search for: 96kb

Sorry for the crosspost but it's relevant to Vorbis and Icecast I believe. I'm seeing more and more streaming stations using AACplus, with many listeners being amazed at the sound quality. Most say that a 48kb/s sounds better than a 128kb/s MP3, which would put Ogg Vorbis at around 96kb/s IMO. That means only half the bitrate is required in AACplus compared to Ogg Vorbis for the same sound quality. Up until this codec was available, Ogg Vorbis compared favourably with all the others. Is there anything that can be done to bring Ogg Vorbis up to this type of quality in the fut...

Sorry for the crosspost but it's relevant to Vorbis and Icecast I believe. I'm seeing more and more streaming stations using AACplus, with many listeners being amazed at the sound quality. Most say that a 48kb/s sounds better than a 128kb/s MP3, which would put Ogg Vorbis at around 96kb/s IMO. That means only half the bitrate is required in AACplus compared to Ogg Vorbis for the same sound quality. Up until this codec was available, Ogg Vorbis compared favourably with all the others. Is there anything that can be done to bring Ogg Vorbis up to this type of quality in the fut...

Hi, I have just begin to look into Speex.Speex encodes and decodes fine under SH platform (96MHz). But encoding is taking lot of time i,e for 20ms frame it is taking ~890ms to encode it. Which is very high [To encode 96KB data it takes around 4.5min.] Whether is this architecture suitable for speex ? Whether writing some assembly instructions and some optimization will bring down the encoding time to very low? Which part of the code I need to write with the assembly or DSP instructions? Thanks, Vinay

...that has been mentioned in the readme-TIDSP. the code runs perfectly ( i think) but it takes a lot of time for the *.out file to "run". the output file, which is supposed to be "maleout5x" has a size much bigger than the input file "male.snd" (the input file is around 96kb). is the program running correctly??? m desperate for the solution. thanking you nikhil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20080220/2c7d5f4a/attachment.htm

...e CELT codec by using the 0.8.0 testcelt tool to encode and decode the input. I want to test different bitrates and selected the below parameters for 64, 96, 128, 196, 256kB: ./celt-0.8.0/libcelt/testcelt.exe 44100 2 256 46 $1.sw $1-64kb.sw ./celt-0.8.0/libcelt/testcelt.exe 44100 2 192 46 $1.sw $1-96kb.sw ./celt-0.8.0/libcelt/testcelt.exe 44100 2 128 46 $1.sw $1-128kb.sw ./celt-0.8.0/libcelt/testcelt.exe 44100 2 96 46 $1.sw $1-196kb.sw ./celt-0.8.0/libcelt/testcelt.exe 44100 2 64 46 $1.sw $1-256kb.sw can someone comment on the chosen parameters? I am not sure if these are right settings for fr...

...2, 44100, -1, 96000, 96000); to initialize the encoding. To my surprise, it seems the maxbitrate value of 96000 doesn't have an effect, the bitrate of the generated vorbis audio hovers around 96kb/s, going up to 105 kb/s every once in a while. If I call vorbis_encode_init( &vorbisInfo, 2, 44100, 96000, 96000, 96000); than the bitrate will vary between 83 and 98 kb/s approx. (The da...

> No comments? Mostly no. There are no presets because the encoder is smart enough to act intelligently by default. The exposed knobs are exposed only for those who know they actually need them. This is pretty much how Xiph encoders have always been. The programming API is a bit different, because there are more use cases for streaming. Monty

Hi everyone, I have profiled Opus on AArch64. I just run opus_demo with some pcm files. Following is time proportion of FFT with different bitrate. Bitrate | Time cost by FFT/iFFT 24kb/s | 15% 48kb/s | 15% 96kb/s | 13% Any comment? I want some data close to real application, any suggestion? Thanks, Phil Wang -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/opus/attachments/20141125/ae6c1c8c/attachment.htm

...l vorbis_encode_init()? For rc2, it worked calling it the following way: vorbis_encode_init( &vorbisInfo, 2, // two channels 44100, // 44.1kHz -1, 96000, // 96kb/s -1 ); And this produced an Ogg Vorbis stream of about 80 kbps on average. But in rc3, the above call produces a circa 240 kbps Ogg Vorbis stream (as reported by xmms). I tried putting 96000 in place of the -1 values, and also putting 96 instread of 96000, but no luck. An...

Hello. I have same problem, and I think that it's because AAC+ uses VBR and when it's changes, is not the same bitrate, because varies... El jue., 9 may. 2019 a las 21:29, Michel van Dop (<mvandop at xs4all.nl>) escribió: > Hi Paul, > > Thank you for the information. I already try a player to discover the > stream type of both streams if the are the same. > But i can

...el_stack:0kB pagetables:0kB unstable:0kB bounce:0kB writeback_tmp:0kB pages_scanned:0 all_unreclaimable? no lowmem_reserve[]: 0 1464 1936 1936 Node 0 DMA32 free:1122012kB min:4240kB low:5300kB high:6360kB active_anon:23324kB inactive_anon:8kB active_file:15332kB inactive_file:230356kB unevictable:4496kB isolated(anon):0kB isolated(file):0kB present:1499360kB mlocked:4496kB dirty:396kB writeback:0kB mapped:13448kB shmem:144kB slab_reclaimable:19140kB slab_unreclaimable:9520kB kernel_stack:984kB pagetables:4116kB unstable:0kB bounce:0kB writeback_tmp:0kB pages_scanned:0 all_unreclaimable? no lowmem_...

I'm trying to send out multiple streams. My encoder is a PII 233, so I have to conserve CPU cycles. I encoded my mp3s as 96kb/s with a 44.1 sampling rate. I send my 96k stream without reencoding and it sounds great. I also sned a 24k stream that is being reencoded from the 96k mp3s. It works pretty good, the load on my linux box is about 0.5 on average. However, the sampling rate is 8k. I'd rather bump that up to 2...

...t;For rc2, it worked calling it the following way: > >vorbis_encode_init( &vorbisInfo, > 2, // two channels > 44100, // 44.1kHz > -1, > 96000, // 96kb/s > -1 ); > >And this produced an Ogg Vorbis stream of about 80 kbps on average. > >But in rc3, the above call produces a circa 240 kbps Ogg Vorbis stream >(as reported by xmms). I tried putting 96000 in place of the -1 values, >and also putting 96 instrea...

...ut it's relevant to Vorbis and Icecast I > believe. > > I'm seeing more and more streaming stations using AACplus, with many > listeners being amazed at the sound quality. Most say that a 48kb/s > sounds > better than a 128kb/s MP3, which would put Ogg Vorbis at around 96kb/s > IMO. > That means only half the bitrate is required in AACplus compared to Ogg > Vorbis for the same sound quality. Up until this codec was available, > Ogg > Vorbis compared favourably with all the others. > > Is there anything that can be done to bring Ogg Vorbis up t...

I am using HTB for traffic shaping, my scenario is Eth1 is my local interface and I wan to shape bandwidth for IP 172.16.173.245. Now I want if any request from above IP comes its get 1mb speed and all others get 96kb speed, for this setup I make following script but it does not work. Can anyone help me ? htb script -------------- tc qdisc del dev eth1 root 2> /dev/null > /dev/null tc qdisc add dev eth1 root handle 11: htb tc class add dev eth1 parent 11: classid 11:1 htb rate 10mbit tc class add dev eth...

...senc inputfile outputfile will be as small as possible and of as high quality as possible for human ears? So no need to up the knobs for listening on good output devices and digital connections (cabling)? I remember that for oggenc it was important to select the right -q level. For opusenc it seems 96kb avg is chosen and on mid-end speakers I didn't notice quality issues.

...1): 85kB/sec (100%/70%) Data from /dev/zero: lzo1x-1: 483kB/sec (92%/89%) lzo1x-999: 152kB/sec (100%/55%) zlib: 92kB/sec (100%/67%) Large .html: lzo1x-1: 380kB/sec (95%/82%) lzo1x-999: 106kB/sec (100%/41%) zlib: 88kB/sec (100%/67%) Files from /usr/bin: lzo1x-1: 330kB/sec (90%/74%) lzo1x-999: 96kB/sec (100%/39%) zlib: 76kB/sec (100%/72%) An mpeg video: lzo1x-1: 250kB/sec (80%/61%) lzo1x-999: 83kB/sec (100%/37%) zlib: 84kB/sec (100%/65%) Lzo1x-1 is much faster than the fastest zlib compression. Great for slow tunnel-gateways. Teemu -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.1 (GNU...

...und 30kb/s. During the encoding process i was "warned" that the encoder is tuned toward 44.1kHz and encoding files in any other may result in degraded quality. How worried should I be? Is the advantage in size lost due to crappy quality (versus 44.1kHz mono files)? comment: twinhq 96kb/s and oggvorbis beta4 ~100kb/s are virtually identical. however, once twinhq is dropped down to 80kb/s, all hell breaks loss and they sound shot-to-shit. they hardly compare to 50kb/s OggVorbis files. I used the latest version of the twinhq encoder (as of last week). -- noodlez: Karol Pie...

...s "stalled". (Identified by code inspection). Secondly, the ssh process spawned by the scp process to perform the actual encryption and transfer includes a substantial internal buffer (>64KB) and appears to implement hysteresis. ktrace output of a sample transfer shows a peak of over 96KB buffered - at which point the ssh process stops reading until the buffer drops to about 32KB. This implies that there is approximately 64KB hysteresis and a transfer rate below about 13KB/sec can result in "stalled" reports. This behaviour is a regression from from an earlier version of...

...tack:0kB pagetables:0kB unstable:0kB bounce:0kB writeback_tmp:0kB pages_scanned:0 all_unreclaimable? no lowmem_reserve[]: 0 1464 1936 1936 Node 0 DMA32 free:1122012kB min:4240kB low:5300kB high:6360kB active_anon:23324kB inactive_anon:8kB active_file:15332kB inactive_file:230356kB unevictable:4496kB isolated(anon):0kB isolated(file):0kB present:1499360kB mlocked:4496kB dirty:396kB writeback:0kB mapped:13448kB shmem:144kB slab_reclaimable:19140kB slab_unreclaimable:9520kB kernel_stack:984kB pagetables:4116kB unstable:0kB bounce:0kB writeback_tmp:0kB pages_scanned:0 all_unreclaimable? no lo...