similar to: [Bug] PCM file not recognized

I am looking for any application for windows or qnx that can encode a wave file. --- >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 'speex-dev-request@xiph.org' containing only the word 'unsubscribe' in the body. No subject is needed. Unsubscribe messages sent to the list

I upgraded to Leopard (version 10.5 of OS X) a few weeks ago. Although I was a bit disappointed, but not surprised, to see that FLAC support isn't built natively into the OS, I was very happy to notice recently that Apple ships source code for a FLAC encoder and decoder codec component in /Developer/Examples/CoreAudio/AudioCodecs/ FLAC.xcodeproj. All that is necessary to build the

>> recently that Apple ships source code for a FLAC encoder and decoder >> codec component in /Developer/Examples/CoreAudio/AudioCodecs/ >> FLAC.xcodeproj. > > You can find a slightly more detailed review of the Apple's FLAC codec > implementation in my blog post at: > http://barelyfocused.net/blog/2007/10/28/flac-support-in-mac-os-x-105-leopard/ Thanks for the

I forget the correct terminology, but anyway, sumthin struck me while getting help for sorting out my previous problem. Among other things (apart from doing the gigs with my computer) I use the G4s (and soon hopefully 1 of those nice dual G4s we just got) to record tracks of the different ppl & bands on our course (which is BTEC Popular music btw) using Pro Tools. Since Vorbis has the

Hi list, I'm trying to use an AudioCodec Mp114, 4 FXO Media gateway. I went trough what i could find in wiki and also trixbox forum and so far no good results. i had this in trixbox frorum : http://www.trixbox.org/forums/trixbox-forums/trunks/trixbox-2-2-and-audio-codes-mp-114-fxo-setup any successful installation? or how to? -------------- next part -------------- An HTML attachment was

Hi, Another newbie question. I'm trying to figure out how to fit vorbis into the Apple AudioCodec API and it asks your codec to be able to answer questions about itself. One I'm not sure about, does vorbis encode a fixed number of frames per packet for any given track? I know it's VBR, but that doesn't necessarily imply variable # of frames per packet. Thanks, -n8 --

Hello, I have an p2p voice chat application using WAVE PCM (winmm). Now i am trying to add opus encoding to it to send it over the TCP/IP and then decoding it back to play - but without success (without opus it works ok) Here is an example of my code. I get message from input device then encode with opus then decode it back to output wave header and play. Doing it i hear only noise in my

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.

I need it to be able to take a stream of PCM data and encode it. Right now it works if I am recording from a source (i.e. microphone or with "what you hear") but when I try to use another API (Team Speak) to capture the PCM data it doesn't seem to be encoding correctly. (I know the PCM data is correct. I am able to use it to create a wav file.) I'm also having a problem with

On 07/01/16 10:04 AM, Amit Ashara wrote: > opus_decoder_ctl(sOpusDec, OPUS_SET_LSB_DEPTH(ui32BitsPerSample)); OPUS_SET_LSB_DEPTH only affects the encoder. If you check the return value here you should get OPUS_UNIMPLEMENTED. > output_samples = opus_decode(sOpusDec, (const unsigned char > *)&pcRdBuf[0], len, opi16_out, (ui32SizeOfWrBuf/ui8ScaleFactor), 0); I suspect the issue is

Hello Mark, Thanks. Let me try the proposed configuration first to make sure that linear 8-bit PCM is retrieved. Regards Amit On Sat, Jan 9, 2016 at 12:28 PM, Mark Harris <mark.hsj at gmail.com> wrote: > opus_decode() produces 16-bit signed linear PCM, and > opus_decode_float() produces 32-bit floating point PCM that is useful > when you want a higher bit depth. > > If you

Hello All, Turned out to be a coding error. The modified buffer was not being allocated to write back to the File System. After correction the mono 8 and 16 bit for 8K-48K works well now. Regards Amit On Mon, Jan 11, 2016 at 11:05 AM, Amit Ashara <ashara.amit at gmail.com> wrote: > Hello Mark, > > The resulting 8 bit file has a lot of squelching noise compared to the 16 > bit

When I first install wine using the default repository in Ubuntu 9.04 and run winecfg I get the following error: wine: created the configuration directory '/home/bill/.wine' err:alsa:ALSA_CheckSetVolume Could not find 'PCM Playback Volume' element err:alsa:ALSA_CheckSetVolume Could not find 'PCM Playback Volume' element fixme:system:SetProcessDPIAware stub!

Hi, I have asterisk-0.4.0 running. When I make a call between an ATA186 and Asterisk using ulaw or alaw codec, all is fine. I installed the CVS version and tried the same thing but the voice is choppy. The installation was done on the same linux server. The stats on the ATA186 show no packet loss but a great number of "late packets". The stats when running version *-.0.4.0 do not

I have just finished installing all the pieces of Redhat Linux 9 (2.4.20-8), Asterisk-0.9.1, Gnophone-0.2.4 on my IBM T30. Audio card is SoundMAX Integrated Digital Audio. Not sure what the chips are. But everything works on both W/XP and RH9. (Machine is obviously dual boot) Everything starts, Asterisk is up, sound/audio is great for CD player, Volume controls, Voice recorder, RealPlayer, etc.

This is my first post here so Welcome everyone :D. Few days ago I compiled wine 1.1.8 on my gentoo box and I successfully installed Lord Of the Rings On line. The game is running surprisingly good for me but I run into minor but pretty irritating problem, every time I lunch or stop the game my pcm channels from first sound card device are being muted. I'm using oss driver. On alsa driver

Hello ! Could you be so kind to hint me about supporting Powercom PCM SPD-1000U by nut ? I didn't find this model in your HCL. Do you have plans for adding this model in your HCL or maybe I can use some similar driver ? I tried to connect my SPD-1000U with FreeNAS using another drivers for Powercom from HCL, that use nut, but without success, seems because it's not in HCL. Some output

Hello, given a Speex-encoded ogg file, is there a way I can compute the total pcm size (that is, the number of samples after decompression)? I know I can get the frame size from the decoder with SPEEX_GET_FRAME_SIZE, and that the number of frames per packet is written inside the header. But this brings me to the question: how many packets are there in the file? I guess this is more related to

Speex 1.0.3 uses floats, but the unstable version 1.1.x uses short. Jean-Marc Le ven 16/04/2004 16:04, Kenji Chan a crit : > Im confused about the format of the PCM/WAV data for speex_encode & > speex_decode > > speex_encode(enc_state, input_frame, &bits); > > In the manual, it says input_frame is a (float *) pointing to the > beginning of a speech

Le ven 16/04/2004 à 16:39, Kenji Chan a écrit : > Ok, I'm using 1.1.4, so I should pass signed short(wav data) directly to > speex_encode()? First, I'd like to ak why you want to use 1.1.4. If you don't need the new features (fixed-point or preprocessor), you should probably stick to 1.0.3, which has received more testing. If you choose to use 1.1.4, you need to send shorts to