similar to: Compressing small (<2k) .wav files

Hello, On the Windows platform, I would like to compile a version of the codec files ogg.dll, vorbis.dll, and vorbisfile.dll that are small. I notice that the provided Windows projects are very large...the release Vorbis.dll is > 1 mb. To me, that is huge. I am only using decoding features. I looked in the Zinf project, and they seem to have a version with the encoding functions

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 have found that there are some WAV files which have a "fmt" chunk size greater then 16 bytes (length indicator greater then 16). The patch is attached. What is the normal method for submiting patches? Regards, Chris Wolf <HR NOSHADE> <UL> <LI>application/octet-stream attachment: audio.c.patch </UL> -------------- next part -------------- A non-text

I am working on a project involving the conversion of a Mu-law sound format (Dialogic Mu-law wav format) file into a standard PCM encoded wav file format. Could somebody tell me if this feature is supported in the Vorbis software and if there is any source code available that performs this task. If not, does anybody know of any resources that might provide me with this tools or information.

Greetings one and all; I'm not subscribed to this list so I'm first sending this message to verify that mails from me make it through, and then later I'll send the juicy messages with patches. Also, the address I'm using is IPv6-only and doesn't often work, so drop me from any replies and I'll catch the archives, or drop only the hostname part to get an IPv4 address that

I've had this problem encoding oggs where the output file is small, like 24kbytes for a 4 minute song (tested at 128 and 160kbit). I'm running Windows 2000 and this has happened in oggenc, oggdrop, and CDEX, though I've also been able to get good encodings with each of these. I think the only clean encodings have been .wav's that I've made myself with SoundForge or CoolEdit,

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

Hello everyone, I tried to compress audio with opus-1.3.1/src/opus_demo.c recently, which works fine on mono and stereo data . Now I want to compress interleaved 7 channels pcm/wav ( recorded by Microphone array :6mic+ 1reference signal ) with opus, But I have not found an interface that compress multi-channels pcm/wav. 1、Is there a multi-channel compression interface can be used in my case? If

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

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

hi can the flac encoder be used to compress non-wave data? So suppose I rename a data file to *.wav, can i compress it then using the flac encoder to make it smaller? just wondering if this is possible -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/flac/attachments/20070903/b3091c47/attachment.html

I want to use Vorbis to stream music off the HD in my DirectX app. Currently, we have a generic streamer that writes raw digital sound data into the primary sound buffer. We plug sound sources (e.g. wav files, PCM files) into this streamer by having the sound sources have an interface something like this: class SourceDataSourceInterface{ virtual bool Init(char* pSource) = 0;

Sorry to bug you all, but I have some questions (I have beta4 at the moment; don't ask): I was reading through the VorbisFile API documentation because I'm writing code to deal with Ogg Vorbis-encoded files, and nowhere did I see a function or method for retrieving the byte-width of a stream. However, I did notice that ov_read() takes a word size as a parameter. Am I correct in assuming

Hi all, I'm having a little trouble with a Vorbis decoder hanging (actually hard looping the CPU) when I call ov_pcm_seek(&vorbis_file, 0) on a multiplexed theora/vorbis file. It appears to be the vorbisfile library that has the problem. The call to ov_pcm_seek never returns. Seeking in non-multiplexed, including chained, files is OK. Seeking to other values, including sample 1, works

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

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

More newbie questions, please bear with me =) I've got my high level code pretty well structured, and I'm now trying to decode vorbis streams on the fly using the ov_open_callbacks() routines. I can't seem to figure out how to tell it what output format I want -- is this possible? For example, I would like to be able to specify number of channels output (e.g. stereo vs. sum-to-mono

Hi, how can I convert .wav files to .WAV: # file greet.* greet.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz greet.WAV: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz using 'sox'? Thanks -- Domenico Viggiani

Hello. I'm running the swfdec-extract tool built from the 0.6.0 source tarball under gentoo and I'm finding that it's not able to rip the MP3s from the SWF file. I'm getting this: SWFDEC: ERROR: swfdec_codec_gst.c(269): swfdec_gst_decoder_init: failed to create decoder It manages to save the file as a WAV file, but I'd prefer the MP3 to get at the ID3 info in the file.

I am having trouble converting a .flv file uploaded from Adobe Flex / Flash Player 10 to a Red5 server using the speex coder: http://livedocs.adobe.com/flex/3/langref/flash/media/Microphone.html http://jira.red5.org/confluence/display/codecs/Speex+Codec Questions: 1. How do I extract the audio track out of such a .flv file? 2. How do I convert it from Speex to .wav PCM? Thanks.