudioSetPassthroughDevice(const char *);
 extern void AudioSetChannel(const char *);	///< set mixer channel
 extern void AudioInit(void);		///< setup audio module
 extern void AudioExit(void);		///< cleanup and exit audio module
diff --git a/codec.c b/codec.c
index e3860f7..13801dc 100644
--- a/codec.c
+++ b/codec.c
@@ -1,7 +1,7 @@
 ///
 ///	@file codec.c	@brief Codec functions
 ///
-///	Copyright (c) 2009 - 2012 by Johns.  All Rights Reserved.
+///	Copyright (c) 2009 - 2013 by Johns.  All Rights Reserved.
 ///
 ///	Contributor(s):
 ///
@@ -30,13 +30,13 @@
 ///		many bugs and incompatiblity in it.  Don't use this shit.
 ///
 
-    /// compile with passthrough support (stable, ac3 only)
+    /// compile with pass-through support (stable, AC-3, E-AC-3 only)
 #define USE_PASSTHROUGH
-    /// compile audio drift correction support (experimental)
+    /// compile audio drift correction support (very experimental)
 #define USE_AUDIO_DRIFT_CORRECTION
-    /// compile AC3 audio drift correction support (experimental)
+    /// compile AC-3 audio drift correction support (very experimental)
 #define USE_AC3_DRIFT_CORRECTION
-    /// use ffmpeg libswresample API
+    /// use ffmpeg libswresample API (autodected, Makefile)
 #define noUSE_SWRESAMPLE
 
 #include <stdio.h>
@@ -633,7 +633,7 @@ struct _audio_decoder_
     AVCodec *AudioCodec;		///< audio codec
     AVCodecContext *AudioCtx;		///< audio codec context
 
-    int PassthroughAC3;			///< current ac-3 pass-through
+    char Passthrough;			///< current pass-through flags
     int SampleRate;			///< current stream sample rate
     int Channels;			///< current stream channels
 
@@ -651,6 +651,10 @@ struct _audio_decoder_
 #endif
 #endif
 
+    uint16_t Spdif[24576 / 2];		///< SPDIF output buffer
+    int SpdifIndex;			///< index into SPDIF output buffer
+    int SpdifCount;			///< SPDIF repeat counter
+
     int64_t LastDelay;			///< last delay
     struct timespec LastTime;		///< last time
     int64_t LastPTS;			///< last PTS
@@ -670,24 +674,32 @@ struct _audio_decoder_
 #endif
 };
 
+///
+///	IEC Data type enumeration.
+///
+enum IEC61937
+{
+    IEC61937_AC3 = 0x01,		///< AC-3 data
+    // FIXME: more data types
+    IEC61937_EAC3 = 0x15,		///< E-AC-3 data
+};
+
 #ifdef USE_AUDIO_DRIFT_CORRECTION
 #define CORRECT_PCM	1		///< do PCM audio-drift correction
-#define CORRECT_AC3	2		///< do AC3§ audio-drift correction
+#define CORRECT_AC3	2		///< do AC3 audio-drift correction
 static char CodecAudioDrift;		///< flag: enable audio-drift correction
 #else
 static const int CodecAudioDrift = 0;
 #endif
 #ifdef USE_PASSTHROUGH
-//static char CodecPassthroughPCM;	///< pass pcm through (unsupported)
-static char CodecPassthroughAC3;	///< pass ac3 through
-
-//static char CodecPassthroughDTS;	///< pass dts through (unsupported)
-//static char CodecPassthroughMPA;	///< pass mpa through (unsupported)
+    ///
+    /// Pass-through flags: CodecPCM, CodecAC3, CodecEAC3, ...
+    ///
+static char CodecPassthrough;
 #else
-
-static const int CodecPassthroughAC3 = 0;
+static const int CodecPassthrough = 0;
 #endif
-static char CodecDownmix;		///< enable ac-3 downmix
+static char CodecDownmix;		///< enable AC-3 decoder downmix
 
 /**
 **	Allocate a new audio decoder context.
@@ -840,7 +852,7 @@ void CodecAudioClose(AudioDecoder * audio_decoder)
 void CodecSetAudioDrift(int mask)
 {
 #ifdef USE_AUDIO_DRIFT_CORRECTION
-    CodecAudioDrift = mask & 3;
+    CodecAudioDrift = mask & (CORRECT_PCM | CORRECT_AC3);
 #endif
     (void)mask;
 }
@@ -848,12 +860,12 @@ void CodecSetAudioDrift(int mask)
 /**
 **	Set audio pass-through.
 **
-**	@param mask	enable mask (PCM, AC3)
+**	@param mask	enable mask (PCM, AC3, EAC3)
 */
 void CodecSetAudioPassthrough(int mask)
 {
 #ifdef USE_PASSTHROUGH
-    CodecPassthroughAC3 = mask & 1 ? 1 : 0;
+    CodecPassthrough = mask & (CodecPCM | CodecAC3 | CodecEAC3);
 #endif
     (void)mask;
 }
@@ -932,6 +944,178 @@ static void CodecReorderAudioFrame(int16_t * buf, int size, int channels)
     }
 }
 
+/**
+**	Handle audio format changes helper.
+**
+**	@param audio_decoder	audio decoder data
+**	@param[out] passthrough	pass-through output
+*/
+static int CodecAudioUpdateHelper(AudioDecoder * audio_decoder,
+    int *passthrough)
+{
+    const AVCodecContext *audio_ctx;
+    int err;
+
+    audio_ctx = audio_decoder->AudioCtx;
+    Debug(3, "codec/audio: format change %s %dHz *%d channels%s%s%s%s%s\n",
+	av_get_sample_fmt_name(audio_ctx->sample_fmt), audio_ctx->sample_rate,
+	audio_ctx->channels, CodecPassthrough & CodecPCM ? " PCM" : "",
+	CodecPassthrough & CodecMPA ? " MPA" : "",
+	CodecPassthrough & CodecAC3 ? " AC3" : "",
+	CodecPassthrough & CodecEAC3 ? " EAC3" : "",
+	CodecPassthrough ? " pass-through" : "");
+
+    *passthrough = 0;
+    audio_decoder->SampleRate = audio_ctx->sample_rate;
+    audio_decoder->HwSampleRate = audio_ctx->sample_rate;
+    audio_decoder->Channels = audio_ctx->channels;
+    audio_decoder->HwChannels = audio_ctx->channels;
+    audio_decoder->Passthrough = CodecPassthrough;
+
+    // SPDIF/HDMI pass-through
+    if ((CodecPassthrough & CodecAC3 && audio_ctx->codec_id == CODEC_ID_AC3)
+	|| (CodecPassthrough & CodecEAC3
+	    && audio_ctx->codec_id == CODEC_ID_EAC3)) {
+	audio_decoder->HwChannels = 2;
+	audio_decoder->SpdifIndex = 0;	// reset buffer
+	audio_decoder->SpdifCount = 0;
+	*passthrough = 1;
+    }
+    // channels not support?
+    if ((err =
+	    AudioSetup(&audio_decoder->HwSampleRate,
+		&audio_decoder->HwChannels, *passthrough))) {
+
+	Debug(3, "codec/audio: audio setup error\n");
+	// FIXME: handle errors
+	audio_decoder->HwChannels = 0;
+	audio_decoder->HwSampleRate = 0;
+	return err;
+    }
+
+    Debug(3, "codec/audio: resample %s %dHz *%d -> %s %dHz *%d\n",
+	av_get_sample_fmt_name(audio_ctx->sample_fmt), audio_ctx->sample_rate,
+	audio_ctx->channels, av_get_sample_fmt_name(AV_SAMPLE_FMT_S16),
+	audio_decoder->HwSampleRate, audio_decoder->HwChannels);
+
+    return 0;
+}
+
+/**
+**	Audio pass-through decoder helper.
+**
+**	@param audio_decoder	audio decoder data
+**	@param avpkt		undecoded audio packet
+*/
+static int CodecAudioPassthroughHelper(AudioDecoder * audio_decoder,
+    const AVPacket * avpkt)
+{
+#ifdef USE_PASSTHROUGH
+    const AVCodecContext *audio_ctx;
+
+    audio_ctx = audio_decoder->AudioCtx;
+    // SPDIF/HDMI passthrough
+    if (CodecPassthrough & CodecAC3 && audio_ctx->codec_id == CODEC_ID_AC3) {
+	uint16_t *spdif;
+	int spdif_sz;
+
+	spdif = audio_decoder->Spdif;
+	spdif_sz = 6144;
+
+#ifdef USE_AC3_DRIFT_CORRECTION
+	// FIXME: this works with some TVs/AVReceivers
+	// FIXME: write burst size drift correction, which should work with all
+	if (CodecAudioDrift & CORRECT_AC3) {
+	    int x;
+
+	    x = (audio_decoder->DriftFrac +
+		(audio_decoder->DriftCorr * spdif_sz)) / (10 *
+		audio_decoder->HwSampleRate * 100);
+	    audio_decoder->DriftFrac =
+		(audio_decoder->DriftFrac +
+		(audio_decoder->DriftCorr * spdif_sz)) % (10 *
+		audio_decoder->HwSampleRate * 100);
+	    // round to word border
+	    x *= audio_decoder->HwChannels * 4;
+	    if (x < -64) {		// limit correction
+		x = -64;
+	    } else if (x > 64) {
+		x = 64;
+	    }
+	    spdif_sz += x;
+	}
+#endif
+
+	// build SPDIF header and append A52 audio to it
+	// avpkt is the original data
+	if (spdif_sz < avpkt->size + 8) {
+	    Error(_("codec/audio: decoded data smaller than encoded\n"));
+	    return -1;
+	}
+	spdif[0] = htole16(0xF872);	// iec 61937 sync word
+	spdif[1] = htole16(0x4E1F);
+	spdif[2] = htole16(IEC61937_AC3 | (avpkt->data[5] & 0x07) << 8);
+	spdif[3] = htole16(avpkt->size * 8);
+	// copy original data for output
+	// FIXME: not 100% sure, if endian is correct on not intel hardware
+	swab(avpkt->data, spdif + 4, avpkt->size);
+	// FIXME: don't need to clear always
+	memset(spdif + 4 + avpkt->size / 2, 0, spdif_sz - 8 - avpkt->size);
+	// don't play with the ac-3 samples
+	AudioEnqueue(spdif, spdif_sz);
+	return 1;
+    }
+    if (CodecPassthrough & CodecEAC3 && audio_ctx->codec_id == CODEC_ID_EAC3) {
+	uint16_t *spdif;
+	int spdif_sz;
+	int repeat;
+
+	// build SPDIF header and append A52 audio to it
+	// avpkt is the original data
+	spdif = audio_decoder->Spdif;
+	spdif_sz = 6144;
+	// 24576 = 4 * 6144
+	if (spdif_sz < audio_decoder->SpdifIndex + avpkt->size + 8) {
+	    Error(_("codec/audio: decoded data smaller than encoded\n"));
+	    return -1;
+	}
+	// check if we must pack multiple packets
+	repeat = 1;
+	if ((avpkt->data[4] & 0xc0) != 0xc0) {	// fscod
+	    static const uint8_t eac3_repeat[4] = { 6, 3, 2, 1 };
+
+	    // fscod2
+	    repeat = eac3_repeat[(avpkt->data[4] & 0x30) >> 4];
+	}
+	//fprintf(stderr, "repeat %d\n", repeat);
+
+	// copy original data for output
+	// pack upto repeat EAC-3 pakets into one IEC 61937 burst
+	// FIXME: not 100% sure, if endian is correct on not intel hardware
+	swab(avpkt->data, spdif + 4 + audio_decoder->SpdifIndex, avpkt->size);
+	audio_decoder->SpdifIndex += avpkt->size;
+	if (++audio_decoder->SpdifCount < repeat) {
+	    return 1;
+	}
+
+	spdif[0] = htole16(0xF872);	// iec 61937 sync word
+	spdif[1] = htole16(0x4E1F);
+	spdif[2] = htole16(IEC61937_EAC3);
+	spdif[3] = htole16(audio_decoder->SpdifIndex * 8);
+	memset(spdif + 4 + audio_decoder->SpdifIndex / 2, 0,
+	    spdif_sz - 8 - audio_decoder->SpdifIndex);
+
+	// don't play with the eac-3 samples
+	AudioEnqueue(spdif, spdif_sz);
+
+	audio_decoder->SpdifIndex = 0;
+	audio_decoder->SpdifCount = 0;
+	return 1;
+    }
+#endif
+    return 0;
+}
+
 #ifndef USE_SWRESAMPLE
 
 /**
@@ -1007,8 +1191,10 @@ static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
 	audio_decoder->Drift = drift;
 	corr = (10 * audio_decoder->HwSampleRate * drift) / (90 * 1000);
 	// SPDIF/HDMI passthrough
-	if ((CodecAudioDrift & 2) && (!CodecPassthroughAC3
-		|| audio_decoder->AudioCtx->codec_id != CODEC_ID_AC3)) {
+	if ((CodecAudioDrift & CORRECT_AC3) && (!CodecPassthroughAC3
+		|| audio_decoder->AudioCtx->codec_id != CODEC_ID_AC3)
+	    && (!CodecPassthroughEAC3
+		|| audio_decoder->AudioCtx->codec_id != CODEC_ID_EAC3)) {
 	    audio_decoder->DriftCorr = -corr;
 	}
 
@@ -1045,14 +1231,15 @@ static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
 **	Handle audio format changes.
 **
 **	@param audio_decoder	audio decoder data
+**
+**	@note this is the old not good supported version
 */
 static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
 {
+    int passthrough;
     const AVCodecContext *audio_ctx;
     int err;
-    int isAC3;
 
-    // FIXME: use swr_convert from swresample (only in ffmpeg!)
     if (audio_decoder->ReSample) {
 	audio_resample_close(audio_decoder->ReSample);
 	audio_decoder->ReSample = NULL;
@@ -1064,28 +1251,8 @@ static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
     }
 
     audio_ctx = audio_decoder->AudioCtx;
-    Debug(3, "codec/audio: format change %dHz %d channels %s\n",
-	audio_ctx->sample_rate, audio_ctx->channels,
-	CodecPassthroughAC3 ? "pass-through" : "");
-
-    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;
-    }
+    if ((err = CodecAudioUpdateHelper(audio_decoder, &passthrough))) {
 
-    // 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);
@@ -1101,19 +1268,21 @@ static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
 		Error(_("codec/audio: resample setup error\n"));
 		audio_decoder->HwChannels = 0;
 		audio_decoder->HwSampleRate = 0;
-		return;
 	    }
-	} else {
-	    Debug(3, "codec/audio: audio setup error\n");
-	    // FIXME: handle errors
-	    audio_decoder->HwChannels = 0;
-	    audio_decoder->HwSampleRate = 0;
 	    return;
 	}
+	Debug(3, "codec/audio: audio setup error\n");
+	// FIXME: handle errors
+	audio_decoder->HwChannels = 0;
+	audio_decoder->HwSampleRate = 0;
+	return;
+    }
+    if (passthrough) {			// pass-through no conversion allowed
+	return;
     }
     // prepare audio drift resample
 #ifdef USE_AUDIO_DRIFT_CORRECTION
-    if ((CodecAudioDrift & 1) && !isAC3) {
+    if (CodecAudioDrift & CORRECT_PCM) {
 	if (audio_decoder->AvResample) {
 	    Error(_("codec/audio: overwrite resample\n"));
 	}
@@ -1144,7 +1313,7 @@ static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
 void CodecAudioEnqueue(AudioDecoder * audio_decoder, int16_t * data, int count)
 {
 #ifdef USE_AUDIO_DRIFT_CORRECTION
-    if ((CodecAudioDrift & 1) && audio_decoder->AvResample) {
+    if ((CodecAudioDrift & CORRECT_PCM) && audio_decoder->AvResample) {
 	int16_t buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 +
 	    FF_INPUT_BUFFER_PADDING_SIZE] __attribute__ ((aligned(16)));
 	int16_t buftmp[MAX_CHANNELS][(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4];
@@ -1205,12 +1374,16 @@ void CodecAudioEnqueue(AudioDecoder * audio_decoder, int16_t * data, int count)
 	n *= 2;
 
 	n *= audio_decoder->HwChannels;
-	CodecReorderAudioFrame(buf, n, audio_decoder->HwChannels);
+	if (!(audio_decoder->Passthrough & CodecPCM)) {
+	    CodecReorderAudioFrame(buf, n, audio_decoder->HwChannels);
+	}
 	AudioEnqueue(buf, n);
 	return;
     }
 #endif
-    CodecReorderAudioFrame(data, count, audio_decoder->HwChannels);
+    if (!(audio_decoder->Passthrough & CodecPCM)) {
+	CodecReorderAudioFrame(data, count, audio_decoder->HwChannels);
+    }
     AudioEnqueue(data, count);
 }
 
@@ -1232,6 +1405,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
 
     audio_ctx = audio_decoder->AudioCtx;
 
+    // FIXME: don't need to decode pass-through codecs
     buf_sz = sizeof(buf);
     l = avcodec_decode_audio3(audio_ctx, buf, &buf_sz, (AVPacket *) avpkt);
     if (avpkt->size != l) {
@@ -1250,7 +1424,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
 	CodecAudioSetClock(audio_decoder, avpkt->pts);
     }
     // FIXME: must first play remainings bytes, than change and play new.
-    if (audio_decoder->PassthroughAC3 != CodecPassthroughAC3
+    if (audio_decoder->Passthrough != CodecPassthrough
 	|| audio_decoder->SampleRate != audio_ctx->sample_rate
 	|| audio_decoder->Channels != audio_ctx->channels) {
 	CodecAudioUpdateFormat(audio_decoder);
@@ -1283,48 +1457,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
 		CodecAudioEnqueue(audio_decoder, outbuf, outlen);
 	    }
 	} else {
-#ifdef USE_PASSTHROUGH
-	    // SPDIF/HDMI passthrough
-	    if (CodecPassthroughAC3 && audio_ctx->codec_id == CODEC_ID_AC3) {
-		// build SPDIF header and append A52 audio to it
-		// avpkt is the original data
-		buf_sz = 6144;
-
-#ifdef USE_AC3_DRIFT_CORRECTION
-		if (CodecAudioDrift & 2) {
-		    int x;
-
-		    x = (audio_decoder->DriftFrac +
-			(audio_decoder->DriftCorr * buf_sz)) / (10 *
-			audio_decoder->HwSampleRate * 100);
-		    audio_decoder->DriftFrac =
-			(audio_decoder->DriftFrac +
-			(audio_decoder->DriftCorr * buf_sz)) % (10 *
-			audio_decoder->HwSampleRate * 100);
-		    x *= audio_decoder->HwChannels * 4;
-		    if (x < -64) {	// limit correction
-			x = -64;
-		    } else if (x > 64) {
-			x = 64;
-		    }
-		    buf_sz += x;
-		}
-#endif
-		if (buf_sz < avpkt->size + 8) {
-		    Error(_
-			("codec/audio: decoded data smaller than encoded\n"));
-		    return;
-		}
-		// copy original data for output
-		// FIXME: not 100% sure, if endian is correct
-		buf[0] = htole16(0xF872);	// iec 61937 sync word
-		buf[1] = htole16(0x4E1F);
-		buf[2] = htole16(0x01 | (avpkt->data[5] & 0x07) << 8);
-		buf[3] = htole16(avpkt->size * 8);
-		swab(avpkt->data, buf + 4, avpkt->size);
-		memset(buf + 4 + avpkt->size / 2, 0, buf_sz - 8 - avpkt->size);
-		// don't play with the ac-3 samples
-		AudioEnqueue(buf, buf_sz);
+	    if (CodecAudioPassthroughHelper(audio_decoder, avpkt)) {
 		return;
 	    }
 #if 0
@@ -1378,7 +1511,6 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
 	    // DTS HD?
 	    // True HD?
 #endif
-#endif
 	    CodecAudioEnqueue(audio_decoder, buf, buf_sz);
 	}
     }
@@ -1461,8 +1593,10 @@ static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
 	audio_decoder->Drift = drift;
 	corr = (10 * audio_decoder->HwSampleRate * drift) / (90 * 1000);
 	// SPDIF/HDMI passthrough
-	if ((CodecAudioDrift & 2) && (!CodecPassthroughAC3
-		|| audio_decoder->AudioCtx->codec_id != CODEC_ID_AC3)) {
+	if ((CodecAudioDrift & CORRECT_AC3) && (!(CodecPassthrough & CodecAC3)
+		|| audio_decoder->AudioCtx->codec_id != CODEC_ID_AC3)
+	    && (!(CodecPassthrough & CodecEAC3)
+		|| audio_decoder->AudioCtx->codec_id != CODEC_ID_EAC3)) {
 	    audio_decoder->DriftCorr = -corr;
 	}
 
@@ -1504,49 +1638,27 @@ static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
 */
 static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
 {
+    int passthrough;
     const AVCodecContext *audio_ctx;
-    int err;
-    int isAC3;
-
-    audio_ctx = audio_decoder->AudioCtx;
-    Debug(3, "codec/audio: format change %s %dHz *%d channels %s\n",
-	av_get_sample_fmt_name(audio_ctx->sample_fmt), 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: audio setup error\n");
-	// FIXME: handle errors
-	audio_decoder->HwChannels = 0;
-	audio_decoder->HwSampleRate = 0;
+    if (CodecAudioUpdateHelper(audio_decoder, &passthrough)) {
+	// FIXME: handle swresample format conversions.
 	return;
     }
-
-    if (isAC3) {			// no AC3 conversion allowed
+    if (passthrough) {			// pass-through no conversion allowed
 	return;
     }
 
-    Debug(3, "codec/audio: resample %s %dHz *%d -> %s %dHz *%d\n",
-	av_get_sample_fmt_name(audio_ctx->sample_fmt), audio_ctx->sample_rate,
-	audio_ctx->channels, av_get_sample_fmt_name(AV_SAMPLE_FMT_S16),
-	audio_decoder->HwSampleRate, audio_decoder->HwChannels);
+    audio_ctx = audio_decoder->AudioCtx;
+
+#ifdef DEBUG
+    if (audio_ctx->sample_fmt == AV_SAMPLE_FMT_S16
+	&& audio_ctx->sample_rate == audio_decoder->HwSampleRate
+	&& !CodecAudioDrift) {
+	// FIXME: use Resample only, when it is needed!
+	fprintf(stderr, "no resample needed\n");
+    }
+#endif
 
     audio_decoder->Resample =
 	swr_alloc_set_opts(audio_decoder->Resample, audio_ctx->channel_layout,
@@ -1579,6 +1691,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
 
     audio_ctx = audio_decoder->AudioCtx;
 
+    // FIXME: don't need to decode pass-through codecs
     frame.data[0] = NULL;
     n = avcodec_decode_audio4(audio_ctx, &frame, &got_frame,
 	(AVPacket *) avpkt);
@@ -1602,42 +1715,20 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
 	CodecAudioSetClock(audio_decoder, avpkt->pts);
     }
     // format change
-    if (audio_decoder->PassthroughAC3 != CodecPassthroughAC3
+    if (audio_decoder->Passthrough != CodecPassthrough
 	|| audio_decoder->SampleRate != audio_ctx->sample_rate
 	|| audio_decoder->Channels != audio_ctx->channels) {
 	CodecAudioUpdateFormat(audio_decoder);
-
     }
 
     if (!audio_decoder->HwSampleRate || !audio_decoder->HwChannels) {
 	return;				// unsupported sample format
     }
-#ifdef USE_PASSTHROUGH
-    // SPDIF/HDMI passthrough
-    if (CodecPassthroughAC3 && audio_ctx->codec_id == CODEC_ID_AC3) {
-	int16_t spdif[6144 / 2];
-	int spdif_sz;
 
-	// build SPDIF header and append A52 audio to it
-	// avpkt is the original data
-	spdif_sz = 6144;
-	if (spdif_sz < avpkt->size + 8) {
-	    Error(_("codec/audio: decoded data smaller than encoded\n"));
-	    return;
-	}
-	// copy original data for output
-	spdif[0] = htole16(0xF872);	// iec 61937 sync word
-	spdif[1] = htole16(0x4E1F);
-	spdif[2] = htole16(0x01 | (avpkt->data[5] & 0x07) << 8);
-	spdif[3] = htole16(avpkt->size * 8);
-	// FIXME: not 100% sure, if endian is correct on not intel hardware
-	swab(avpkt->data, spdif + 4, avpkt->size);
-	memset(spdif + 4 + avpkt->size / 2, 0, spdif_sz - 8 - avpkt->size);
-	// don't play with the ac-3 samples
-	AudioEnqueue(spdif, spdif_sz);
+    if (CodecAudioPassthroughHelper(audio_decoder, avpkt)) {
 	return;
     }
-#endif
+
     if (0) {
 	char strbuf[32];
 	int data_sz;
@@ -1665,12 +1756,19 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
 	    sizeof(outbuf) / (2 * audio_decoder->HwChannels),
 	    (const uint8_t **)frame.extended_data, frame.nb_samples);
 	if (n > 0) {
-	    CodecReorderAudioFrame((int16_t *) outbuf,
-		n * 2 * audio_decoder->HwChannels, audio_decoder->HwChannels);
+	    if (!(audio_decoder->Passthrough & CodecPCM)) {
+		CodecReorderAudioFrame((int16_t *) outbuf,
+		    n * 2 * audio_decoder->HwChannels,
+		    audio_decoder->HwChannels);
+	    }
 	    AudioEnqueue(outbuf, n * 2 * audio_decoder->HwChannels);
 	}
 	return;
     }
+#ifdef DEBUG
+    // should be never reached
+    fprintf(stderr, "oops\n");
+#endif
 }
 
 #endif
diff --git a/codec.h b/codec.h
index 08cc5ac..4a11607 100644
--- a/codec.h
+++ b/codec.h
@@ -1,7 +1,7 @@
 ///
 ///	@file codec.h	@brief Codec module headerfile
 ///
-///	Copyright (c) 2009 - 2012 by Johns.  All Rights Reserved.
+///	Copyright (c) 2009 - 2013 by Johns.  All Rights Reserved.
 ///
 ///	Contributor(s):
 ///
@@ -24,6 +24,16 @@
 /// @{
 
 //----------------------------------------------------------------------------
+//	Defines
+//----------------------------------------------------------------------------
+
+#define CodecPCM 0x01			///< PCM bit mask
+#define CodecMPA 0x02			///< MPA bit mask (planned)
+#define CodecAC3 0x04			///< AC-3 bit mask
+#define CodecEAC3 0x08			///< EAC-3 bit mask
+#define CodecDTS 0x10			///< DTS bit mask (planned)
+
+//----------------------------------------------------------------------------
 //	Typedefs
 //----------------------------------------------------------------------------
 
diff --git a/softhddev.c b/softhddev.c
index 3cded90..c638ebb 100644
--- a/softhddev.c
+++ b/softhddev.c
@@ -314,7 +314,7 @@ const uint16_t Ac3FrameSizeTable[38][3] = {
 };
 
 ///
-///	Fast check for AC3 audio.
+///	Fast check for (E)AC3 audio.
 ///
 ///	5 bytes 0x0B77xxxxxx AC3 audio
 ///
@@ -326,17 +326,11 @@ static inline int FastAc3Check(const uint8_t * p)
     if (p[1] != 0x77) {
 	return 0;
     }
-    if ((p[4] & 0xC0) == 0xC0) {	// invalid sample rate
-	return 0;
-    }
-    if ((p[4] & 0x3F) > 37) {		// invalid frame size
-	return 0;
-    }
     return 1;
 }
 
 ///
-///	Check for AC-3 audio.
+///	Check for (E)AC-3 audio.
 ///
 ///	0x0B77xxxxxx already checked.
 ///
@@ -347,20 +341,61 @@ static inline int FastAc3Check(const uint8_t * p)
 ///	@retval 0	no valid AC-3 audio
 ///	@retval >0	valid AC-3 audio
 ///
+///	o AC3 Header
+///	AAAAAAAA AAAAAAAA BBBBBBBB BBBBBBBB CCDDDDDD EEEEEFFF
+///
+///	o a 16x Frame sync, always 0x0B77
+///	o b 16x CRC 16
+///	o c 2x	Samplerate
+///	o d 6x	Framesize code
+///	o e 5x	Bitstream ID
+///	o f 3x	Bitstream mode
+///
+///	o EAC3 Header
+///	AAAAAAAA AAAAAAAA BBCCCDDD DDDDDDDD EEFFGGGH IIIII...
+///
+///	o a 16x Frame sync, always 0x0B77
+///	o b 2x	Frame type
+///	o c 3x	Sub stream ID
+///	o d 10x Framesize - 1 in words
+///	o e 2x	Framesize code
+///	o f 2x	Framesize code 2
+///
 static int Ac3Check(const uint8_t * data, int size)
 {
-    int fscod;
-    int frmsizcod;
     int frame_size;
 
-    // crc1 crc1 fscod|frmsizcod
-    fscod = data[4] >> 6;
-    frmsizcod = data[4] & 0x3F;		// invalid is checked by fast
-    frame_size = Ac3FrameSizeTable[frmsizcod][fscod] * 2;
+    if (size < 5) {			// need 5 bytes to see if AC3/EAC3
+	return -5;
+    }
+
+    if (data[5] > (10 << 3)) {		// EAC3
+	if ((data[4] & 0xF0) == 0xF0) {	// invalid fscod fscod2
+	    return 0;
+	}
+	frame_size = ((data[2] & 0x03) << 8) + data[3] + 1;
+	frame_size *= 2;
+    } else {				// AC3
+	int fscod;
+	int frmsizcod;
+
+	// crc1 crc1 fscod|frmsizcod
+	fscod = data[4] >> 6;
+	if (fscod == 0x03) {		// invalid sample rate
+	    return 0;
+	}
+	frmsizcod = data[4] & 0x3F;
+	if (frmsizcod > 37) {		// invalid frame size
+	    return 0;
+	}
+	// invalid is checked above
+	frame_size = Ac3FrameSizeTable[frmsizcod][fscod] * 2;
+    }
 
     if (frame_size + 5 > size) {
 	return -frame_size - 5;
     }
+    // FIXME: relaxed checks if codec is already detected
     // check if after this frame a new AC-3 frame starts
     if (FastAc3Check(data + frame_size)) {
 	return frame_size;
@@ -617,6 +652,7 @@ static void PesParse(PesDemux * pesdx, const uint8_t * data, int size,
 
 		    // 4 bytes 0xFFExxxxx Mpeg audio
 		    // 5 bytes 0x0B77xxxxxx AC3 audio
+		    // 6 bytes 0x0B77xxxxxxxx EAC3 audio
 		    // 3 bytes 0x56Exxx AAC LATM audio
 		    // 7/9 bytes 0xFFFxxxxxxxxxxx ADTS audio
 		    // PCM audio can't be found
@@ -629,6 +665,9 @@ static void PesParse(PesDemux * pesdx, const uint8_t * data, int size,
 		    if (!r && FastAc3Check(q)) {
 			r = Ac3Check(q, n);
 			codec_id = CODEC_ID_AC3;
+			if (r > 0 && q[5] > (10 << 3)) {
+			    codec_id = CODEC_ID_EAC3;
+			}
 		    }
 		    if (!r && FastLatmCheck(q)) {
 			r = LatmCheck(q, n);
@@ -1119,6 +1158,7 @@ int PlayAudio(const uint8_t * data, int size, uint8_t id)
 	// 4 bytes 0xFFExxxxx Mpeg audio
 	// 3 bytes 0x56Exxx AAC LATM audio
 	// 5 bytes 0x0B77xxxxxx AC3 audio
+	// 6 bytes 0x0B77xxxxxxxx EAC3 audio
 	// 7/9 bytes 0xFFFxxxxxxxxxxx ADTS audio
 	// PCM audio can't be found
 	r = 0;
@@ -1134,6 +1174,9 @@ int PlayAudio(const uint8_t * data, int size, uint8_t id)
 	if ((id == 0xbd || (id & 0xF0) == 0x80) && !r && FastAc3Check(p)) {
 	    r = Ac3Check(p, n);
 	    codec_id = CODEC_ID_AC3;
+	    if (r > 0 && p[5] > (10 << 3)) {
+		codec_id = CODEC_ID_EAC3;
+	    }
 	    /* faster ac3 detection at end of pes packet (no improvemnts)
 	       if (AudioCodecID == codec_id && -r - 2 == n) {
 	       r = n;
@@ -2787,7 +2830,7 @@ int ProcessArgs(int argc, char *const argv[])
 		AudioSetChannel(optarg);
 		continue;
 	    case 'p':			// pass-through audio device
-		AudioSetDeviceAC3(optarg);
+		AudioSetPassthroughDevice(optarg);
 		continue;
 	    case 'd':			// x11 display name
 		X11DisplayName = optarg;
diff --git a/softhddev.h b/softhddev.h
index 69fa32e..6e446dc 100644
--- a/softhddev.h
+++ b/softhddev.h
@@ -106,6 +106,8 @@ extern "C"
     extern void PipStop(void);
     /// Pip play video packet
     extern int PipPlayVideo(const uint8_t *, int);
+
+    extern const char *X11DisplayName;	///< x11 display name
 #ifdef __cplusplus
 }
 #endif
diff --git a/softhddevice.cpp b/softhddevice.cpp
index a9c2ab3..9f35371 100644
--- a/softhddevice.cpp
+++ b/softhddevice.cpp
@@ -20,6 +20,8 @@
 ///	$Id$
 //////////////////////////////////////////////////////////////////////////////
 
+#define __STDC_CONSTANT_MACROS		///< needed for ffmpeg UINT64_C
+
 #include <vdr/interface.h>
 #include <vdr/plugin.h>
 #include <vdr/player.h>
@@ -34,15 +36,15 @@
 #include "softhddev.h"
 #include "softhddevice.h"
 #include "softhddevice_service.h"
+
 extern "C"
 {
+#include <stdint.h>
+#include <libavcodec/avcodec.h>
+
 #include "audio.h"
 #include "video.h"
-    extern const char *X11DisplayName;	///< x11 display name
-
-    extern void CodecSetAudioDrift(int);
-    extern void CodecSetAudioPassthrough(int);
-    extern void CodecSetAudioDownmix(int);
+#include "codec.h"
 }
 
 //////////////////////////////////////////////////////////////////////////////
@@ -126,7 +128,8 @@ static int ConfigAutoCropTolerance;	///< auto crop detection tolerance
 
 static int ConfigVideoAudioDelay;	///< config audio delay
 static char ConfigAudioDrift;		///< config audio drift
-static char ConfigAudioPassthrough;	///< config audio pass-through
+static char ConfigAudioPassthrough;	///< config audio pass-through mask
+static char AudioPassthroughState;	///< flag audio pass-through on/off
 static char ConfigAudioDownmix;		///< config ffmpeg audio downmix
 static char ConfigAudioSoftvol;		///< config use software volume
 static char ConfigAudioNormalize;	///< config use normalize volume
@@ -138,7 +141,7 @@ int ConfigAudioBufferTime;		///< config size ms of audio buffer
 
 static char *ConfigX11Display;		///< config x11 display
 static char *ConfigAudioDevice;		///< config audio stereo device
-static char *ConfigAC3Device;		///< config audio passthrough device
+static char *ConfigPassthroughDevice;	///< config audio pass-through device
 
 #ifdef USE_PIP
 static int ConfigPipX = 100 - 3 - 18;	///< config pip pip x in %
@@ -615,7 +618,9 @@ class cMenuSetupSoft:public cMenuSetupPage
     int Audio;
     int AudioDelay;
     int AudioDrift;
-    int AudioPassthrough;
+    int AudioPassthroughPCM;
+    int AudioPassthroughAC3;
+    int AudioPassthroughEAC3;
     int AudioDownmix;
     int AudioSoftvol;
     int AudioNormalize;
@@ -720,9 +725,6 @@ void cMenuSetupSoft::Create(void)
     static const char *const audiodrift[] = {
 	"None", "PCM", "AC-3", "PCM + AC-3"
     };
-    static const char *const passthrough[] = {
-	"None", "AC-3"
-    };
     static const char *const resolution[RESOLUTIONS] = {
 	"576i", "720p", "fake 1080i", "1080i"
     };
@@ -844,9 +846,13 @@ void cMenuSetupSoft::Create(void)
 		-1000, 1000));
 	Add(new cMenuEditStraItem(tr("Audio drift correction"), &AudioDrift, 4,
 		audiodrift));
-	Add(new cMenuEditStraItem(tr("Audio pass-through"), &AudioPassthrough,
-		2, passthrough));
-	Add(new cMenuEditBoolItem(tr("Enable AC-3 (decoder) downmix"),
+	Add(new cMenuEditBoolItem(tr("Enable PCM pass-through"),
+		&AudioPassthroughPCM, trVDR("no"), trVDR("yes")));
+	Add(new cMenuEditBoolItem(tr("Enable AC-3 pass-through"),
+		&AudioPassthroughAC3, trVDR("no"), trVDR("yes")));
+	Add(new cMenuEditBoolItem(tr("Enable EAC-3 pass-through"),
+		&AudioPassthroughEAC3, trVDR("no"), trVDR("yes")));
+	Add(new cMenuEditBoolItem(tr("Enable (E)AC-3 (decoder) downmix"),
 		&AudioDownmix, trVDR("no"), trVDR("yes")));
 	Add(new cMenuEditBoolItem(tr("Volume control"), &AudioSoftvol,
 		tr("Hardware"), tr("Software")));
@@ -1031,7 +1037,9 @@ cMenuSetupSoft::cMenuSetupSoft(void)
     Audio = 0;
     AudioDelay = ConfigVideoAudioDelay;
     AudioDrift = ConfigAudioDrift;
-    AudioPassthrough = ConfigAudioPassthrough;
+    AudioPassthroughPCM = ConfigAudioPassthrough & CodecPCM;
+    AudioPassthroughAC3 = ConfigAudioPassthrough & CodecAC3;
+    AudioPassthroughEAC3 = ConfigAudioPassthrough & CodecEAC3;
     AudioDownmix = ConfigAudioDownmix;
     AudioSoftvol = ConfigAudioSoftvol;
     AudioNormalize = ConfigAudioNormalize;
@@ -1174,8 +1182,12 @@ void cMenuSetupSoft::Store(void)
     VideoSetAudioDelay(ConfigVideoAudioDelay);
     SetupStore("AudioDrift", ConfigAudioDrift = AudioDrift);
     CodecSetAudioDrift(ConfigAudioDrift);
-    SetupStore("AudioPassthrough", ConfigAudioPassthrough = AudioPassthrough);
+    ConfigAudioPassthrough = (AudioPassthroughPCM ? CodecPCM : 0)
+	| (AudioPassthroughAC3 ? CodecAC3 : 0)
+	| (AudioPassthroughEAC3 ? CodecEAC3 : 0);
+    SetupStore("AudioPassthrough", ConfigAudioPassthrough);
     CodecSetAudioPassthrough(ConfigAudioPassthrough);
+    AudioPassthroughState = 1;
     SetupStore("AudioDownmix", ConfigAudioDownmix = AudioDownmix);
     CodecSetAudioDownmix(ConfigAudioDownmix);
     SetupStore("AudioSoftvol", ConfigAudioSoftvol = AudioSoftvol);
@@ -1786,18 +1798,23 @@ static void HandleHotkey(int code)
 {
     switch (code) {
 	case 10:			// disable pass-through
-	    CodecSetAudioPassthrough(ConfigAudioPassthrough = 0);
+	    AudioPassthroughState = 0;
+	    CodecSetAudioPassthrough(0);
 	    Skins.QueueMessage(mtInfo, tr("pass-through disabled"));
 	    break;
 	case 11:			// enable pass-through
-	    CodecSetAudioPassthrough(ConfigAudioPassthrough = 1);
+	    // note: you can't enable, without configured pass-through
+	    AudioPassthroughState = 1;
+	    CodecSetAudioPassthrough(ConfigAudioPassthrough);
 	    Skins.QueueMessage(mtInfo, tr("pass-through enabled"));
 	    break;
 	case 12:			// toggle pass-through
-	    CodecSetAudioPassthrough(ConfigAudioPassthrough ^= 1);
-	    if (ConfigAudioPassthrough) {
+	    AudioPassthroughState ^= 1;
+	    if (AudioPassthroughState) {
+		CodecSetAudioPassthrough(ConfigAudioPassthrough);
 		Skins.QueueMessage(mtInfo, tr("pass-through enabled"));
 	    } else {
+		CodecSetAudioPassthrough(0);
 		Skins.QueueMessage(mtInfo, tr("pass-through disabled"));
 	    }
 	    break;
@@ -2902,6 +2919,9 @@ bool cPluginSoftHdDevice::SetupParse(const char *name, const char *value)
     }
     if (!strcasecmp(name, "AudioPassthrough")) {
 	CodecSetAudioPassthrough(ConfigAudioPassthrough = atoi(value));
+	if (ConfigAudioPassthrough) {
+	    AudioPassthroughState = 1;
+	}
 	return true;
     }
     if (!strcasecmp(name, "AudioDownmix")) {
@@ -3272,13 +3292,13 @@ cString cPluginSoftHdDevice::SVDRPCommand(const char *command,
 		    free(tmp);
 		    return "missing option argument";
 		}
-		free(ConfigAC3Device);
-		ConfigAC3Device = strdup(o);
-		AudioSetDeviceAC3(ConfigAC3Device);
+		free(ConfigPassthroughDevice);
+		ConfigPassthroughDevice = strdup(o);
+		AudioSetPassthroughDevice(ConfigPassthroughDevice);
 	    } else if (!strncmp(s, "-p", 2)) {
-		free(ConfigAC3Device);
-		ConfigAC3Device = strdup(s + 2);
-		AudioSetDeviceAC3(ConfigAC3Device);
+		free(ConfigPassthroughDevice);
+		ConfigPassthroughDevice = strdup(s + 2);
+		AudioSetPassthroughDevice(ConfigPassthroughDevice);
 
 	    } else if (*s) {
 		free(tmp);