similar to: vorbisfile ov_pack_pcm patch

Hello. I noticed that when reading data with ov_read_float(), you can get values outside [-1..1] when the stream is encoded at lower quality, but with higher quality, the values trim down to inside [-1..1]. Looking at the plot from -q10, the data from ov_read_float seems clipped. I've made some plots of encoding the start of Rammsteins Feuer from the xXx soundtrack (it's a pretty loud

Is there any particular reason why ov_read() packs floats to integer PCM inline, rather than being implemented in terms of ov_read_float() and a separate packing fucntion? There are obviously many advantages doing audio manipulation on the floats before packing, but right now you have to reinvent the packing stage yourself - in a replaygain backend that I'm working on, I ended up copying

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

Hi! I solved the problem using ov_read_float instead ov_read. Still do not know why ov_read returned garbage, but ov_read_float works fine! Thanks! El 17 de julio de 2011 10:41, Juan Miguel Mart?n Mu?oz <jmcubo at gmail.com>escribi?: > Hi, thanks for your replies! > > ogg.k, I'm sure MACOSX is defined. When I'm debugging line by line the > program runs into ov_read.

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()

Hello, Am I missing something? float** pcm_channels; int bitstream; int where = ov_pcm_tell(&vf); -> The Result is "0"; ov_read_float(&vf, &pcm_channels, 1, &bitstream); -> pcm_channels[0][0] is "0.000338580" ov_pcm_seek(&vf, 0); char* buffer = new char[2]; ov_read(&vf, buffer, sizeof(buffer), 0, 2, 1, &bitstream); short temp =

Hi, in ov_read_float.html there is a small mistake: long ov_read(OggVorbis_File *vf, float ***pcm_channels, int *bitstream); hould read: long ov_read_float(OggVorbis_File *vf, float ***pcm_channels, int samples, int *bitstream); It seems this function has been changed (compared to the RC3). Anything else worked fine for me with the 1.0 release. Olaf <p>--- >8 ---- List archives:

Some months ago I tried to get some answers (and didn't) for a problem I've had with ov_pcm_total for files 8kb or smaller. I have no idea whether it should make a difference, but I operate inside Visual Studio, with C++ 6.0 and MFC. The value returned by ov_pcm_total doesn't agree with ov_read. My solution had been to add 0.2 to the buffer size, so that ov_read would make it to

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm writing a simple application to decompress a set of Ogg files to raw PCM format, using vorbisfile, because that is the simplest way of doing it. These Ogg files are sounds for a game, and thus sometimes very short (about .25 s). However, vorbisfile has severe problems with Ogg files that are smaller than the CHUNKSIZE which is #defined

Hi, I need to adjust the output volume from ogg123 the same way that mpg123 has the "-f n change scalefactor [32768]" option. I need to do this within ogg123, instead of alternatives that involve pipes, such as piping the output to sox. I do not believe that this is currently possible in ogg123, and I'm looking for some suggestions as to how it can be accomplished by modifying

Hi, I need to adjust the output volume from ogg123 the same way that mpg123 has the "-f n change scalefactor [32768]" option. I need to do this within ogg123, instead of alternatives that involve pipes, such as piping the output to sox. I do not believe that this is currently possible in ogg123, and I'm looking for some suggestions as to how it can be accomplished by modifying

Greetings, I'm having some sort of problem using ov_read_float(). Everything looks good to me, but I'm trashing memory somehow, so clearly I'm screwing something up. What confuses me is why it takes a ***float for the buffer. I call it like so: float **buffer bytes_read = ov_read_float(&vf, &buffer, 0) memcpy (b, *buffer, 0); This compiles fine, but after a few reads,

Hi, this is my first post, I hope it's the right mailing list (or should it be vorbis@xiph.org?) I just downloaded "SDK for MacOS9" and tried to compile the "vorbisfile_example.c" sample with CodeWarrior for Mac 8 on a Mac OS 9.2 machine. I included the libOgg and libVorbis libraries in my project, but the linker cannot file ov_clear(), ov_read(), ov_pcm_total(),

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 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 =

I think I have bumped into the small file error, or I''m doing something stupid. The files are short audio effects for a game (embedded in our own data format). Sample info: Vorbis packets: 1 (4 kb) Samples: 28672 Samplerate: 22 kHz Channels: 2 This is what I''m doing when I want to get a number of bytes from the stream: The problem is that ov_pcm_tell always returns 0. And so

Hello, hope you're fine. And hopefully someone can help me. I wrote a short demonstration to show you my problem (see below). I'm asking always for 64 samples. When "ov_read_float()" is finished, I want to seek back to sample 1, and ask immediately for the next 64 samples. But when I run this small app, I get an totally high CPU load. Is there a way to it speed up? Or do I

Ok here is the breakdown. I am trying to impliment streaming into our game using openAL and ogg-vorbis. In the non-streaming function when I call ov_read on the file I should be streaming the buffer gets filled with the correct data. However, when I run ov_read on the same OggVorbis_File when I read the buffer gets filled with 0's. The buffer has not been zero'd and starts filled

Has anyone had experience using ov_read in a thread with streaming directsound buffers? i have tried the following, but it just produces a repeating garbage noise. i would appreciate some help. notes: test.ogg in my code was a song that i converted to the vorbis format (Gorillaz - Cling Eastwood) pcmData is defined like this: char pcmData[4096]; the ogg file and the directsound buffer