similar to: Simple Function to Convert .wav file to .ogg file

Hi I'm new with ogg and also with using third party libraries. I'm making a C++ PocketPC program and I need a simple wav->ogg encoder function. But as I have tried to use ogg vorbis libraries I don't know what or how to use them. I think that best way would be to use libogg 1.1.3 and libvorbis 1.1.2. Do I even need both of them, since I only need to have capability of

Hi everyone, I am new programming with sound codecs. I am making a c++ application that needs to convert a .wav audio file to a .ogg one. If you can send me a demo or something similar I will be very pleased. Thank you for your time, Wanhaven. _________________________________________________________________ Busca desde cualquier p?gina Web con una protecci?n excepcional. Consigue la Barra de

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)|

Hi everyone, I am a newbe with the Ogg Vorbis libraries. I am trying to create a program in C++ that will allow me to take a 16 bit, stereo wave file at 44.1 khz and encode it into ogg vorbis. I have looked at the encoder example in the vorbis SDK, but I did not quite understand it. My question, therefore, is how to encode such a stream into vorbis? A simple tutorial would be very much

I am trying to convert some a few WAV files to ogg format (using vorbis-tools-1.0beta4 oggenc). I tried: $ oggenc -o nash.ogg nash.wav ERROR: Wav file is unsupported subformat (must be 16 bit PCM or floating point PCM ERROR: Input file "nash.wav" is not a supported format And: splay -d - nash.wav | oggenc -r -o nash.ogg - The resulting file when played with ogg123 played so

Hello. I'm trying to encode an Ogg Theora file from an unencrypted dvd using mplayer and encoder_example. I set up named pipes stream.wav and stream.yuv. I then run: mplayer -ao pcm -aofile stream.wav -vo null /dev/dvd & mplayer -vo yuv4mpeg -ao null -nosound /dev/dvd & encoder_example -v1 -a 1 stream.wav stream.yuv > lec1.ogg but I get the following error: File stream.wav is

hi, I downloaded the vorbis-sdk-1.0rc3 and wrote an application that plays ogg files using DirectSound(win32 application). It uses streaming technique - I'm implementing and passing IStream interface to the DirectX. The interface is used by DirectX to fill his buffers during playing. DirectX is assuming that it's readind from a wav file, so the interface decodes on-the-fly so that

Hello, I'm new to the list, so if this is an FAQ, please point me to the relevant docs. I'm playing with Theora, to see what kind of quality I can expect (I realise its not for production use yet). I've download the source from CVS, and got it to build, with the example encoder and player. While trying to find something to serve as a good source of input, I decided I'd use a

Hello list, it seems that Asterisk is unable to convert a wav-file into an alaw-file : [root at asterisk testing]# asterisk -rx "file convert testExtended2.wav testExtended2.alaw" Unable to open input file: testExtended2.wav [root at asterisk testing]# asterisk -rx "file convert testLong2.wav testLong2.alaw" Unable to open input file: testLong2.wav The wav-file is MONO,

A friend of mine recently had a problem with a "broken" .WAV file (as you call them) because oggenc first printed out a warning and then didn't accept the file because of a "unexpected EOF error". Because I was interested in the issue, I decided to take a look at the oggenc source, and in fact, it is your .WAV reader that's wrong. More precisely: there are two version

hello, i have vorbis sdk 1.0rc2 and i have could compile encoder_example.c in a console aplication project on VC 6. But when i execute the program, only appears the black visual display screen without obtaining result. where is my problems?? other question: in sdk there is for example, vorbis.dll and vorbis_d.dll, which is the difference? i have could compile any project like libvorbis

Okay, thanks to some stubbornness on my part and helpfulness on the part of people on this mailing list, I got the Ogg Theora alpha compiled up and installed and running. Yesterday, I went to the Prelinger archives (the public-domain video archive at www.archive.com) to find a high-quality original to re-encode as a test. The encoder worked just fine, and playback as well, and I thought I'd

By the way, I was delighted to find out that I could join .ogg files with cat, i.e. cat {file1.ogg} file2.ogg > outputfile.ogg <p>I successfully used this capability, together with oggenc and sox, to join two .WAV files in the following manner: <p>oggenc -q9 file1.wav ... oggenc -q9 file2.wav --- cat file1.ogg file2.ogg > outfile.ogg ox outfile.ogg outfile.wav I get

Dear All Can you please do me favor and let me know how can I convert *.wav files into 32 bit 44 KHz ? Please be informed that I have specific sound files in *.wav format that I converted them into *.gsm format with the aid of the following command : #sox FR00003.wav FR00003.gsm It got through but the voice quality is poor . I need to convert the original *.wav sound files (their file attribute is

On 9 November 2011 10:40, Micol lupen <galaxi_999 at yahoo.it> wrote: > Hi to all, > i am a university student, i have this problem, i want to create application for android this application must to record and listen ogg file, > i read that android use only ogg decoder for this problem i must use external library,wich library i can use ? can you show me some ?HowTo or book or site

There isn't a lot of guidance on how exactly to run the codec I compiled out of cvs just now, but I expected the following to work: cat test.wav | encoder_example | decoder_example But the decoder fails with: End of file before finding all Vorbis headers! Is this a bug? Or am I missing something fundamental? Should I attach the test.wav? -- Daniel --- >8 ---- List

There isn't a lot of guidance on how exactly to run the codec I compiled out of cvs just now, but I expected the following to work: cat test.wav | encoder_example | decoder_example But the decoder fails with: End of file before finding all Vorbis headers! Is this a bug? Or am I missing something fundamental? Should I attach the test.wav? -- Daniel --- >8 ---- List

Hi, Why do I get an error message when i'm running the "encoder_example.c" which is included in the SDK? The error occurs then the 44 bits are read and the vorbis_info struct is initiated (i.e. when the vorbis_encode_init(&vi,2,44100, -1, 128000, -1) - funcion is called. I also tried to read the data from a 16 bits 44,1 KHz stereo PCM - wave file but recived the same error.

I would like to convert a file to WAV49 for use with Asterisk using the linux command line. Specifically I would like to upload sounds to use for unavail.wav and busy.wav, but I'd like them to be compressed so that space is not wasted. I tried using SOX but havent found the correct command-line options to make it or Asterisk happy. Ideas?

Hi, I'm a highschool student, I'm looking for information about converting Wave file (*.wav) into Ogg file (*.ogg) and i found encoder_example.c inside libvorbis's example directory. I try to compile it using Microsoft Visual C++ 6.0 ,but it goes error like this : --------------------Configuration: encoder_example - Win32 Debug-------------------- Linking... encoder_example.obj :