similar to: encoding wave files into ogg vorbis?

________________________________ Hi David, Thank you very much for your reply. Since I need to resample the audio in the program itself, I decided to try out the resampling API in speex. But now, I have another problem. The resampled sound is very much distorted and clicks appear quite often. (I have attached the source code I used for testing it below). The test data I had was a file sampled

Hi everyone, I recently began using libspeex 1.2 Beta 1 on Windows using MS Visual C++. I have gotten a decoder and an encoder to work fine from the excellent sample code posted at the website. But I face a problem. I am working on using Speex in a program to play and create audio books encoded using Speex (currently testing it only; for these tests, I do not use Ogg to save the encoded

Hi everyone, I have a very annoying character encoding problem. Have a look to this: # ls -l M*mo-1.* -rw-rw-rw- 1 root root 8417218 6 sept. 2013 Mémo-1.aif -rwxr--r-- 1 hope hope 8417218 6 sept. 2013 Mémo-1.aif -rw-rw-rw- 1 root root 363175 6 sept. 2013 Mémo-1.m4a -rwxr--r-- 1 hope hope 363175 6 sept. 2013 Mémo-1.m4a Yes, it looks like two files have exactly the same name, but

I recently found myself in a situation where WAVE files from one PC were to be stored compressed on a second PC. As the second PC has a much faster CPU it made sense to transfer the WAVE file over the network and encode it on the destination PC. This also needs a mechanism to transfer the metadata (comments). The solution that sprung to mind was to add vorbis comments to the WAVE file. WAVE is

I recently found myself in a situation where WAVE files from one PC were to be stored compressed on a second PC. As the second PC has a much faster CPU it made sense to transfer the WAVE file over the network and encode it on the destination PC. This also needs a mechanism to transfer the metadata (comments). The solution that sprung to mind was to add vorbis comments to the WAVE file. WAVE is

Hi, I'm recording wave files but I cant get Asterisk to play them, only if they are in 8000 Hz. What is the maximum sample rate Asterisk can handle? I have been using 16-bit 44.1, 22050 and finally 8000 kHz. Many thanks, Christian

Hello forum I have there to files #----- virtual_host_controller.rb class VirtualHostsController < ApplicationController before_filter :capture_domain # GET /domain/1/virtual_hosts/1 def show @virtual_host = @domain.virtual_hosts.find(params[:id]) respond_to do |format| format.html # show.rhtml end end private def capture_domain if

If I use Speex, JSpeex actually, to compress an otherwise valid wave file with zero lengths in the header would it impact the compression at all? Here's what I'm doing during compression in Java: AudioFormat wavFormat = ais.getFormat(); AudioFormat speexFormat = new AudioFormat(SpeexEncoding.SPEEX_Q5, wavFormat.getSampleRate(),

Hi all, I'm newbe with asterisk and i'm facing with this problem that i'm not able to solve. I've to put an asterisk box after an analogic pbx wich require a 0 digit to give the dialtone. So when a client ask asterisk to dial an extension it should 1) send the 0 digit 2) wait for the dialtone 3) dial the extension the client send. How can i obtain this result? Thank's in

Hello All, Yep I'm a newbe. I'm just started to play with asterisk. What I have Redhat Fedora Core 2 (New install) 3 X100P cards. I installed zaptel-1.0.3 libpri-1.0.3 asterisk-1.0.3 Where should I start?? -- Thanks, David -- This message has been scanned for viruses and dangerous content by KE6UPI, and is believed to be clean. KE6UPI thanks MailScanner for their support. Please

--- Jan Engelhardt <jengelh@computergmbh.de> wrote: > Hi, > > > there is an interesting case when the FLAC encoder (using 1.2.0) is > given simple waves. Simple waves means: I have a list of {frequency, > duration, pause} tuples that define the monophonic tune. In other > words, exactly one frequency is played at a time. > > This is the original dataset from

On 02/08/16 10:59 AM, Kolesov, Alexander wrote: > opus_demo -e audio 48000 1 2min_1kHz_Sine_16bit_48kHz.wave > 2min_1kHz_Sine_16bit_48kHz.opus_raw You're missing the "bitrate" argument to opus_demo... Though I'm surprised it didn't complain about it. Jean-Marc

Well I think a sinusoid shouldn't totally trash the encoder state, so I still think theres a bug lurking around in there. The denoiser just prevents it from ever making it to the encoder. Just from browsing through the speex code and from what I've learned reading through the mailing list, it looks like there are a few places where checks are in place to prevent sinusoids from

I wonder if anybody try to compress a pure sine wave using OPUS codec. When I compressed the mono 1KHz, 16bits 48000 samples per sec. audio stream using the 'opus_demo' utility: opus_demo -e audio 48000 1 2min_1kHz_Sine_16bit_48kHz.wave 2min_1kHz_Sine_16bit_48kHz.opus_raw I had the output stream that is shown below. 00 00 00 01 00 00 00 00 78 00 00 00 01 00 00 00 00 78 00 00 00 01 00

I am currently using speex and ogg to archive voice data. The data comes in PCM ulaw at 8kHz and I use a table look up to convert it to normal 16-bit PCM data. Whenver the sound coming in is voice everything works perfectly. However, we periodically run test signals through our system to determine link problems.. etc. This test signal totally hoses speex during playback, but only when you try to

So i'm trying to write an app that enocodes WAV files to "Ogg in a RIFF-WAV" files. Theese are used in Fruity Loops, and since the software itself does not have this feature i wanted to write one myself. The point for it is to save space when sharing "zipped loop packages". Such packages contain the samples (WAV) and the song file (FLP). At first i wrote just an encoder

Hi, there is an interesting case when the FLAC encoder (using 1.2.0) is given simple waves. Simple waves means: I have a list of {frequency, duration, pause} tuples that define the monophonic tune. In other words, exactly one frequency is played at a time. This is the original dataset from 1989 (driving a PC speaker back then): $ ls -l ihold.sd -rw-r--r-- 1 jengelh users 20616 Aug 14 00:57

i'm trying to disconnect idle users from my system by editing /etc/ssh/sshd_config i have set TCPKeepAlive no ClientAliveInterval 2 and restarting sshd services /etc/rc.d/sshd restart but it still wont disconnect any idle client any advice is highly appreciated areadamai freebsd user

On Thu, 2005-01-13 at 12:42 -0500, Jean-Marc Valin wrote: > Le jeudi 13 janvier 2005 ? 10:59 -0500, Jared Whitby a ?crit : > > Interestingly enough.. I started playing around with preprocessing > > options in 1.1.6 and happened upon the denoise filter > > (SPEEX_PREPROCESS_SET_DENOISE). When i run the test tone using that > > option it is completely filtered out and I

The patch I submitted only reads AIFF files. I'm about to start the patch to write AIFF files. To do so, we need a command-line option to specify AIFF. My inclination is to add an option: -ff { raw | wav | aif } In some sense, "-ff" is silly since it probably stands for "format format". Still, I think it's better than just "-f", since the first