search for: 8000hz

Hi We're getting requests coming in for higher quality audio in our call recordings. We currently use MixMonitor and everything is being saved in it's native 8000Hz, 16 bit wav format. I have seen information on using Monitor and specifying a conversion to mp3 when the call ends and the 2 channels get mixed but surely the 2 channels are already saved as 16bit 8000Hz wav files so the quality is lost already? Is there any way of making high quality recordings...

Oops, CTRL+Enter send strikes again ... At the other end for playback you can convert it back to 48000 (or whatever) by repeating each sample 3 times (48/16 == 3), then running a 8000Hz lowpass over the result to remove any aliasing artifacts. Cheers, David Hogan > -----Original Message----- > From: David Hogan > Sent: Tuesday, 12 December 2006 10:44 AM > To: 'speex-dev@xiph.org' > Subject: RE: [Speex-dev] Sampling Rate > > Hi, > > I'm n...

Hi, I'm no DSP or audio expert by any means, but I can share what works for me. People in the know, I would appreciate tips on whether this stuff is ok. You could sample at 32000Hz (or 48000Hz, any AC97 card will support this), run a 8000Hz lowpass filter over the data (16000Hz sample rate can only represent frequencies up to 8000Hz) and then drop every second (or 2 out of 3 for 48000->16000) sample. The result being, 16000Hz sampled audio. If you omit the filter the result will conta...

...logical stream (#1, serial: 00003419): type opus Encoded with libopus 1.1 User comments section follows... ENCODER=opusenc from opus-tools 0.1.8 Opus stream 1: Pre-skip: 312 Playback gain: 0 dB Channels: 2 Original sample rate: 8000Hz Packet duration: 20.0ms (max), 20.0ms (avg), 20.0ms (min) Page duration: 1000.0ms (max), 989.6ms (avg), 480.0ms (min) Total data length: 247227 bytes (overhead: 1.89%) Playback length: 0m:49.471s Average bitrate: 39.98 kb/s, w/o...

...data conversion tip too. Reed, DirectSound uses void* as a datatype. In general it takes bytes (for playback), but when in 16-bit mode it expects shorts for capturing packets. It accepts ranges from -128 -> 127 (8-bit mode) or -32768 -> 32767 (16-bit mode).Currently I'm recording mono 8000Hz at 16-bits/sample and I'm expecting the decoder to produce the same. Default settings show it operates with 1.875KB/sec bitrate, 8000Hz sample, and 16-bits/sample. Seems appropriate. Nate > ----- Original Message ----- > From: "Nathaniel Meyer" <nath_meyer@hotmail.com>...

...EC algorithms to be used in most computer. Maybe it can be summaried as follows: 1. Different sample rate of sampling and rendering does exists in most low-cost soundcards (In my experiments over more than 20 soundcards, the differences range from 0.5Hz to more than 50Hz when sample rate is set to 8000Hz). Maybe this is totally caused by hardware which can't be solved by software settings. 2. Static measurement of the difference between sample rates is far from enough. Accurater measurement requires more time to record echo signal in order to get accurater frequency shift from spectral structu...

Hi everyone, I'm about to release version 1.2-beta1 (which I could have called 1.1.13), which includes many, many changes. It would help if everyone could give the svn version (http://svn.xiph.org/trunk/speex/) a try and see if it works fine. I'll check my email next week when I'm back from some vacations and if nothing bad has been reported, I'll make the release. Have fun,

Hi, I am very fresh man at CODEC part.. I wanna know SPEEX's compression rate.. to say , I tested wav file at 8000 Hz, 8bits to spx file.. I hoped 1/16 compression rate more.. but,, the result was not good for me.. to say example.. I made wav file of 254Kb at 8000Hz , 8bits ...and then encode that file.. the result was that spx file of 33.7Kb is made... so, I though that compression rate is 33.7 / 254 .. about 1/8.. and,,, there was noise in decoded file... of course,, that result may be wrong because of my wrong method... could you let me know the best comp...

Hi, I am trying to encode raw wave data stored in a buffer using the Speex API (The raw wave data is created using the waveIn* functions - probably irrelevant information here). It is a 5 second clip, 16bits/sample, 8000Hz mono (which gives a buffer size of 80kb for the wave data). I have followed the exact procedure found in the manual available from the web site, except that instead of writing the speex stream into stdout, I am writing it onto a file from where I read later to decode. It decodes to 80kb...

...gt; Shri. > > 20 ms is usually a good idea, but you can use shorter (I don't recommend > much longer). In any case, just try it and see if it works for you. > > Jean-Marc > Hi Jean-Marc, I am using speex-1.1.12 version AEC with the following configuration SAMPLE RATE : 8000Hz FRAME LENGTH: 32 , i.e. 4msec(125usec*32) TAIL LENGTH : 256 , 32msec I have configured AEC for ARM-920T, 200MHz Here are my observations [a] AEC CPU usage 50% [b] Still 10%trace of echo is still present NOTE: I am not using preprocessor as my cpu usage goes upto 99% What should i do to increase t...

Hi. I trying to encode PCM16 8000Hz stereo data to speex and put it into the .flv file format. But at the output I can hear only noise. What I doing wrong? Here is the code: void main() { SpeexBits bits; void *enc_state; int frame_size; int quality = 10; char cbits[MAX_FRAME_BYTES]; FILE *fin, *speex; sh...

Hello All, I have several soundfiles that are recorded ub 44100Hz, 16-bit Mono. What is the best way and right tools to use to downsample these to 8000Hz so that they can be used with Asterisk. I've tried using sox with the -r switch and Audacity on the mac and Goldwave on Windows and they all generate files that sound like a bad acid trip. I tried increasing the speed 551.25 percent after doing the resample on these files and then it sounds lik...

...which version you want to support (though supporting both > wouldn't be really hard). Will do stable first, and I agree it will be easy to support both when 1.2 is stable. So, the floats are basically PCM, eh? For VOCAL's g.711 codec I've been pre-converting files to 8-bit pcm at 8000hz with sox. So I can just not down-sample the wave files so badly, and let Speex use the 16-bit pcm values converted to floats? I'll dig into the RTP rfc and see how that goes. Thanks! Ben -- Ben Greear <greearb@candelatech.com> Candela Technologies Inc http://www.candelatech.com &l...

...llo, I have figured out that my audio problem was just how I was converting the sound files. I am trying to convert the Asterisk gsm files to ULAW. I just did a: sox file.gsm file.ul, open it in Audacity. I used: Project, Import Raw, U-law, No endian, 1 channel, start offest 1 byte, sample rate 8000hz. The file sounds fine in Audacity. Now, if I do a record on Asterisk, using pcm, au, or ul, I get pops in the audio. What am I doing wrong? Also, what types of wav (not WAV) files does Asterisk generate? I have a couple wav files to convert too. I believe if I do a sox file.wav -r 8000 -c 1 f...

In sox terms is SLIN .ul (as in unsigned linear). Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN steve@gbnet.net Euro Tech News Blog http://eurotechnews.blogspot.com

...exten => 64,2,MusicOnHold(default) exten => 64,3,Hangup I have converted my mp3 files so that they have the following characteristics )but I don't really think this matters since when the music is on the Linux machine, without slim server, it works: MPEG 2.5 layer 3 16kbit, 1385 frames 8000Hz Mono Any help would be really appreciated!!! Thank you Dov -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050323/4b9f29a1/attachment.htm

Hi, I have a module sampling raw PCM data on Win CE as 10ms time slice (160 bytes), mono, 8000HZ, 16 bits per sample. Does anyone know what is the mflops for using fixed point on a Win CE compared to using floating point? Looking at the manual, "In practice, frame_size will correspond to 20 ms when using 8, 16, or 32 kHz sampling rate." for a 8 kHz sampling, the framesize should...

...most computer. > Maybe it can be summaried as follows: > 1. Different sample rate of sampling and rendering does exists in most > low-cost soundcards (In my experiments over more than 20 soundcards, > the differences range from 0.5Hz to more than 50Hz when sample rate is > set to 8000Hz). Maybe this is totally caused by hardware which can't > be solved by software settings. > 2. Static measurement of the difference between sample rates is far > from enough. Accurater measurement requires more time to record echo > signal in order to get accurater frequency shift...

...gh for realtime encoding - the Speex encode function takes 20+ ms to encode one 20ms sample frame without compiler optimizations, but with compiler optimizations and tracing removed, only on occaisional tests did I notice gaps in the stream. My experiments were done with 1 channel, 16 bit, 8000hz samples, using Speex quality 3 and complexity 4. <p><p>--- >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 wo...

...sterisk 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, 8000Hz according to SoX and confirmed by the sox mailinglist (http://sourceforge.net/mailarchive/message.php?msg_name=4C6A4661.9000209%40telenet.be) [root at asterisk testing]# soxi testLong2.wav Input File : 'testLong2.wav' Channels : 1 Sample Rate : 8000 Precision : 16-bit...