search for: 44100hz

Hi, Is there any difference between decoding of CBR or VBR streams in realtime in terms of CPU usage/cost? Are sample rates other than 44100Hz supported by Vorbis officially ? If it is supported, is it efficient to use them or is Vorbis specifically (or at the moment) optimized for 44100Hz ? Thanks in advance. --- Mete BALCI Senior Game Programmer Momentum DMT Istanbul, TURKEY ----------------------------------------------------------...

opusdec -V opusdec opus-tools f2a2e88 (using libopus unknown) I've got an opus file encoded from a .wav off a cd, 44100Hz: opusinfo 2-24-Overture_in_C_\(In_Memoriam\).opus Processing file "2-24-Overture_in_C_(In_Memoriam).opus"... New logical stream (#1, serial: 38134f1f): type opus Encoded with libopus unknown User comments section follows... ENCODER=opusenc from opus-tools f2a2e88 ENCODER_OPTIONS=--vbr -...

...function call is used by the JUCE ogg decoder we are using in an application. Any thoughts? thanks in advance, Jim. The output is: jim at snoopy ~/trouble.vorbis/a $ ./a < 53884c2935e6b47938f91ca78b0c2473a85730833138370caedbb66a74995130.ogg encoder=Lavf53.32.100 Bitstream is 2 channel, 44100Hz Encoded by: Lavf53.32.100 Length: 0 jim at snoopy ~/trouble.vorbis/a $ ./a < d7d5ef8e6c7fad33f922924d0a05591277654d53941d43518f1fe8580c7ea3f9.ogg encoder=Lavf53.32.100 Bitstream is 2 channel, 44100Hz Encoded by: Lavf53.32.100 Length: 1146624 Source: #include <stdio.h> #include <...

...function call is used by the JUCE ogg decoder we are using in an application. Any thoughts? thanks in advance, Jim. The output is: jim at snoopy ~/trouble.vorbis/a $ ./a < 53884c2935e6b47938f91ca78b0c2473a85730833138370caedbb66a74995130.ogg encoder=Lavf53.32.100 Bitstream is 2 channel, 44100Hz Encoded by: Lavf53.32.100 Length: 0 jim at snoopy ~/trouble.vorbis/a $ ./a < d7d5ef8e6c7fad33f922924d0a05591277654d53941d43518f1fe8580c7ea3f9.ogg encoder=Lavf53.32.100 Bitstream is 2 channel, 44100Hz Encoded by: Lavf53.32.100 Length: 1146624 Source: #include <stdio.h> #include <...

...you'll need to make /system read-write and copy lib/libshine.so.1.0.0 to /system/lib/libshine.so.1 and bin/shineenc to /system/bin. On my low-end HTC, however, encoding is not real-time, which is a bit disappointing: $ shineenc bla.wav bla.mp3 shineenc (Liquidsoap version) WAV PCM DATA, stereo 44100Hz 16bit, Length: 0: 0:18 MPEG-I layer III, stereo Psychoacoustic Model: Shine Bitrate=128 kbps De-emphasis: none Original Encoding "bla.wav" to "bla.mp3" Finished in 0: 0:20 If anyone else feels like testing on another ARM android phone, I'll be interested to know if t...

https://bugs.freedesktop.org/show_bug.cgi?id=111213 Bug ID: 111213 Summary: VA-API nouveau SIGSEGV and asserts Product: Mesa Version: 19.0 Hardware: x86-64 (AMD64) OS: Linux (All) Status: NEW Severity: normal Priority: medium Component: Drivers/DRI/nouveau Assignee: nouveau at

I found the details: 44100hz, 16bits, stereo I am looking at rewriting the program to record at 8Khz/16Khz/32Khz mono only should I record at 44100hz (and convert down) or record at the required hz level? ----- Original Message ----- From: <speex-dev-request@xiph.org> To: <speex-dev@xiph.org> Sent: Thursday, J...

...by the file format, which is IMHO the interesting number. The file format itself allows block lengths between 1 and 32768 samples (in powers of two), but this is restricted in Vorbis I to be between 64 and 8192. The encoder has a minimum latency equal to the block length. With a sample rate of 44100Hz, this is between 1.5 and 186ms. Due to block overlapping, the minimum possible latency in the decoder would be 1.5 * max block length. Using only blocks with 64 samples, you have the "worst case" when computing the 33rd sample of each block, for which you need the following block to ha...

Hi, I use speex 1.1.12 in my program, but I found a strange thing about Echo canceller. That is, I'll hear an explosion in a certain frequency inputing. And I use the echo canceller in 44100Hz sample rate, disable AGC and Denoise. Can you tell me what's wrong with me? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20060412/b6a628cf/attachment.html

...ytes=10000k /dev/zero > ! zero.raw snotfish% ./encoder_example < zero.raw > zero.ogg snotfish% chaining_example < zero.ogg Input bitstream contained 1 logical bitstream section(s). Total bitstream playing time: 58 seconds logical bitstream section 1 information: 44100Hz 2 channels bitrate 0kbps serial number=146195353 compressed length: 17485 bytes play time: 58s Monty --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/

In beta 3 (Windows, libvorbis-1.0beta3.zip, compiled using BCB), I get a runtime overflow on line 322 when encoding a simple pure tone (44100Hz, 96kbps, 128kbps, or 160kbps). Changing the float to a double avoids the problem. float val=decay[i]*decscale; float att=fabs(f[i]/val); /// floating point overflow ... Chris http://www.goldwave.com --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http:/...

......] options: -P 0 -b 4608 -m -l 8 -q 0 -r 3,3 -R 0 -V test: 101% complete, ratio=0.078 Which is clearly wrong (101%), and I think should probably trigger an error (but doesn't). Based on a few quick tests, this kind of problem seems to occur with any file smaller than 4k (at least when using 44100Hz/16bps/2chan). -- - mdz

...t;> thanks in advance, Jim. >> >> >> >> The output is: >> >> jim at snoopy ~/trouble.vorbis/a >> $ ./a < 53884c2935e6b47938f91ca78b0c2473a85730833138370caedbb66a74995130.ogg >> encoder=Lavf53.32.100 >> >> Bitstream is 2 channel, 44100Hz >> Encoded by: Lavf53.32.100 >> >> Length: 0 >> >> jim at snoopy ~/trouble.vorbis/a >> $ ./a < d7d5ef8e6c7fad33f922924d0a05591277654d53941d43518f1fe8580c7ea3f9.ogg >> encoder=Lavf53.32.100 >> >> Bitstream is 2 channel, 44100Hz >> Enc...

On Sat, 7 Jun 2003, Karl Heyes wrote: > your sound card/driver does not deal with 44100hz samplerates, try 48000 > instead. Remember to change the resample tags according. Are there many cards in this category? I'dve thought that all cards would support 44100, it being the standard that it is. Your comments however suggest experience of this problem. Geoff. --- >8 ---- Li...

...9;flac' decoder could decode > the file just fine, and unless I'm mistaken, that uses the file > decoder also. > > The only odd thing about the input wave file was that it was 8bit mono > (22050Hz). oN a hunch, I converted the wave file to CD audio format > (16bit/stereo/44100Hz) with sox, and re-encoded it. The plugin played the > resulting file fine. I tried a few tests (converting to stereo and leaving > everything else the same, converting to 16 bit and leaving everything else the > same), and in every case the flac file wouldn't play. > > The doc...

...; decoder could > decode > the file just fine, and unless I'm mistaken, that uses the file > decoder also. > > The only odd thing about the input wave file was that it was 8bit > mono > (22050Hz). oN a hunch, I converted the wave file to CD audio format > (16bit/stereo/44100Hz) with sox, and re-encoded it. The plugin > played the > resulting file fine. I tried a few tests (converting to stereo and > leaving > everything else the same, converting to 16 bit and leaving everything > else the > same), and in every case the flac file wouldn't play. &gt...

...lication and the FLAC frontend (although using the same settings). for example: FLAC frontend (quality = 8) -------------------------------- FLAC_vendor = reference libFLAC 1.1.1 20041001 bitrate = 1047 samplerate = 44100 channels = 2 bitspersample = 16 codec = FLAC ---------- 4831008 samples @ 44100Hz File size: 14 334 137 bytes My Application (quality = 8) ------------------------------- FLAC_vendor = reference libFLAC 1.1.1 20041001 bitrate = 1314 samplerate = 44100 channels = 2 bitspersample = 16 codec = FLAC ---------- 4831008 samples @ 44100Hz File size: 17 995 449 bytes >Hello all...

...= I could not see a significant difference in foobar's (version 1.3.2 before = you ask) display. Is this the way things should be? The other thing that confused me is that = foobar displays 48kHz for the sampling rate in file properties (a bug?). = Opusinfo displays "Original sample rate: 44100Hz". Here are the essential data of my sample: User comments section follows... ENCODER=3Dopusenc from opus-tools 0.1.9 ENCODER_OPTIONS=3D--bitrate 256 --vbr --comp 10 --ignorelength title=3DCD Track 04 tracknumber=3D4 TRACKTOTAL=3D16 Opus stream 1:...

...ication. > > Any thoughts? > > thanks in advance, Jim. > > > > The output is: > > jim at snoopy ~/trouble.vorbis/a > $ ./a < 53884c2935e6b47938f91ca78b0c2473a85730833138370caedbb66a74995130.ogg > encoder=Lavf53.32.100 > > Bitstream is 2 channel, 44100Hz > Encoded by: Lavf53.32.100 > > Length: 0 > > jim at snoopy ~/trouble.vorbis/a > $ ./a < d7d5ef8e6c7fad33f922924d0a05591277654d53941d43518f1fe8580c7ea3f9.ogg > encoder=Lavf53.32.100 > > Bitstream is 2 channel, 44100Hz > Encoded by: Lavf53.32.100 > > Length...

Thanks all 5000 128kbit 44100Hz mp3/ogg vorbis streams 50 sources 128kbit 44100Hz from other hardware send via dedicated network card to the icecast server The network traffic not be a problem I chose the server that can fit? ---------- Messaggio inoltrato ---------- Da: "Michael Smith" <msmith at xiph.org> Da...