gmane.comp.multimedia.ogg.vorbis.devel

[PATCH] vcut - Fix cut failure with discarded output

Michael Gold — 2012-05-19T16:04:58

Hi all,

Here's a patch to fix a spurious failure in vcut.  Please apply.

If the user wants to drop the output (i.e. the filename is given as ".")
and the cutpoint is found before open_output_file is called, drop_output
will not have been set, and vcut will fail with the error "Can't produce
a file starting and ending between sample positions [...]".  But when no
output file is desired, we can safely ignore this condition.

This changes the code to call the new function will_drop_output, instead
of reading s->drop_output.  This function checks the filename as well as
s->drop_output.

Memory Leak in vorbis_info_clear()

William Ahern — 2012-05-14T19:10:14

I'm having trouble tracking down why it leaks, but below is an example
program which shows--using valgrind--that vorbis_info_clear() leaks memory
if called before vorbis_dsp_clear(), but not if called after
vorbis_dsp_clear(). Just compile and run under valgrind, using the -l switch
to the example program to trigger a leak. Tested under OS X 10.7 and Ubuntu
12.04.

This may be by design, or just, "don't do that". If it is the Wiki page
should probably be edited to mention this. I tried, but it didn't look like
I could make the edit.

My solution is to clear in reverse order of initialization, though I've
rearranged the order of things in the example program to highlight the
immediate issue.

/* compile: cc -Wall -o vbs vbs.c -lvorbisenc -lvorbis */
/* noleak:  valgrind --leak-check=yes ./vbs */
/* leak:    valgrind --leak-check=yes ./vbs -l */
#include <unistd.h> /* getopt(3) */

#include <vorbis/codec.h>
#include <vorbis/vorbisenc.h>


static struct vorbis_info info;
static struct vorbis_dsp_state dsp;
stat

Patch for statically allocated tremor

戴明仁 (mingjen_tai — 2012-04-24T03:39:01

Hi all,

    I checked the mailing list of tremor and found that there was a patch for statically memory allocation. But the link(http://x256.org/~hb/Tremor.patch) seems to be dead since the site was down. Can I get that patch elsewhere? Thanks for your help.

 

 

Best regards,

Jen

 

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command line perl tool to create ogg/Vorbis picturecomments

Ian Malone — 2012-04-17T00:56:00

It turns out I'm crazy...

https://github.com/imalone/oggflacblock
Or direct from (n.b. BSD license):
https://raw.github.com/imalone/oggflacblock/master/oggflacblock.pl

Now handles png and jpeg, the only notable dependency is the Perl
Image::ExifTool module. (Which, I've noticed, can also read Vorbis
comments)

Includes usage info with -h or -help, but interesting things:
./oggflacblock.pl -i sample.png -o sample.tag -write tag -desc "Some
description" -type FileIconSmall

-write tag : produces a file with the "METADATA_BLOCK_PICTURE=" start,
so you can plug it straight into vorbiscomment. This is the default,
so you can do this:
vorbiscomment startfile.ogg newfile.ogg -a -c sample.tag
the other options are block64 (just the block as before) and blockraw
(don't 64bit encode)

-type, can be numeric, or I've come up with some short-names for the
different entries that you can specify instead.

FileIconSmall is restricted to 32x32 png, this is enforced.

Shortcomings:
* Doesn't do the "->" mode.
* Doesn't pro

ffmpeg adaptor for ogg123

Gary Benson — 2012-03-23T21:26:27

Hi all,

I'm not sure if this is interesting to you guys (I'm also not 100%
convinced this is the correct list) but I hacked up an adaptor for ogg123
to allow it to use ffmpeg as a decoder.  It currently only recognises WMA
files (I wanted to be able to play my wife's music from my Linux music
server) but it would be trivial to extend to any other formats ffmpeg
supports. Anyway, if you're interested let me know and I'll put a patch
together for you.

Cheers,
Gary

--
http://inauspicious.org/
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Need for help about using vorbis in embedded system

Mashal al-shboul — 2012-02-28T23:47:32



Hi All,
   I am a new member to the vorbis-dev mailing list. i hope that u receive the help that i've been searching for.
I need to compress audio samples captured by wireless sensor node (16-bit PCM at 8Khz). can i use vorbis i such an embedded system environment that has the following HW/SW specifications:

-416 MHz Microprocessor(ARM architecture, Intel Xscale family)

-32 MB RAM
Beside that,i can use C language for developmnet.

Do these specifications work for compressing the captured audio samples so that i get them in smaller size for transmitting them wirelessly?

  Any Help or Advice Is Very Appreciated.

Regards.
Mash'al
Jordan Univeristy Of science and Technology
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New repo and moving tremor discussion here

Ralph Giles — 2012-02-10T19:22:37

All,

I've consolidated our various svn branches of the fixed-point vorbis
decoder (known as tremor or tremolo) into a single git repository,
hoping that it facilitates review and merging of the various trees. In
particular, android has a branch of Robin Watts' Tremolo code
(arm-optimized tremor-lowmem) which I don't think anyone on the Xiph
side has looked at.

I used git-svn to import the history, which isn't great at following
svn branches, so a sanity check that we've got the appropriate history
would be appreciated. You can find the new repository at

  https://git.xiph.org/?p=tremor.git

Core developers should already have push access using the git< at >git.xiph.org url.


N.B. We used to have a separate list for tremor discussions, but it's
seen very little traffic in the past few years. I've marked it as
deprecated; we can just as well discuss maintenance issues for all the
implementations here.

Cheers,
 -r

Enhanced Podcasts with Ogg Vorbis (Chapter Marks)

Georg Holzmann — 2012-01-06T10:25:06

Hello list!

Since it is more and more common these days to create enhanced podcasts 
(= audio files with additional chapter markers [1]), I am searching for 
an alternative to the currently used MP4 format.
ID3 also puplished an official standard for chapters [2], but nobody 
seems to use and support it.


Now I am trying to get something similar with vorbis audio (and maybe an 
additional container).
So far I am aware of the following possibilities:

1) Matroska Audio with Chapter Marks

Using the matroska container, it is possible to generate a matroska 
audio file (mka) with included chapters (see [3] for the exact command).

PROS:
- many media players and some hardware players support matroska, see [4]
- with e.g. VLC (or foobar) you see chapters and are able to seek in the 
audio file - very nice!
- android 4 should support matroska natively [5] (but I don't think that 
it's possible to see the chapters ?)

CONS:
- many hardward players, which support ogg vorbis, are not able to play 
mka files
- I fou

Enhanced Podcasts with Ogg Vorbis (Chapter Marks)

Georg Holzmann — 2012-01-06T10:25:06

Hello list!

Since it is more and more common these days to create enhanced podcasts 
(= audio files with additional chapter markers [1]), I am searching for 
an alternative to the currently used MP4 format.
ID3 also puplished an official standard for chapters [2], but nobody 
seems to use and support it.


Now I am trying to get something similar with vorbis audio (and maybe an 
additional container).
So far I am aware of the following possibilities:

1) Matroska Audio with Chapter Marks

Using the matroska container, it is possible to generate a matroska 
audio file (mka) with included chapters (see [3] for the exact command).

PROS:
- many media players and some hardware players support matroska, see [4]
- with e.g. VLC (or foobar) you see chapters and are able to seek in the 
audio file - very nice!
- android 4 should support matroska natively [5] (but I don't think that 
it's possible to see the chapters ?)

CONS:
- many hardward players, which support ogg vorbis, are not able to play 
mka files
- I fou

Enhanced Podcasts with Ogg Vorbis (Chapter Marks)

Georg Holzmann — 2012-01-06T10:25:06

Hello list!

Since it is more and more common these days to create enhanced podcasts 
(= audio files with additional chapter markers [1]), I am searching for 
an alternative to the currently used MP4 format.
ID3 also puplished an official standard for chapters [2], but nobody 
seems to use and support it.


Now I am trying to get something similar with vorbis audio (and maybe an 
additional container).
So far I am aware of the following possibilities:

1) Matroska Audio with Chapter Marks

Using the matroska container, it is possible to generate a matroska 
audio file (mka) with included chapters (see [3] for the exact command).

PROS:
- many media players and some hardware players support matroska, see [4]
- with e.g. VLC (or foobar) you see chapters and are able to seek in the 
audio file - very nice!
- android 4 should support matroska natively [5] (but I don't think that 
it's possible to see the chapters ?)

CONS:
- many hardward players, which support ogg vorbis, are not able to play 
mka files
- I fou

multithreaded vorbis encoding

Thomas Eschenbacher — 2011-11-17T18:33:55

Hello,

I have some code that encodes PCM data into a vorbis stream, using the
workflow shown in http://www.xiph.org/vorbis/doc/libvorbis/overview.html.

By analyzing that process with valgrind/cachegrind I found out that 95%
of the time used for encoding is spent in the function
"vorbis_analysis(&vb, 0)".

My question now is: would it be technically possible to parallelize this?
I think about using multiple worker threads for the vorbis_analysis calls,
especiall on multi core systems this would be a big benefit.

=> Are the blocks passed as "vb" independent from each other,
   so that I can encode them in parallel?

regards,
   Thomas

Multi-channel labels in Vorbis comments

Daniel James — 2011-11-16T10:54:36

Hi devs :-)

At the Mozilla Festival in London recently, I took part in a Hack the DJ
workshop, looking at ways to take digital DJ'ing to the next level:
http://www.bbc.co.uk/blogs/researchanddevelopment/2011/11/setting-a-dj-challenge-at-the.shtml

One of the ideas proposed was stem mixing, using multichannel files in
DJ applications. A proprietary implementation of this idea is Fireplayer
(http://itunes.apple.com/gb/app/fireplayer-remixer/id367791455?mt=8) but
this app is built with the intention that users will buy remixable
versions of (a very limited number of) well-known songs from an in-app
store.

We would like to work towards a new open standard for stem mixing,
something that is compatible with sharing our mixes on the open web -
legally, of course - but could also be used by record labels that sell
tracks to DJs.

For example, eight channel Ogg Vorbis files where the first two tracks
are a stereo mix of the drums, third and fourth stereo bass, fifth and
sixth stereo vocals, and seventh and eighth t

Out of memory handling?

Gábor Kovács — 2011-11-07T08:43:02

Hi,

As far as I see in libvorbis most of the places where _ogg_malloc(),
_ogg_realloc and _ogg_calloc() is called the return value is not checked
for out of memory condition. Is there any plan to change this?

For my use-case the application is not allowed to crash. So if libvorbis
can not handle this condition, I guess I would need to preallocate a large
enough custom memory heap for all kind of vorbis files and override the ogg
allocation macros.

Best regards,
Gábor
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vorbis_fpu_setround()

Gábor Kovács — 2011-11-07T08:02:14

Hi,

I think there is a minor bug in vorbis_fpu_setround().
The "temp" is an output argument instead of input.
Intel compiler warning:

lib/os.h(102): warning #592: variable "temp" is used before its value is set
            "fldcw %1\n\t":"=m"(ret):"m"(temp): "dx");

I would recommend something like:
            "fldcw %1\n\t":"=m"(ret),"=m"(temp): : "dx");

Best regards,
Gábor
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Inverse MDCT formula missing from Specification

Foo Bar — 2011-08-01T13:05:02

Hi

The inverse MDCT formula seems to be missing from the vorbis specification. From reading the source code of stb_vorbis (http://nothings.org/stb_vorbis/) I've a good idea of what it might be, but cannot find the formula anywhere in the official specification (the one on http://xiph.org/vorbis/doc/Vorbis_I_spec.html ). I also understand that the formula stated in the eusipco_corrected.ps paper is incorrect (the link from the specification to the paper is also dead).

From the stb_vorbis.c source:

void inverse_mdct_slow(float *buffer, int n)
{
   int i,j;
   int n2 = n >> 1;
   float *x = (float *) malloc(sizeof(*x) * n2);
   memcpy(x, buffer, sizeof(*x) * n2);
   for (i=0; i < n; ++i) {
      float acc = 0;
      for (j=0; j < n2; ++j)
         // formula from paper:
         //acc += n/4.0f * x[j] * (float) cos(M_PI / 2 / n * (2 * i + 1 + n/2.0)*(2*j+1));
         // formula from wikipedia
         //acc += 2.0f / n2 * x[j] * (float) cos(M_PI/n2 * (i +

ov_read error on macosx

Juan Miguel Martín Muñoz — 2011-07-15T11:48:33

Hi, I have this code to decode ogg data:

  unsigned long PSS_OggStream::DecodeOggVorbis(OggVorbis_File
*psOggVorbisFile, char *pDecodeBuffer, unsigned long ulBufferSize, unsigned
long ulChannels)
  {
  int current_section;
  long lDecodeSize;
  unsigned long ulSamples;
  short *pSamples;

  unsigned long ulBytesDone = 0;
  while (true)
  {
#ifdef WIN32
lDecodeSize = ov_read(psOggVorbisFile, pDecodeBuffer + ulBytesDone,
ulBufferSize - ulBytesDone, 0, 2, 1, &current_section);
#elif MACOSX // El cuarto parametro es lo del big endian de los cojones
lDecodeSize = ov_read(psOggVorbisFile, pDecodeBuffer + ulBytesDone,
ulBufferSize - ulBytesDone, 1, 2, 1, &current_section);
#endif
  if (lDecodeSize > 0)
  {
  ulBytesDone += lDecodeSize;

  if (ulBytesDone >= ulBufferSize)
  break;
  }
  else
  {
  break;
  }
  }

  return ulBytesDone;
  }

This works fine on Windows, but on MacOSX, after ov_read call, pDecodeBuffer
is full of 128, -128, 0... I mean, non sense data.

Any idea???

Thanks!
____________________________

Porting TREMOR low mem on memory limited systems

Ramkumar R — 2011-07-01T12:08:03

Hi,

We are trying to port Vorbis Tremor low memory branch on a system with 48KB
RAM. Doing mallocs is out of question and we have to store the code books in
external DRAM and retrieve them back.

Is there somebody who has done that before and any pointers on how to do it?

Regards
Ramkumar
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Error in floor0 decode specification

Роман Донченко — 2011-04-18T16:37:23

Hello,

It seems there is an error in the algorithm for floor0 decode in section
6.2.2 of the Vorbis specification.

Here's the relevant part:

3         3) [coefficients] is an empty, zero length vector
4         4) [booknumber] = read an unsigned integer of ilog(
   [floor0_number_of_books] ) bits
5         5) if ( [booknumber] is greater than the highest number decode
   codebook ) then packet is undecodable
6         6) [last] = zero;
7         7) vector [temp_vector] = read vector from bitstream using
   codebook number [floor0_book_list] element [booknumber] in VQ context.
8         8) add the scalar value [last] to each scalar in vector
   [temp_vector]
9         9) [last] = the value of the last scalar in vector [temp_vector]
10        10) concatenate [temp_vector] onto the end of the [coefficients]
   vector
11        11) if (length of vector [coefficients] is less than
   [floor0_order], continue at step 6

If implemented as written, it would mean that [last] does nothing, since
the only time it is

Wav to Ogg Vorbis conversion

Dmitriy Reznik — 2011-03-10T16:27:18

Dear experts,

 

I am trying to use an example I found here:

 

http://svn.xiph.org/trunk/vorbis/examples/encoder_example.c

 

 

Could you please tell me what float** buffer is needed for (I don't see
it used anywhere). 

Also, could you please tell me what this code is doing:

 

      for(i=0;i<bytes/4;i++){

        buffer[0][i]=((readbuffer[i*4+1]<<8)|

                      (0x00ff&(int)readbuffer[i*4]))/32768.f;

        buffer[1][i]=((readbuffer[i*4+3]<<8)|

                      (0x00ff&(int)readbuffer[i*4+2]))/32768.f;

      }

Again, buffer is not used afterwards.

And where does the actual encoding happens?

Maybe you could recommend some tutorial to understand how this works?

Thank you very much,

 

Dmitriy

 

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libao 1.1.0 released

Monty Montgomery — 2011-02-22T23:21:53

Xiph.Org is pleased to announce the release of libao version
1.1.0. This release consists of minor functionality improvements and
several important bugfixes. The 1.1.0 API/ABI is fully backwards
compatible with 1.0.0; All users of libao are encouraged to
upgrade to the 1.1.0 release.

Bugfixes in this release include:

    * Fix Mac OS X AUHAL support to properly handle suspend/wakeup,
      headphone plug/unplug, other hardware events
    * Fix Pulse driver handling of empty channel matrix
    * Fix Roar driver to not block on SLP lookup during probe
    * Improve/correct latency setup in ALSA (see Trac #1762)
    * Add missing ctype.h header in build (see Trac #1760)
    * Correct ao_example.c source to not pass dangling pointer for
      the matrix argument.
    * Fix 24 bit playback in ALSA plugin
    * Fix segfaults when closing a driver that did not successfully open.
    * Fix building Mac OS X driver AUHAL compilation for 10.5
    * Fix compilation of sndio plugin

Improvements in this release includ

vorbis encoder performances under arm platforms

marco — 2011-02-16T12:29:47

Hi, 

i am trying to do vorbis encoding on an arm-powered device (an iPhone), but i am getting really poor performances. 
I am using vbr encoding and setup it with:
vorbis_encode_init_vbr(&vrb_info, 2, 44100, .2);

the cpu load seems not to change according to the quality, on a iPhone 3gS seems to average at about 60-75 % of the cpu time (i capture audio from theinternal mic and encode it on-the-fly to vorbis). 
Some questions i seem not to find a definitive answer:
- floating point operations are the main cause of poor performances?
- i see there's the tremor project for decoding. Is something similar planned for encoding (is it feasible?)?
- any compile-time optimizations i could try? (i am quite sure there is nothing i could do on this side)
- which is the fastest encoding mode using vorbis? i am using that vbr without bitrate management at the moment, is there something faster?

sorry to bother you,

marco