Re: Davinci-linux-open-source Digest, Vol 28, Issue 24

manas lenka Tue, 08 Apr 2008 07:35:36 -0700

Hi,
Niclas
Do u mean that the demo codec can't play file having B-Frames..
Thanks
MANAS

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Today's Topics:

   1. RE: outArgs.displayBufs.bufs[0]  return NULL (Anderberg, Niclas)
   2. Warnings in CE 1.2 (Raghavendra B K)

----------------------------------------------------------------------

Message: 1
Date: Tue, 8 Apr 2008 15:58:08 +0200
From: "Anderberg, Niclas" 
Subject: RE: outArgs.displayBufs.bufs[0]  return NULL
To: , "davinci"

Message-ID:
 <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"

Hello,

VIDDEC_OutArgs.displayBufs.bufs[0] is supposed to be used by the codec
to give you a buffer back for display. Due to B-frames, this may not be
the same as the buffer you pass in to process (outBufs). This means that
outBufs should be considered an input parameter to the codec (here's a
buffer to fill).

A video decoder should always return the buffers for display using
VIDDEC_OutArgs.displayBufs to allow the same application code to work
for codecs which use B-frames (when display order is different from
decode order).

Regards,
Niclas

-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Behalf Of steven.zhang
Sent: Tuesday, April 08, 2008 5:55 AM
To: davinci
Subject: outArgs.displayBufs.bufs[0] return NULL

in TI decode demo, 

VIDDEC_process(hDecode, &inBufDesc, &outBufDesc, &inArgs,
                            &outArgs);

what is outArgs.displayBufs.bufs[0] used for?

I try to use codec provide by other company, it always return NULL.
However, ti codec does not.

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------------------------------

Message: 2
Date: Tue, 8 Apr 2008 19:34:59 +0530
From: "Raghavendra B K" 
Subject: Warnings in CE 1.2
To: 
Message-ID:
 <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="utf-8"

Following are the warnings that pop up when an audio component is run in CE 1.2 
environment:

      1. osalGlobal.armDspLinkConfig = osalGlobal.DEFAULT_ARMDSPLINKCONFIG;

      line has to be added in application's .cfg file.

      -- why is this required?

      2. ti.sdo.ce.osal's Global.defaultMemSegId is null

      -- should be set to "DDR".

      -- why is this required?

      3. DMAN3 NOT CONFIGURED BY THE APPLICATION!

      -- Should this be configured?

      -- Will using the default value cause any issue or slow down the build 
process?

      4. GBL.CALLUSERINITFXN = 1; and GBL.USERINITFXN = "HAL_init"; have to be 
removed from the server's tcf file.

      -- why?

      5. warning: xdc.cfg.Program: 
"/backup/mp_vps/vps/dvevm_1_2/dvevm_1_20/xdc_2_95/xdc_2_95_01/packages/xdc/runtime/package.xs",

      -- line 57: xdc.cfg.Program : assignment to Program.system is deprecated

      -- what is the meaning of this warning and how should it be resolved?

      -- observation : while building a lot of time is spent after this warning 
is displayed. 

      6. Warning: [ti.sdo.ce/package.xs] Configuring server's trace buffer size 
by writing to Server's traceBufferSize field

      -- is being deprecated; please set the server trace buffer size by 
setting the traceBufferSize field in the ti.sdo.ce.osal.Global module 

      -- (e.g.: xdc.useModule('ti.sdo.ce.osal.Global').traceBufferSize = 
262144; ) to avoid this warning 

      -- and a possible error with the future releases of Codec Engine. 

      -- how to avoid this warning?

Regards,

Raghavendra.

-----Original Message-----
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Tuesday, April 08, 2008 7:55 AM
To: [email protected]
Subject: Davinci-linux-open-source Digest, Vol 28, Issue 22

Send Davinci-linux-open-source mailing list submissions to

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Today's Topics:

   1. Re: TI's GStreamer demo (Tyler Nielsen)

   2. RE: SPI over EDMA settings (Griffis, Brad)

   3. Re: DM355 MPEG-4 Streaming (Viet Hoang)

----------------------------------------------------------------------

Message: 1

Date: Mon, 7 Apr 2008 14:21:57 -0600

From: "Tyler Nielsen" 

Subject: Re: TI's GStreamer demo

To: Vijay 

Cc: [email protected]

Message-ID:

      <[EMAIL PROTECTED]>

Content-Type: text/plain; charset="iso-8859-1"

Hello,

There are a few things that I can comment on that may be of help.  First, it

actually does pause, or at least that isn't your problem.  For most plugins

(adecoder falls into this category) pause and play are identical.  It

appears that the problem is with the way the plugin is passing along the

capabilities to the osssink.

Here are a few things you can try.  First, have you tried multiple files?

Or is it just one mp3 that causes problems?  If you can post the mp3 file

somewhere I can try playing it.  Second, there is a plugin called mp3parse.

You probably have to install it by downloading gst-plugins-ugly.  If I don't

use this, I get audio corruption when playing an mp3 stream, but it might

help with what your seeing.  You would use it like this "gst-launch-0.10

filesrc location=$1 ! mp3parse ! adecoder Codec=3 ! ossink".  The last thing

you can do is try to start debugging the issue.  If I had to guess, it looks

like gst_caps_copy is returning NULL in gst_adecoder_src_getcaps in

adecoder.c.  That might be a good place to start.  Additionally, you can try

compiling the version of gstreamer I just sent out.  Then run:

GST_DEBUG=adecoder:5 ./test_MP3.sh MP3_file.mp3

That should dump log information that may help debug.  It doesn't have much

debug about caps negotiation though.

Good luck,

Tyler

2008/4/7 Vijay :

> Hi all,

> I've noticed that the last element in the pipeline consistently fails to

> pause. Actually, I'm not sure if (a) it doesn't pause because it doesn't

> finish preroll or (b) it says it hasn't finished preroll because it doesn't

> send a pause signal! (I'm not sure which is the cause and which is the

> effect). (Also, I'm not the only one facing this problem - see the mail

> below)

> 

> Would anyone know what the problem might be? Any help would be greatly

> appreciated!

> 

> 

> 

> On Tue, Mar 25, 2008 at 4:39 PM, Vijay  wrote:

> 

> > Hi,

> > I received TI's port of gstreamer to it's DaVinci processors from

> > http://focus.ti.com/dsp/docs/dspsplash.tsp?contentId=3100

> >

> > I've tried to run the example (scripts) provided by TI and I've faced

> > what seem to be two separate issues.

> > I've copied the stdout log below:

> >

> > 
:/opt/system_files_gstreamer# ./test_MP3.sh MP3_file.mp3

> > Setting pipeline to PAUSED ...

> > Pipeline is PREROLLING ...

> > (gst-launch-0.10:1204): GStreamer-WARNING **: pad adecoder0:src returned

> > caps which are not a real subset of its template caps

> > (gst-launch-0.10:1204): GStreamer-CRITICAL **: gst_caps_get_structure:

> > assertion `index < caps->structs->len' failed

> > (gst-launch-0.10:1204): GStreamer-CRITICAL **: gst_structure_get_int:

> > assertion `structure != NULL' failed

> >

> > (program hangs here)

> >

> > The two issues:

> > (a) The last element in the gstreamer pipeline does not reply with a

> > message to the app saying it has paused (This is ascertained with debug

> > prints that I inserted in gst-launch.c. It's possible that for some reason,

> > the element does not pause. I've faced this issue with pipelines which have

> > no decode/render elements as well.) *** I'm not sure if (a) it doesn't pause

> > because it doesn't finish preroll or (b) it says it hasn't finished preroll

> > because it doesn't send a pause signal! ***

> > (b) The critical errors printed (seen above. I guess these are caused

> > because of TI's mp3 decoder element plugin.) Could the adecoder capabilities

> > be incompatible? Seems unlikely.

> >

> >

> > Would anyone know what the issue might be? Has anyone seen a similar

> > issue with TI's gstreamer examples?

> > I'd greatly appreciate any help.

> >

> >

> > Vijay

> >

> 

> 

> 

> 

> 

> 

> 

> 

> 

> ------------------------------

> 

> 

> 

> Message: 3

> 

> Date: Mon, 7 Apr 2008 13:53:46 +0800

> 

> From: "Hans Kuo" 

> 

> Subject: Question about GStreamer Demo

> 

> To: 

> 

> Message-ID: <[EMAIL PROTECTED]>

> 

> Content-Type: text/plain; charset="big5"

> 

> 

> 

> Hi,   I have the same problem when trying to run Run the Demo on the

> Target  On the Davinci Board :1.        Copy the davincieffect.aac ( from

> dvevm_1_20/data ) to the system_fileS_gstreamer directory2.        Run the

> ./start_demo.sh file3.        Run the following shell to play an AAC

> file#./test_AAC.sh davincieffect.aac The following message is displayed

> :=======gdbcoder:gst_gdecoder_get_codec_type

> Beginegdbcoder:gst_gdecoder_get_codec_type

> Endgdbcoder:gst_gdecoder_get_codec_type

> Beginegdbcoder:gst_gdecoder_get_codec_type End Setting pipeline to

> PAUSEDPipeline is PREROLLING Â¡KÂ¡K. ( It has been staying in this state all

> the time )======= Question : Is it correct ?  I am unable to be heard any

> sound by the loudspeakerAny idea how to solve this problem? Regards,clara

> Hsieh.

> 

> _______________________________________________

> Davinci-linux-open-source mailing list

> [email protected]

> http://linux.davincidsp.com/mailman/listinfo/davinci-linux-open-source

> 

> 

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Message: 2

Date: Mon, 7 Apr 2008 21:03:51 -0500

From: "Griffis, Brad" 

Subject: RE: SPI over EDMA settings

To: "Winston Ma" ,

Message-ID:

      <[EMAIL PROTECTED]>

Content-Type: text/plain; charset="utf-8"

If the SPI interface has a 16-bit interface then you should do an A-sync 
transfer with ACNT=2.  It should be A-sync and not AB-sync because the SPI 
interface is only ready to receive one element when it gives the 
Ã¢â¬ÅreadyÃ¢â¬Â signal to the EDMA.  If for example the SPI had a FIFO on 
it then you would be able to do AB-sync with BCNT equal to the FIFO depth, but 
thatÃ¢â¬â¢s not the case.

The BIDX is the src/dst pointer post-increment used after transferring ACNT 
bytes (i.e. an array).  The CIDX is the src/dst pointer post increment used 
after transferring BCNT arrays (i.e. a frame).  Note that CIDX is measured 
differently depending on whether you are doing A-sync or AB-sync transfers.  If 
CCNT=1 (which will probably be the case for you) then the CIDX is a 
donÃ¢â¬â¢t care.  Also, remember that the total number of bytes transferred 
is ACNT*BCNT*CCNT so donÃ¢â¬â¢t make CCNT=0 or else nothing at all will 
transfer!

Brad

________________________________

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Winston Ma

Sent: Sunday, April 06, 2008 10:13 PM

To: [email protected]

Subject: SPI over EDMA settings

Dear all,

I would like to know what types of transfer I should use? A-Synchronized or 
AB-Synchronized? also what is the BIDX and CIDX settings?

Thanks a lot

________________________________

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Message: 3

Date: Tue, 8 Apr 2008 03:23:53 +0100 (BST)

From: Viet Hoang 

Subject: Re: DM355 MPEG-4 Streaming

To: Albert Burbea 

Cc: [email protected]

Message-ID: <[EMAIL PROTECTED]>

Content-Type: text/plain; charset="iso-8859-1"

Hi Albert,

    I think this contains all you need

    http://www.live555.com/mediaServer/

Regards,

Viet Hoang.

Albert Burbea  wrote: Thank you so much

 if you can add some explanation about how to use it

 Albert 

 On 4/7/08, Viet Hoang  wrote: Sorry, I forgot to attach the file.

Viet Hoang.  

Viet Hoang  wrote:

   Hi,

   I cannot send all the live555 package compiled on my board. Attached is the 
executable file live555MediaServer, put it any where on your board and run it. 
Let me know if there's any problem or you make any progress :).

 Regards,

Viet Hoang

Albert Burbea  wrote:   all the cfg

 Albert 

 On 4/7/08, Viet Hoang  wrote:  Hi Albert,

   I do not know exactly what you meant for "your setup". You meant 
live555MediaServer, or all the configuration used to test mpeg4 streamed from 
recorded file?

 Regards,

  Viet Hoang.  

Albert Burbea  wrote:   Hi Viet

 it would be nice to have a copy (or some explanation) of your setup

 Albert 

 On Fri, Apr 4, 2008 at 6:43 AM, Viet Hoang  wrote:

  Hi all,

    Prateek, did you really try this (using live555MediaServer)?

Here are what I tried and failed:

      1) Record a short .mpeg4 clip using DM355, change the file's extension to 
.m4e and copy the file to live555MediaServer's directory, run 
live555MediaServer and then on a Windows PC use VLC to play that file using 
RTSP: FAILED - I didn't see any data received at VLC site.

     2) Copy file davincieffect_ntsc.mpeg4 from /opt/dvsdk/data/videos/ to 
live555MediaServer's directory, change the extension to m4e, run 
live555MediaServer and use VLC to play that file: FAILED.

    3) Play back those 2 files using ffplay (from ffmpeg) on Fedora 5: OK.

        Play back those 2 files using VLC 0.8.6f on Windows: FAILED.

    4) Try 1 and 2 using files from DM6446, work well.

    I think 1 & 2 are due to the one-time sent VOL header as Antonio said. Is 
this due to the Beta released version of DM355EVM and mpeg4 codec? Do you have 
any idea how to fix this?

 Regards,

Viet Hoang.   

"Bansal, Prateek" 
 wrote:    Albert, Anotonio,

 I will suggest looking at 

 http://www.live555.com/mediaServer/

 Open source C++ libraries under LGPL to support RTSP streaming. There are test 
programs that are available with this library that can be easily cross-compiled 
on Montavista Linux platform. You can use the test program to stream MPEG4 
elementary stream file contents generated by DM355 encoder and can be played 
back using any PC client like VLC, Mplayer to playback the stream contents.

 Regards,

 Prateek

---------------------------------

 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Albert Burbea

 Sent: Wednesday, April 02, 2008 5:58 AM

To: AntÃ³nio Silva

Cc: [email protected]

 Subject: Re: DM355 MPEG-4 Streaming

  Thanks

  but I would not like to go and build a new RTP/RTSP etc stack on my own... I 
an sure there are readymade stacks like these on the net. I have heard of 
live555 and others.

  Any ideas ?

  Albert 

  On 4/1/08, AntÃ³nio Silva  wrote: 

  The most helpful resources are RFCs from IETF. I can point you out the most 
important rfc's about the subject you pretend.

     RTP: rfc1889

      RTSP: rfc2326

     SDP: rfc2327 (this is useful for rtsp or multicast streams)

     RTCP: rfc3550

Then, depending on which standard are you working at (MPEG-4 SP, AVC) you have 
to verify the corresponding RFC to see how to packetizes the encoded frame.

Regards,

A.Silva

  On Mon, Mar 31, 2008 at 11:03 PM, Albert Burbea  wrote:

   hi all

  can somebody explain a little about video streaming in general, or give us 
some pointers?

  Especially on rtp, rtsp, and different kind of media

  thanks all

    On 3/31/08, AntÃ³nio Silva  wrote: 

      Hi all,

  I am trying to perform a test which consists in perform an MPEG-4 Encoder & 
Streaming application using the board DM355.

  I already worked with an H.264 Encoder & Streaming application, and with 
H.264 we have sps and pps periodically transmited which allows a client to 
receive the stream even is the sender is not at the beggining of the 
transmission.

  In MPEG-4 we only have the MPEG-4 header (I think is VOL header) at the 
beggining an then we just have VOP headers. The receiver cannot receive the 
stream if he dows not get the VOL header.

  My question is, how can we do this using the MPEG-4 Encoder at DM355?  Does 
anybody knows the size of the VOL header generated by the encoder at DM355??

  Thanks in advance,

  A.Silva

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Tel/Fax + 972-3-7526016

Mobile: +972-52-3541842 

-- 

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Mobile: +972-52-3541842 

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