similar to: ov_raw_tell() not working properly!

Hi all, i use speex preprocessor features in this way: =================================== #define NN 160 /* 20msec di audio */ ... int tbc=0,c,d,ret; spx_int16_t TEMP_Buffer[NN]; char DLECODE; /* Inizializza il preprocessore Speex se non inizializzato */ if(Modem->speex_pp_state == NULL) { Modem->speex_pp_state = speex_preprocess_state_init(NN,AUDIO_SAMPLERATE); }

Dear People, I just went back to an old piece of C code. On trying to compile it with the R math standalone C library I got the following error. Can anyone enlighten me what I am doing wrong, if anything? C file (rr-sa.c) follows. I'm on Debian sarge. I'm running R version 1.8.1. Gcc is version 3.3.1. Thanks in advance. Faheem.

Hi all, as in subject, speex_preprocess inject noise in my data. Someone can help ? Here's the way that i'm using: #define NN 160 /* 20msec di audio */ #define AUDIO_SAMPLERATE 8000 spx_int16_t TEMP_Buffer[NN]; speex_pp_state = speex_preprocess_state_init(NN,AUDIO_SAMPLERATE); c = denoise; speex_preprocess_ctl(speex_pp_state, SPEEX_PREPROCESS_SET_DENOISE,&c); c = agc;

Hi, I'm busy working on a method of synchronising playback between two ogg decoders. To do this reliably, I need to create a cross-reference of the raw & pcm offsets in the bitstream. I have tried using the ov_raw/pcm_tell calls, but it appears as if the ov_raw_tell() only updates itsself in steps of 4096 bytes (as each read callback is done). This gives me a cross-reference every

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,

Hello, i want to use speex out of visual basic. So i have written a few wrapper functions in VC++ (v6) wich i can call from VB (v6). But i have problems with the encode and decode functions. Encode wrapper Function: I want to copy the value of an single array in VB to a float array in C which can be encoded. And the output of the encoder have to be unsigned char because in VB its an byte

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

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 *

I found a problem with ov_raw_seek function in oggvorbis win32sdk 1.0. When i called this function it returned OV_EINVAL immediatelly. I solved the issue by replacing the ov_raw_seek by ov_pcm_seek, which worked perfectly, but i would like to use the raw variant due to performace reason . Is anybody aware of this strange behaviour ? Martin Cesky <p><p>--- >8 ---- List

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'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

Hi, While testing ReplayGain (so it could be related to a bug), I noticed the following gain comments for a file: RG_PEAK=1.71580 RG_RADIO=-7.91 dB RG_AUDIOPHILE=-6.72 dB I've never seen such a large peak. Not that I've looked much, nor have I analyzed the file further. I just thought I should mention it. :) The file is the track Board Burner by Mixmaster Mike, available at:

Hello, All. I am making an Ogg Vorbis decoder based on Tremor. The Vorbisfile library provides a high-level API which enables us to seek in the file, but it needs to declare a pointer to OggVorbis_File structure. Because I am supposed not to use file, so I can't use the ov_pcm_seek() or ov_raw_seek() functions to seek in the file. For decoding an Ogg Vorbis file, I first put it in

I need to be exposed to a way of seeking theora streams. I have repeatedly found the same question asked on a different forums but could not find an answer. I'm working on a game and started integrating video with theora, the sound works just fine with ogg/vorbis(i find it easy to seek using ov_raw_seek for audio files). Could someone please point me to some example/code of how can i seek with

Is there a straight forward method for seeking straight to a particular logical stream using vorbisfile? I'm guessing I'll have to look the offset for a given link up in vf->offsets, and use that in ov_raw_seek? Thanks, 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

Hi, I have been having some problems with ov_pcm_seek in vorbis_file:- Any seeks to a sample within the first second of the stream (i.e. < 44100 samples) fail. Seeks to samples beyond this 44100 limit are fine. It appears to be associated with vorbis_file trying to read the preceding page (odd for the beginning of the file), attempting an ov_raw_seek, then bombing out with an OV_EINVAL.

Well, your " OggPlayer::release; OggPlayer::open; before OggPlayer::seek" seems to work! Thank you so much! Also, I'll take a look at the Opus codec, thanks for the advice! For your information, here was the problem in details: let's say I download an audio file whose length is 300 seconds. When, for example, 15 seconds of the file were downloaded, I called OggPlayer::open().

Thanks for your answer. But as I said, I don't have any problem seeking in a local file. Also, I have full control on the ogg file I play, since I encode it by myself. I first would like to know if the code I provided ( https://gist.github.com/mregnauld/3f7cdc43b02ec3bbd91641b9333ba072) is good practice when it comes to extracting and seeking in a file that is being downloaded. Thanks.

Hi, I am trying to find the specs for ogg vorbis but cannot find a complete spec online, I need to know the format of the headers and what to look for in order to sync, read, write, and manipulate the format easily with some functions. I know there are some examples of the code within the ogg vorbis dev tools and there is a fileogg library that is mentioned but I have not seen this elsewhere. I

Hi folks, I'm doing a merge of my current branch onto the mainline (for testing) today. I believe it to be stable. Just a little more vorbisfile testing. After the merge, I have a few more patches to apply, then onto cascading/coupling. New stuff: Floor backend 1 and residue backend 1; both are present, but the mainline modes won't use either yet. Naturally, both are enabled for