search for: pcm

...uffer is flushed I guess). Did anyone enountered similar problem? I guess there is something wrong with audio format/samplerate conversion...? Best regards, Tomek More information: emu10kx0: <Creative Audigy 2 ZS [SB0350]> port 0xdc00-0xdc3f irq 21 at device 6.0 on pci4 % sysctl -a | grep pcm dev.pcm.0.%desc: EMU10Kx DSP front PCM interface dev.pcm.0.%driver: pcm dev.pcm.0.%parent: emu10kx0 dev.pcm.0.eapd: 1 dev.pcm.0.play.vchans: 1 dev.pcm.0.play.vchanmode: fixed dev.pcm.0.play.vchanrate: 48000 dev.pcm.0.play.vchanformat: s16le:2.0 dev.pcm.0.rec.vchans: 1 dev.pcm.0.rec.vchanmode: fixed...

...I have on my system (merlin2, merlin3, etc) but I've had no luck . I'm running Asterisk 1.2.7.1. The 7960 phones have 7.4 SIP firmware loaded on them Here is what is in my RINGLIST.DAT file: R2D2 r2d2.raw Meow meow.raw Galaga galaga.raw Ahh! ahh.pcm Doh! doh.pcm Old Style ringer1.pcm Synth Low ringer2.pcm Dungeon ringer3.pcm Lightbulb ringer4.pcm Synth High ringer6.pcm Are You There M AreYouThere.raw Are You There F AreYouThereF.raw ClockShop ClockShop.raw Curley Curley.raw Drums 1...

...cted Time we started stream on 82.67.34.81:443/radio-bro-gwened.ogg [2003-11-25 19:53:17] INFO om_shout/check_shout_connected Connected to server: 82.67.34.81:443/radio-bro-gwened.ogg [2003-11-25 19:53:17] DBUG om_shout/output_ogg_shout initialising output stream [2003-11-26 00:00:01] INFO input-pcm/pcm_read No more data available [2003-11-26 00:00:01] INFO input/input_loop Closing input module [2003-11-26 00:00:01] INFO input-pcm/pcm_close_module Closing PCM input module [2003-11-26 00:00:01] DBUG input/open_next_input_module checking module 1 <p>... [2003-11-26 00:00:01] WARN inp...

I really should have just said that it will require some testing to make sure the FLAC API can handle writing the same file from multiple threads. It may not turn out to be complicated at all. The FLAC decoder has its own code for writing PCM files already. Tweaking this to support multi-threaded writes would not be too difficult. It's simply a matter of building the PCM file, setting aside the data area for the PCM samples, and then providing each thread with information needed to write at different offsets. FLAC never...

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 need 8-bit linear PCM then a simple solution would be to use only the top 8 bits of each 16-bit sample from opus_decode(). Note that the WAV format uses unsigned rather than signed...

...ing two different soundcards (as I understand) for the > playback and the capture, you're *never* going to get echo canceller to > work because the playback and capture need to be *exactly* in sync. I did a bit more testing, and ended up creating a set of testsamples, as follows: ideal.pcm => me saying "This is what I'd like to hear" junk.pcm => me saying "With a bit of luck, this is gone" mixdown.pcm => ideal.pcm, with junk.pcm started 7 ms later and 3 dB lower. (All of this as 8000 Hz, 16-bit signed) I then copied junk.pcm to play.sw, mixdown.pcm...

...10:45:50] INFO ices-core/main libshout version 2.0-kh22 [2003-11-25 10:45:50] INFO ices-core/main realtime scheduling has been enabled [2003-11-25 10:45:50] DBUG stream/start_runners starting runner [2003-11-25 10:45:50] INFO stream/ices_runner Runner 1 ready [2003-11-25 10:45:51] INFO input-pcm/pcm_initialise_module Initialising PCM input module [2003-11-25 10:45:51] DBUG input-pcm/pcm_initialise_module Module PCM using buffers of 4096 samples [2003-11-25 10:45:51] DBUG input/initialise_input_modules Module 1 (pcm) has 20 buffers [2003-11-25 10:45:51] DBUG input/open_next_input_module...

...-1 samba-common-3.0.10-1.4E samba-swat-3.0.10-1.4E samba-3.0.10-1.4E samba-client-3.0.10-1.4E smb.conf: [root@itbsjb1 samba]# cat smb.conf # Samba config file created using SWAT # from 127.0.0.1 (127.0.0.1) # Date: 2005/09/30 15:27:17 # Global parameters [global] workgroup = PCM realm = PCM.PUK.AC.ZA server string = ITBSJB se SAMBA security = ADS password server = dc1-nt.pcm.puk.ac.za db-win1.pcm.puk.ac.za log file = /var/log/samba/%m.log max log size = 50 socket options = TCP_NODELAY SO_RCVBUF=8192 SO_SNDBUF=8...

...e report at bugs.freedesktop.org Mesa 7.11-devel implementation error: bad format in do_row() Please report at bugs.freedesktop.org Warning: detected OpenGL error 'out of memory' after RenderBin::draw(,) RenderStage::drawInner(,) FBO status= 0x8cd5 creating 3D noise texture... DONE ALSA lib pcm.c:2212:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear ALSA lib pcm.c:2212:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe ALSA lib pcm.c:2212:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side ALSA lib pcm.c:2212:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.surround71 ALSA lib pcm.c:2...

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! fixme:iphlpapi:NotifyAddrChange (Handle 0x7d0049e8, overlapped 0x7d0049cc): stub fixme:shell:DllCanUnloadNow stub wine: configuration in '/home/bi...

...; <byrnejb at harte-lyne.ca> Date: Thu, May 16, 2013 16:54 To: users at jitsi.org ---------------------------------------------------------------------- I installed jitsi-2.2 on a CentOS-6.4 x86_64 host. When I run it as a non-privileged user then I see this error: $ jitsi ALSA lib pcm.c:2209:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear ALSA lib pcm.c:2209:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe ALSA lib pcm.c:2209:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side ALSA lib pcm_dmix.c:957:(snd_pcm_dmix_open) The dmix plugin supports only playback stream GT...

hello, can I get out of wine more than 16 bit output? sound_file -> player -> wine_out, which is 24/32 to linux? somethink like that, I am audiophile :( thank you

...it I use pc8bitSamples[ui32Loop] = ((unsigned short)pcop16OutBuf[ui32Loop] ^ 0x80); Regards Amit On Sat, Jan 9, 2016 at 2:39 PM, Amit Ashara <ashara.amit at gmail.com> wrote: > 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 w...

I use Wine for play World of Warcraft, that support on Windows, 6 channel sounds mixing software. On Ubuntu 8.04 and Wine i get only stereo output. I've tried to do this .asoundrc for "force" use on Wine, of 6channels and I don't get any sound. Code: pcm.!default { type dmix ipc_key 1024 ipc_key_add_uid false # let multiple users share ipc_perm 0660 # IPC permissions (octal, default 0600) slave { pcm "hw:0,0" # see below rate 48000 channels 6 period_time 0 period_size 1024 buffer_time 0 buffer_size 4096 } } pcm.asym51...

Hello, I will take your last version, but I have fresher news about the bug that occurs every mornings in fact... I think this is ntpdate which is causing the problem, as the time of the machine change for more than one second. Here is all the info: 09:46:17 up 1 day, 1:22, 2 users, load average: 0.60, 0.71, 0.68 86 processes: 84 sleeping, 2 running, 0 zombie, 0 stopped CPU states: 10.4%

...to WAVE size limitations */ -static FLAC__byte buffer[READSIZE/*samples*/ * 2/*bytes_per_sample*/ * 2/*channels*/]; /* we read the WAVE data into here */ +static FLAC__byte buffer[READSIZE/*samples*/ * 3/*bytes_per_sample*/ * 2/*channels*/]; /* we read the WAVE data into here */ static FLAC__int32 pcm[READSIZE/*samples*/ * 2/*channels*/]; int main(int argc, char *argv[]) @@ -71,13 +71,15 @@ memcmp(buffer+8, "WAVEfmt \020\000\000\000\001\000\002\000", 16) || memcmp(buffer+32, "\004\000\020\000data", 8) ) { + /* Ignoring for now as I work towards supporting 24bps files....

Hi all, i've to encode voice from a voicemodem. I choose speex 1.0.5 for its quality in voice encoding. I've tried to implement an encoder but unsuccesfully. Here's my code: /* ============ SPEEX stream ENCODER ============================================ */ int SPEEX_EncodePCM(struct _IDA_ClientSocket *IDA,char *buffer,unsigned char *PCM,int num_samples) { /* buffer point to the output buffer PCM point to the numsamples bytes 8bit - 8Khz PCM from the modem num_samples is, commonly, 16000 */ int ret,c,d; short int PCM_F[160]; for(c=0;c<num_samples;c++) {...

...VE size limitations */ -static FLAC__byte buffer[READSIZE/*samples*/ * 2/*bytes_per_sample*/ * 2/*channels*/]; /* we read the WAVE data into here */ +static FLAC__byte buffer[READSIZE/*samples*/ * BPS/8 /*bytes_per_sample*/ * 2/*channels*/]; /* we read the WAVE data into here */ static FLAC__int32 pcm[READSIZE/*samples*/ * 2/*channels*/]; int main(int argc, char *argv[]) @@ -73,14 +75,18 @@ int main(int argc, char *argv[]) memcmp(buffer+8, "WAVEfmt \020\000\000\000\001\000\002\000", 16) || memcmp(buffer+32, "\004\000\020\000data", 8) ) { +#if BPS == 16 fprintf(stder...

Hello Ralph, > Likewise opus_encode() takes 16 bit samples, so you need to extend each > sample from an 8 bit source before encoding. Two questions 1. In opusenc.c which API does the extending the 8-bit to 16-bit? 2. If that is the case then how will 24 bit PCM sample work? Regards Amit On Thu, Jan 7, 2016 at 12:21 PM, Ralph Giles <giles at thaumas.net> wrote: > 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...

Dear all, I'm working on ability estimates using Rasch model. Using the "ltm" package, the procedure is quite simple: ## Factor Scores for the Rasch model fit <- rasch(LSAT) factor.scores(fit) What about Partial Credit Model (PCM)? For PCM I use PCM function from eRm package. Is there any similar function like factor.scores to estimate ability scores using PCM model? Best, Andrej