similar to: Sound distorted after normalized.

> 16 bit ints have a range of -32768 to 32767. If you divide > -32768 by 32767.0 you end up with -1.00003051850948 which > is a bad thing. > > Try normalizing with a value of 32768.0. No. Speex expects values in the +-32767 range, not +-1.0. Just converting from int16 to float *is* the right thing to do. Jean-Marc -- Jean-Marc Valin <Jean-Marc.Valin@USherbrooke.ca>

Hmmm... sorry i mislead you Tay... i didn't realise my encoder (libfishsound) was again shifting the data i gave it back to being 32767 based internally. Zen. ----- Original Message ----- From: "Jean-Marc Valin" <Jean-Marc.Valin@USherbrooke.ca> To: "Tay YueWeng" <yueweng@yahoo.com> Cc: "speex" <speex-dev@xiph.org> Sent: Wednesday, December

Dear all, I have one simple question. I understand that speex_encode and speex_decode takes float * as an arguement to encode and decode the sound. However, when I get the PCM data from the sound card under win32, it is a just 16 bit array. May I know how do I convert this 16 bit value to speex float format and to convert back? Is there got any routine to do this? YueWeng

I use Speex with dialup modem users. Even if the dialup modem is "56K", you should assume that upstream bandwidth available is only 20-30kbps at the most. I use 16kHz wideband mode and VBR quality 2. Also, I send 80ms (4 frames) per packet, because there is an overhead of approximately 33 bytes per packet due to UDP (8 bytes), IP (20 bytes), and PPP (~5 bytes) headers. If you

Hi, Ok, what you suggest sound logical to me. Currently, I have done a small trick to prevent this problem. What I did is that whenever windows request a voice packet from me and if I do not have the voice packet, I repeat the previous packet. Hence, all the breaking portion is filled with previous packet. This trick seems to work so far. I am not sure what is the side effect. I think jitter

Hi all, I have deploy speex 1.1.6 in my application. With no option set, I can hear that the voice sounds a little bit artificial like robot. Any idea what causes this? I use openh323 with speex, but it seems ok. Is it neccessary for me to use more other filter prior to encode the sound or after decode my sound? yueweng __________________________________ Do you Yahoo!? Yahoo! Mail -

Hi Jean, > > > Well, just trace it, how many times are you > calling > > > speex_decode_int()? > > > > Maximum is 51 times per second. Will this cause > any > > CPU high utilization? > > That's normal... What CPU are you using? If it's a > fixed-point CPU, then > the reason is probably just the fact that the packet > loss

Hi all! I've problem with playing ogg files. I'm triing to use tripple buffer method: +-------------+ +-| buffer 1 |-+ | +-------------+ | | | | +-------------+-+ | | buffer 2 | | +-------------+-+ | | | +-------------+-+

Hi, Have anyone using speex experience the following effect: When I say "hello", it will be become "hellololololololo....". This does not always happened. The chance of happen is quite random. This effect does not happen in my LAN. It only happen when ping time is about 500 milisecond or more. Is it a speex problem? Any suggestion how to resolve this issue? Thanks in

Dear all, I am using speex 1.17 at this moment, everything works great. However, I face a problem when no packet arrived from network for a few second, my CPU usage is 100%. I step though my code and seems that (not confirmed) the speaker callback WaveOutCallback() function which call speex_decode_int(decoder_state, NULL, shortData) (when no data arrived for PLC purpose) seems to consume a

Hi, > 1) Normal given the bandwidth/bit-rate used Do you mean the bit-rate that I should set in the speex codec? > 2) A conditioning problem with your audio (i.e. DC > not removed) What is DC? YueWeng __________________________________ Do you Yahoo!? Yahoo! Mail - now with 250MB free storage. Learn more. http://info.mail.yahoo.com/mail_250

Hi Ingo, >It is just a WAVEFORMATEX with a GUID in it. For the ACM system it is just a format with the id >WAVE_FORMAT_EXTENSIBLE and since ACM drivers are not registered by wFormatTag it should be working. >For testing purposes I've written a Vorbis ACM driver, that uses a WAVEFORMATEXTENSIBLE and I experienced >a problem when opening files with Media Player. It spend quite some

> Ok, what you suggest sound logical to me. Currently, I > have done a small trick to prevent this problem. What > I did is that whenever windows request a voice packet > from me and if I do not have the voice packet, I > repeat the previous packet. Hence, all the breaking > portion is filled with previous packet. This trick > seems to work so far. I am not sure what is the

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

On 2019-04-01 3:37, Gustaf Ullberg wrote: > Hi everyone, > > Some time ago, I sent a pull request > <https://github.com/xiph/opus/pull/107> to the Opus github page. > Jean-Marc asked me to post it to the mailing list so everyone can have a > look at it. > > You can find the description and code changes below. Please let me know > if you have any questions or

Hi, While I browsing the source of the Oggenc utility, I saw a comment says: /* A common error is to have an 18-byte format chunk with the last two * bytes 0. This is incorrect, but sufficiently common that we only warn * about it instead of refusing it. * Please, if you have a program that's creating these 18 byte chunks, send * a bug report to whoever makes it */ And it was

Thank you Mark. I agree and have now updated the pull request with a new commit, addressing your comments. Please take a look. /Gustaf On Fri, 5 Apr 2019 at 11:41, Mark Harris <mark.hsj at gmail.com> wrote: > On 2019-04-01 3:37, Gustaf Ullberg wrote: > > Hi everyone, > > > > Some time ago, I sent a pull request > > <https://github.com/xiph/opus/pull/107>

Hi, I am using the speex 1.0.4 library from Windows. I have posted my problem before but didn't get a solution. I am doing an VOIP project in which i am recording sound and streaming it to the peer. I wanted to encode and decode wav files that brought me to this site. I am recording sound in the following format:- m_WaveFormatEx.wFormatTag = WAVE_FORMAT_PCM;

Hi everyone, Some time ago, I sent a pull request <https://github.com/xiph/opus/pull/107> to the Opus github page. Jean-Marc asked me to post it to the mailing list so everyone can have a look at it. You can find the description and code changes below. Please let me know if you have any questions or concerns. Best regards Gustaf Ullberg In WebRTC, we would like to be able to

I'm running a: for band in n w u do for quality in 0 1 2 3 4 5 6 7 8 9 10 do for complexity in 3 do for vad in "" "--vad" do for dtx in "" "--dtx" do echo speexenc -${band} --quality ${quality} --comp ${complexity} ${vad} ${dtx} input.${band}.wav