2520 replies to this topic

[christophecvr]

Yes great much better now its like this :

playing 4097:0:0:0:0:0:0:0:0:0:/media/VIDAUD/Robocop-1080p.mkv
<  1473.428605> [eDVBCAService] free slot 0 demux 0 for service 1:0:1:10E0:418:1:C00000:0:0:0:
<  1473.428899> [eDVBCAService] free service 1:0:1:10E0:418:1:C00000:0:0:0:
<  1473.468416> decoder state: play, vpid=-1, apid=-1
<  1473.468639> DEMUX_STOP - pcr - ok
<  1473.469105> DEMUX_STOP - video - ok
<  1473.470355> VIDEO_STOP - ok
<  1473.470800> [eDVBAUDIO] AUDIO_STOP - audio0
<  1473.470961> [eDVBAUDIO] AUDIO_STOP - audio0ok
<  1473.491130> [eDVBAUDIO] DEMUX_STOP - audio0 - ok
<  1473.491399> [eDVBAUDIO] AUDIO_CONTINUE - audio0 ok
<  1473.491759> DEMUX_STOP - ttx - ok
<  1473.513418> [eDVBResourceManager] start release channel timer
<  1473.517107> eServiceMP3::construct!
<  1473.517602> eServiceMP3::playbin uri=file:///media/VIDAUD/Robocop-1080p.mkv
<  1473.616255> eServiceMP3::starting pipeline
<  1473.753861> [eDVBDB] getBouquet failed.. no path given!
resolved to PLAY
<  1473.781451> gst_element_query_position failed in getPlayPosition
new service started! trying to download cuts!
download failed, no cuesheet interface
RemovePopup, id = ZapError
<  1473.812806> gst_element_query_position failed in getPlayPosition
<  1473.821319> gst_element_query_position failed in getPlayPosition
<  1473.824508> gst_element_query_position failed in getPlayPosition
<  1473.832988> gst_element_query_position failed in getPlayPosition
<  1473.836263> gst_element_query_position failed in getPlayPosition
<  1473.913425> gst_element_query_position failed in getPlayPosition
<  1473.976759> gst_element_query_position failed in getPlayPosition
<  1473.992332> gst_element_query_position failed in getPlayPosition
<  1473.993972> gst_element_query_position failed in getPlayPosition
<  1474.012734> gst_element_query_position failed in getPlayPosition
<  1474.015647> gst_element_query_position failed in getPlayPosition
<  1474.050918> gst_element_query_position failed in getPlayPosition
<  1474.053681> gst_element_query_position failed in getPlayPosition
<  1474.057603> gst_element_query_position failed in getPlayPosition
<  1474.059833> gst_element_query_position failed in getPlayPosition
<  1474.065843> gst_element_query_position failed in getPlayPosition
<  1474.069292> gst_element_query_position failed in getPlayPosition
<  1474.088825> eServiceMP3::state transition NULL -> READY
<  1474.521139> gst_element_query_position failed in getPlayPosition
<  1474.940493> gst_element_query_position failed in getPlayPosition
<  1474.964553> gst_element_query_position failed in getPlayPosition
<  1474.967901> gst_element_query_position failed in getPlayPosition
<  1474.971151> gst_element_query_position failed in getPlayPosition
<  1474.973330> gst_element_query_position failed in getPlayPosition
<  1474.977983> gst_element_query_position failed in getPlayPosition
<  1474.981818> gst_element_query_position failed in getPlayPosition
0:03:34.498621883   744  0x1a2d2f0 INFO              dtsdownmix gstdtsdownmix.c:827:gst_dtsdec_change_state: STATE CHANGE OCCURED TYPE = 10
0:03:34.498700994   744  0x1a2d120 INFO              dtsdownmix gstdtsdownmix.c:827:gst_dtsdec_change_state: STATE CHANGE OCCURED TYPE = 10
0:03:34.501756328   744  0x1a2d2f0 INFO              dtsdownmix gstdtsdownmix.c:832:gst_dtsdec_change_state: STATE CHANGE OCCURED NULL_TO_READY = 10
0:03:34.501833550   744  0x1a2d120 INFO              dtsdownmix gstdtsdownmix.c:832:gst_dtsdec_change_state: STATE CHANGE OCCURED NULL_TO_READY = 10
0:03:34.506398587   744  0x1a2d2f0 INFO              dtsdownmix gstdtsdownmix.c:827:gst_dtsdec_change_state: STATE CHANGE OCCURED TYPE = 19
0:03:34.506827883   744  0x1a2d2f0 INFO              dtsdownmix gstdtsdownmix.c:844:gst_dtsdec_change_state: STATE CHANGE OCCURED READY_TO_PAUSED = 19
<  1475.018368> gst_element_query_position failed in getPlayPosition
0:03:34.511607809   744  0x1a2d120 INFO              dtsdownmix gstdtsdownmix.c:827:gst_dtsdec_change_state: STATE CHANGE OCCURED TYPE = 19
0:03:34.511960920   744  0x1a2d120 INFO              dtsdownmix gstdtsdownmix.c:844:gst_dtsdec_change_state: STATE CHANGE OCCURED READY_TO_PAUSED = 19
<  1475.025075> gst_element_query_position failed in getPlayPosition
0:03:34.526336994   744  0x1a2d120 INFO              dtsdownmix gstdtsdownmix.c:755:gst_dtsdec_sink_event: A SINK EVENT name stream-start just occured
0:03:34.526386735   744  0x1a2d2f0 INFO              dtsdownmix gstdtsdownmix.c:755:gst_dtsdec_sink_event: A SINK EVENT name stream-start just occured
0:03:34.527454179   744  0x1a2d120 INFO              dtsdownmix gstdtsdownmix.c:755:gst_dtsdec_sink_event: A SINK EVENT name caps just occured
0:03:34.527526772   744  0x1a2d2f0 INFO              dtsdownmix gstdtsdownmix.c:755:gst_dtsdec_sink_event: A SINK EVENT name caps just occured
0:03:34.533017587   744  0x1a2d120 INFO              dtsdownmix gstdtsdownmix.c:755:gst_dtsdec_sink_event: A SINK EVENT name toc just occured
0:03:34.533881661   744  0x1a2d120 INFO              dtsdownmix gstdtsdownmix.c:755:gst_dtsdec_sink_event: A SINK EVENT name segment just occured
0:03:34.534446216   744  0x1a2d120 INFO              dtsdownmix gstdtsdownmix.c:755:gst_dtsdec_sink_event: A SINK EVENT name tag just occured
0:03:34.536587216   744  0x1a2d120 INFO              dtsdownmix gstdtsdownmix.c:755:gst_dtsdec_sink_event: A SINK EVENT name tag just occured
<  1475.059994> gst_element_query_position failed in getPlayPosition
0:03:34.549053216   744  0x1a2d2f0 INFO              dtsdownmix gstdtsdownmix.c:790:gst_dtsdec_src_event: A SRC EVENT NAME = <reconfigure> JUST OCCORED
<  1475.063255> gst_element_query_position failed in getPlayPosition
0:03:34.553596809   744  0x1a2d120 INFO              dtsdownmix gstdtsdownmix.c:755:gst_dtsdec_sink_event: A SINK EVENT name tag just occured
0:03:34.555105587   744  0x1a2d2f0 INFO              dtsdownmix gstdtsdownmix.c:790:gst_dtsdec_src_event: A SRC EVENT NAME = <reconfigure> JUST OCCORED
0:03:34.558135698   744  0x1a2d2f0 INFO              dtsdownmix gstdtsdownmix.c:790:gst_dtsdec_src_event: A SRC EVENT NAME = <reconfigure> JUST OCCORED
0:03:34.561183920   744  0x1a2d2f0 INFO              dtsdownmix gstdtsdownmix.c:790:gst_dtsdec_src_event: A SRC EVENT NAME = <reconfigure> JUST OCCORED
<  1475.072855> gst_element_query_position failed in getPlayPosition
<  1475.075519> gst_element_query_position failed in getPlayPosition
0:03:34.570224698   744  0x1a2d120 INFO              dtsdownmix gstdtsdownmix.c:390:gst_dtsdec_renegotiate:<dtsdec1> dtsdec renegotiate, channels=2, rate=48000
Stream with high frequencies VQ coding
0:03:34.578843957   744  0x1a2d2f0 INFO              dtsdownmix gstdtsdownmix.c:790:gst_dtsdec_src_event: A SRC EVENT NAME = <reconfigure> JUST OCCORED
0:03:34.580142957   744  0x1a2d2f0 INFO              dtsdownmix gstdtsdownmix.c:790:gst_dtsdec_src_event: A SRC EVENT NAME = <reconfigure> JUST OCCORED
<  1475.097539> gst_element_query_position failed in getPlayPosition
<  1475.107120> gst_element_query_position failed in getPlayPosition
0:03:34.793506216   744  0x1a2d2f0 INFO              dtsdownmix gstdtsdownmix.c:790:gst_dtsdec_src_event: A SRC EVENT NAME = <reconfigure> JUST OCCORED
0:03:34.794235476   744  0x1a2d2f0 INFO              dtsdownmix gstdtsdownmix.c:790:gst_dtsdec_src_event: A SRC EVENT NAME = <reconfigure> JUST OCCORED
0:03:34.795139105   744  0x1a2d2f0 INFO              dtsdownmix gstdtsdownmix.c:790:gst_dtsdec_src_event: A SRC EVENT NAME = <reconfigure> JUST OCCORED
0:03:34.795585624   744  0x1a2d2f0 INFO              dtsdownmix gstdtsdownmix.c:790:gst_dtsdec_src_event: A SRC EVENT NAME = <reconfigure> JUST OCCORED
0:03:34.930248179   744  0x1a2d2f0 INFO              dtsdownmix gstdtsdownmix.c:755:gst_dtsdec_sink_event: A SINK EVENT name toc just occured
<  1475.468952> [DVBCAHandler] no more services
0:03:34.992581549   744  0x1a2d2f0 INFO              dtsdownmix gstdtsdownmix.c:755:gst_dtsdec_sink_event: A SINK EVENT name segment just occured
0:03:35.040390735   744  0x1a2d2f0 INFO              dtsdownmix gstdtsdownmix.c:755:gst_dtsdec_sink_event: A SINK EVENT name tag just occured
0:03:35.107039364   744  0x1a2d2f0 INFO              dtsdownmix gstdtsdownmix.c:755:gst_dtsdec_sink_event: A SINK EVENT name tag just occured
0:03:35.154357735   744  0x1a2d2f0 INFO              dtsdownmix gstdtsdownmix.c:755:gst_dtsdec_sink_event: A SINK EVENT name tag just occured
0:03:35.201507809   744  0x1a2d2f0 INFO              dtsdownmix gstdtsdownmix.c:390:gst_dtsdec_renegotiate:<dtsdec0> dtsdec renegotiate, channels=2, rate=48000
Stream with high frequencies VQ coding
0:03:35.247151142   744 0x753044c0 INFO              dtsdownmix gstdtsdownmix.c:790:gst_dtsdec_src_event: A SRC EVENT NAME = <reconfigure> JUST OCCORED
0:03:35.247992809   744 0x753044c0 INFO              dtsdownmix gstdtsdownmix.c:790:gst_dtsdec_src_event: A SRC EVENT NAME = <reconfigure> JUST OCCORED
0:03:35.249983920   744 0x753044c0 INFO              dtsdownmix gstdtsdownmix.c:790:gst_dtsdec_src_event: A SRC EVENT NAME = <reconfigure> JUST OCCORED
0:03:35.250619401   744 0x753044c0 INFO              dtsdownmix gstdtsdownmix.c:790:gst_dtsdec_src_event: A SRC EVENT NAME = <reconfigure> JUST OCCORED
0:03:35.330280550   744  0x1a2d2f0 INFO              dtsdownmix gstdtsdownmix.c:755:gst_dtsdec_sink_event: A SINK EVENT name tag just occured
0:03:35.374446846   744  0x1a2d120 INFO              dtsdownmix gstdtsdownmix.c:755:gst_dtsdec_sink_event: A SINK EVENT name tag just occured
0:03:35.396587846   744 0x74c40030 INFO              dtsdownmix gstdtsdownmix.c:790:gst_dtsdec_src_event: A SRC EVENT NAME = <latency> JUST OCCORED
0:03:35.397364290   744 0x74c40030 INFO              dtsdownmix gstdtsdownmix.c:790:gst_dtsdec_src_event: A SRC EVENT NAME = <latency> JUST OCCORED
0:03:35.440713328   744 0x74c40030 INFO              dtsdownmix gstdtsdownmix.c:827:gst_dtsdec_change_state: STATE CHANGE OCCURED TYPE = 28
0:03:35.440985290   744 0x74c40030 INFO              dtsdownmix gstdtsdownmix.c:851:gst_dtsdec_change_state: STATE CHANGE OCCURED PAUSED_TO_PLAYING = 28
0:03:35.441711957   744 0x74c40030 INFO              dtsdownmix gstdtsdownmix.c:827:gst_dtsdec_change_state: STATE CHANGE OCCURED TYPE = 28
0:03:35.448228031   744 0x74c40030 INFO              dtsdownmix gstdtsdownmix.c:851:gst_dtsdec_change_state: STATE CHANGE OCCURED PAUSED_TO_PLAYING = 28
<  1475.962331> eServiceMP3::state transition READY -> PAUSED
<  1475.968504> eServiceMP3::subsink properties set!
<  1475.974356> eServiceMP3::async-done - 1 video, 2 audio, 1 subtitle
<  1475.974861> AUDIO STRUCT=audio/x-raw
<  1475.975896> eServiceMP3::audio stream=0 codec=DTS language=en
<  1475.991565> AUDIO STRUCT=audio/x-raw
<  1475.992832> eServiceMP3::audio stream=1 codec=DTS language=de
<  1475.994603> eServiceMP3::subtitle stream=0 language=de codec=(null)
<  1475.995118> getSubtitleType::subtitle probe caps type=text/x-raw
<  1476.082309> eServiceMP3::switched to subtitle stream 0
<  1476.411394> eServiceMP3::state transition PAUSED -> PLAYING
<  1476.513892> [eDVBResourceManager] release cached channel (timer timeout)
<  1476.514248> [eDVBLocalTimerHandler] remove channel 0x1a04b98
<  1476.514431> [eEPGCache] remove channel 0x1a04b98
<  1476.518761> [eDVBResourceManager] stop release channel timer
<  1479.530977> close frontend 0

[christophecvr]

What would be nice is that enigma logging is improved by showing the time for each line. Then its much better to trace where it hangs to long.

[betacentauri]

I thought this was already added some weeks ago:
http://sourceforge.n...08eb8e3068b624/

[christophecvr]

Gues it's not merged with  gst-1 branch yet.

[christophecvr]

Oepss it has nothing to do with the merge it' enigma2 i do use the last one. I gues  they forgot to format time to a readable format.

[malakudi]

It is quite readable, means seconds.useconds from system start.

[athoik]

Subject: [PATCH] Make MAX PCM rate configurable with default 48000

Some boxes support more than 48 kHz PCM rates.
For 96 kHz we need to add --with-maxpcmrate=960000.

 0001-Make-MAX-PCM-rate-configurable-with-default-48000.patch   2.29KB   4 downloads

Apply above one on master dvbmediasink branch.


 0001-Make-MAX-PCM-rate-configurable-with-default-48000.gst-1.patch   2.69KB   2 downloads

Apply above one on gst-1 dvbmediasink branch.

[Erik Slagter]

Applied.

[christophecvr]

Ter info I also just added a change about this max rate.

 

By me it is default MAX. That is the evolution in future.

 

Also the printf is cleaned out replaced by GST_INFO or GST_INFO_OBJECT(if relevant) . OR GST_DEBUG_xxx

 

Changes pushed to master adapted readme. To limit pcm to 48000 just use --with-max-pcmrate-48K . The longer the more in future new boxes will be able to use the real audio sample rate.

And we do not need to reduce sound quality.

 

https://github.com/c.../commits/master

 

p.s. I think about adapting the configure.ac . To have each machine type into it. The type will be the same as the MACHINE=xxxxxx  when You build.

That would made it better to have a dvbmediasink optimized for the specific machine. Whitout unneeded code into the final compiled so files.

Then such extra parameters like --with-max-pcmrate-48K are not needed anymore.

 

Like for et I gues the 10K does support the real audio sample rate but lower versions not and so on ....

Edited by christophecvr, 8 April 2015 - 17:15.

[athoik]

Subject: [PATCH] Remove framed from LPCM caps

For lpcm there is no parser (eg lpcmparse), so asking for framed makes no sense.
Other codecs (audio/x-ac3, audio/x-eac3, audio/x-dts, audio/x-private1-dts) with framed
we have ac3parse and dcaparse plugged.

Thanks @mx3L and @betacentauri for testing

diff --git a/gstdvbaudiosink.c b/gstdvbaudiosink.c
index 526e25d..2b87caa 100644
--- a/gstdvbaudiosink.c
+++ b/gstdvbaudiosink.c
@@ -125,8 +125,7 @@ static guint gst_dvbaudiosink_signals[LAST_SIGNAL] = { 0 };
 		"framed =(boolean) true; "
 
 #define LPCMCAPS \
-		"audio/x-private1-lpcm, " \
-		"framed =(boolean) true; "
+		"audio/x-private1-lpcm; "
 
 #define DTSCAPS \
 		"audio/x-dts, " \
-- 

Please apply on gst-1.0 dvbmediasink branch.

 0001-Remove-framed-from-LPCM-caps.patch   861bytes   1 downloads

[athoik]

By me it is default MAX.

The default MAX it's not a good choise, there are no drivers that support MAX. Currently they support specific rates.

Eg. You can try 24bit/192kHZ flac on your VU+ and tell us if VU+ can play such rate. If not you need to add the maximum supported (eg 96 kHz)

PS. I sure that --with-maxpcmrate=MAX is also a valid option

Edited by athoik, 8 April 2015 - 19:08.

[Erik Slagter]

Subject: [PATCH] Remove framed from LPCM caps

For lpcm there is no parser (eg lpcmparse), so asking for framed makes no sense.
Other codecs (audio/x-ac3, audio/x-eac3, audio/x-dts, audio/x-private1-dts) with framed
we have ac3parse and dcaparse plugged.

Thanks @mx3L and @betacentauri for testing

diff --git a/gstdvbaudiosink.c b/gstdvbaudiosink.c
index 526e25d..2b87caa 100644
--- a/gstdvbaudiosink.c
+++ b/gstdvbaudiosink.c
@@ -125,8 +125,7 @@ static guint gst_dvbaudiosink_signals[LAST_SIGNAL] = { 0 };
 		"framed =(boolean) true; "
 
 #define LPCMCAPS \
-		"audio/x-private1-lpcm, " \
-		"framed =(boolean) true; "
+		"audio/x-private1-lpcm; "
 
 #define DTSCAPS \
 		"audio/x-dts, " \
-- 

Please apply on gst-1.0 dvbmediasink branch.

0001-Remove-framed-from-LPCM-caps.patch

 

Applied.

[Erik Slagter]

PS. I sure that --with-maxpcmrate=MAX is also a valid option

That sounds reasonable

[christophecvr]

Vuduo2 with  flac 24 bit free lossless 192kHz . pcm set to MAX. Plays ok sound perfect.

 

Ouput with gst-launch-1.0 :

root@vuduo2:/media/VIDAUD# gst-launch-1.0 playbin uri=file:///media/VIDAUD/test192.flac -v
Setting pipeline to PAUSED ...
/GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: ring-buffer-max-size = 0
/GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: buffer-size = -1
/GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: buffer-duration = -1
/GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: use-buffering = false
/GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: download = false
/GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: uri = file:///media/VIDAUD/test192.flac
/GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: connection-speed = 0
/GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: source = "\(GstFileSrc\)\ source"
Pipeline is PREROLLING ...
/GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0/GstDecodeBin:decodebin0/GstTypeFindElement:typefind.GstPad:src: caps = audio/x-flac
/GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0/GstDecodeBin:decodebin0/GstTypeFindElement:typefind.GstPad:src: caps = "NULL"
/GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0/GstDecodeBin:decodebin0/GstFlacParse:flacparse0.GstPad:src: caps = "audio/x-flac\,\ channels\=\(int\)2\,\ framed\=\(boolean\)true\,\ rate\=\(int\)192000\,\ streamheader\=\(buffer\)\<\


decoding .....

output :

/GstPlayBin:playbin0/GstPlaySink:playsink/GstBin:abin/GstPlaySinkAudioConvert:aconv.GstGhostPad:sink: caps = "audio/x-raw\,\ format\=\(string\)S24_32LE\,\ layout\=\(string\)interleaved\,\ rate\=\(int\)192000\,\ channels\=\(int\)2\,\ channel-mask\=\(bitmask\)0x0000000000000003"
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
/GstPlayBin:playbin0/GstInputSelector:inputselector0.GstSelectorPad:sink_0: tags = "taglist\,\ audio-codec\=\(string\)\"Free\\\ Lossless\\\ Audio\\\ Codec\\\ \\\(FLAC\\\)\"\,\ minimum-bitrate\=\(uint\)432750\,\ bitrate\=\(uint\)456474\,\ maximum-bitrate\=\(uint\)1450875\;"
New clock: GstSystemClock
/GstPlayBin:playbin0/GstInputSelector:inputselector0.GstSelectorPad:sink_0: tags = "taglist\,\ audio-codec\=\(string\)\"Free\\\ Lossless\\\ Audio\\\ Codec\\\ \\\(FLAC\\\)\"\,\ minimum-bitrate\=\(uint\)432750\,\ bitrate\=\(uint\)510735\,\ maximum-bitrate\=\(uint\)3169500\;"

Seems that a limit is really only for the older hardware. Modern hardware and drivers do it very well with MAX . (p.s. this well with gst-1.0 about the gst-0.10 I don't)

[malakudi]

You really do understand that 96 or 192khz doesn't sound any better from 48khz, don't you? Especially from an STB.

Whoever believes he is hearing difference, I dare you to do a double blind test.

[athoik]

But having audioconvert changing to 48kHz other rates is not required if drivers can do that.

[malakudi]

It is not reasonable to try playback flac 192khz in the first place, that is what I meant. AC3/DTS should playback fine with 48khz and HD audio is just a way to push sales.

[Rob van der Does]

This is not about sales; it's about allowing the box to do what the hardware is actually capable of (without the need to convert).

[christophecvr]

@malakudi

 

Better check and learn what is a sound wave, and what is a digitalised sound wave.

 

1) First the quality off digital sound is always less then the original wave. Just cause you integrate . The higher the sample rate the less quality loss.

2) A wave (original) digitalised with a sample rate off 192000. Is a tremendous difference then with 48000. And You hear the difference clearly. (with classic music very )

 

If an stb only can handle 48000 (due to processor limitations), logic that it never will give a better sound then 48000. And a good driver will then apply sample rate max for the specific hardware.

Modern stb's in the higher quality range and fast processors like vuduo2 will not have any problem with this higher sample rates. If now the driver self downgrade the sample rate or not ? ?

That driver source is closed. But one thing is shure You hear a difference playing a (high quality 192kHz flac on vuduo2 or on dm8000) This for a real test sound specially made for that. Classic music off course.

 

For techno or so You even could play it at 24kHz never you will hear any difference that's trough.

 

And it's evident that a quality amplifier behind is important (and I do have it here over the end analoge fase). Means also good speakers and so on .....

 

Off course some persons never will hear the difference in music as they can't feel the subtility in music. That's another point.

[malakudi]

@christophecvr: You should read about Nyquist Shannon sampling theorem. 48 KHz is more than enough for the frequencies the human ear can perceive.

 

Your (1) and (2) stamements are false.

 

edit: and as I said again, you can try to verify your claims with a double-blind test. The results will be amusing.

Edited by malakudi, 9 April 2015 - 07:17.