similar to: Peak value

I'm considering patching my copy of ogg123 to support vorbisgain tags. Is there any interest in folding this sort of patch into the main line ogg123? Has anyone done this already (or to another command line ogg player)? John --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to

I've been experimenting with the ideas of Replay Gain[1] and find that ogg123 doesn't have a way of specifying the scaling applied to replayed samples (like -f in mpg123). Looking at libvorbisfile, I see no function exactly matching this possibly desirable behaviour. ov_read() scales by either 128 (byte output) or 32768 (word output), but there's nothing in between. ov_read_float()

This is also pretty simple. Just a small patch to configure.in to set the right compiler flags and changes to os.h to (a) not have deeply nested ifdefs, making the file needlessly hard to read, (b) break up the sections by functionality and (c) don't assume that every non-Win32 has the floating versions of math library functions. -tim Index: configure.in

I got this one this morning, but didn't notice it til now: --- (gdb) bt #0 0x4003be91 in vorbis_block_clear (vb=0xbffff71c) at block.c:156 #1 0x4003ce36 in vorbis_analysis_blockout (v=0xbffff71c, vb=0x8050b80) at block.c:591 #2 0x4002ff23 in _prefetch_all_headers (vf=0xbffff540, dataoffset=-1073749296) at vorbisfile.c:324 #3 0x4003039c in _process_packet (vf=0xbffff540, readp=1)

Hello! 1. You should modify the 'vorbis_ftoi(f)' for the undefinied OS/compiler in the 'os.h' on this way: old: static int vorbis_ftoi(double f){ return (int)(f+.5); } new: #define vorbis_ftoi(f) ((int)(f+.5f)) Because in some cases we don't use this function, and the compiler writes many error messages. 2. Could you write a frequency and bitrate independent

Another one at block.c:176: --- Title: We The People Artist: DJ Lithium Presents Bitstream is 2 channel, 44100Hz Time: 58:29.07, Bitrate: 100.1 Program received signal SIGSEGV, Segmentation fault. [Switching to Thread 1024 (LWP 27207)] _vds_shared_init (v=0xbffff73c, vi=0x4024efe0, encp=0) at block.c:176 176 b->modebits=ilog2(ci->modes); (gdb) bt #0 _vds_shared_init

I'm using seekable_stream_decoder, And., this is my write_callback. I'm not getting the required output. The PCM i get is not the proper music. Am I doing something wrong here? FLAC__StreamDecoderWriteStatus AFLACStreamPlayer::StreamWriteCb ( const FLAC__SeekableStreamDecoder *decoder, const FLAC__Frame *frame, const FLAC__int32 * const buffer[], void *client_data) { int Channels,

(I've already posted this message month ago, but nobody answered, may be it was not delivered to newsgroup?) Hello, All. I wrote an encoder-decoder based on example from OGG-Vorbis SDK. This encoder can encode a large amount of small WAV-files with equal parameters into one sound archive. Then I decode this sound archive back into large amount of small WAV-files. Theese files are

I'm working on an application where I need to record the current playing position and return to it later. and I need this to be done the most efficient way, so I used ov_raw_tell() and ov_raw_seek() because the documentation says they are the best when speed is a concern. but the problem is that sometimes ov_raw_tell returns the same value before and after calling ov_read; here's an

This one may be my fault.... I've put a temporary fix for libvorbis not linking, I altered libvorbis_la_LIBADD = to libvorbis_la_LIBADD = $(OGG_LIBS) in the makefile, hence the temporary and continued with the build. I'm now getting gcc -DPACKAGE=\"libvorbis\" -DVERSION=\"1.0.0\" -DHAVE_ALLOCA_H=1 - DHAVE_ALLOCA=1 -I. -I. -I../include

I can't be totally sure what 'short' is on your platform, but if it's (probably) 16-bits, it looks like you're treating the samples in buffer[] as packed 16 bit numbers. but all samples in buffer[] are 32-bit signed integers in host order, regardless of the bits-per-sample of the frame. so to get them down to shorts (assuming they'll fit), do like: FLAC__int32 *

Here's a patch for the current CVS version of libao which makes the IRIX plug-in compile and work properly. Michael --- ./src/plugins/irix/ao_irix.c Fri Aug 3 19:56:17 2001 +++ ../libao-0.8.2/./src/plugins/irix/ao_irix.c Thu Apr 11 20:16:41 2002 @@ -36,12 +36,13 @@ #include <ao/ao.h> +#define AO_IRIX_BUFFER_SIZE 32768 typedef struct ao_irix_internal { - static ALport alport =

Hi all, I have implemented ReplayGain support for Vorbis. If you are not familiar with it, it is basically a method of making sure all your files have equal loudness, remove the need for normalization and prevent clipping during playback. The process is totally lossless, and supporting it requires minimal work. More info about the exact workings can be found on www.replaygain.org (recommended

Here's a patch to vorbisfile.c in libvorbis that splits the float to integer pcm packing out of ov_read into it's own function. The good stuff: - ov_pack_pcm can be used by folks who decode to float pcm with ov_read_float to do various sound manipulations (eg, preamp, fades, replaygain). No more need to roll your own, which will really begin to pay off when 24 bit output gets

Hi! I tried to do an x64 build of libvorbis but was stopped by the inline assembly in lib/os.h. Attached is a patch that modifies this file so libvorbis will compile for Windows x64. Tested with Visual C++ 2008 Express (x86_x64 compiler from current Windows SDK; 15.00.21022.08), seems to be working fine with no other issues so far. Important: As you can see in the patch, I also changed the

A common need in R frontends is to provide some sort of read, (parse), evaluate, print loop. However, there are also a number of points where frontends may want to differ from the standard REPL as available e.g. in R_ReplDLLdo1(). First some thoughts on what is needed, and what is already there, or missing. If you want to skip over this, a short summary is provided in the second half, below

Hi, Not sure if this should be in the dev or users list, but here goes. I am using the FLAC seekable stream decoder in my MACH3 laser games emulation in MAME (www.boldtower.com/mach3). The game has a soundtrack (originally on a laser disk) which I have flac encoded and play back in the game. I did the implementation under Linux which works fine, the callbacks are easy to set up and I've

On Thu, 14 Dec 2000 10:06:39 +1100, Michael Smith wrote: >>I've done a port for OS/2 using Watcom C++, including a simple command line >>player based on vorbisfile_example.c, which can be downloaded from >>http://silk.apana.org.au/utils.html >>It'd be nice if my changes could be commited to CVS, should I just post >>patches here? > >Yes, unless

When I tried to compile the vorbis module as a Framework, using ProjectBuilder, I got 1058 warnings (excess elements in scalar initializer, near initialization for ''psy_set0.toneatt', 'vorbis_ftoi' defined but not used, unused variable 'k', braces around scalar initializer, and so on) and 65 errors. Some examples of errors: ------- vorbis/lib/psytune.c: undefined

Hi, This series of patches contains the second part of an initial implementation of the IEEE 802.1Qbg standard: code for the exchange of VDP VSI TLVs between a host with virtual machines and an adjacent switch. VDP profiles consisting of mode,mgrid,typeid,typeidversion,instanceid,mac,vlan can be given to lldpad with lldptool. A way to deliver profiles to lldpad from libvirt using netlink