similar to: Bitrate scaling

I'm trying to get a bitrate scaled live ogg stream working, but I've not found out how to get this running with the current tools. I've downloaded the CVS icecast2, and darkice, but the only option seems to be if the same stream is actually encoded multiple times at different bitrates. This is not good for my cpu. Can I get bitrate scaled ogg streams working with the current tools?

Hi, Although this maybe considered out of scope here, but I'll ask anyway. Opus has remarkable flexibility for changing encoder bitrate during a call. Are there any suggestions about how bandwidth/capacity between the two endpoints can be measured/estimated during a call so that the outgoing bitrate can be adjusted accordingly? Thanks, Manpreet. -------------- next part -------------- An

Hi Dragos, The issue is that not all packet loss maybe congestion related. Often, reducing bitrate seems to have no impact on improving packet loss. Thanks, Manpreet. On Tue, Dec 16, 2014 at 2:09 AM, Dragos Oancea <droancea at yahoo.com> wrote: > > Hi > > You can start a VOIP call with 50 kbps bitrate and reduce the bitrate if > there is packet loss. You know if

I had expected that the default bitrate for the encoder would be the same as setting it to OPUS_AUTO, but I'm getting difference results: >opusenc --comp 4 sample.wav sample.opus Encoding using libopus 1.3-rc2 (audio) ----------------------------------------------------- Input: 8 kHz, 1 channel Output: 1 channel (1 uncoupled) 20ms packets, 25 kbit/s VBR Preskip: 312

Hi! I have some MP3 audio material which is basically speech with some background noises, essentially > 120Hz and < 5kHz. I had the idea to reduce the file size by recoding the material to Opus at 56kbps. Unfortunately the result is a file sampled at 48kHz much larger than the original. I hope you agree that it does not make sense to create a file larger than the original (MP3). Of course

Hello, good morning ! Running Icecast2 and Ices2 with live audio from Line IN, what is the correct way to change the bitrate ?. In the Icecast2 configuration file I set: <mount type="normal"> ????<mount-name>/entzun.opus</mount-name> ????<bitrate>128</bitrate> </mount> But it seems that is not enough since the bitrate does not change. Receiving the

Hey. I just wondered,... which is the recommended bitrate and further settings of opusenc (like complexity and framesize) for general music (e.g. including classical music) to achieve more or less transparency? Talking about CD level audio (16bit; 44,1 kHz; Stereo) The only source I could find regarding that was: http://wiki.hydrogenaud.io/index.php?title=Opus#Music_encoding_quality But that

El mié, 23-08-2017 a las 14:13 +0200, Marvin Scholz escribió: > > On 23 Aug 2017, at 14:08, José Luis Artuch wrote: > > > Hello, good morning ! > > Running Icecast2 and Ices2 with live audio from Line IN, what is > > the > > correct way to change the bitrate ?. > > In the Icecast2 configuration file I set: > > > > <mount

Hi! So here is what I got with different encoder settings; still not sure what the "best" setting is. 6kbps seems to add distortions, so I tried 12kbps. MP3-Original: LAME 3.99r, 120kbps, 44100Hz, Stereo, VBR V5 (22:23, 19.8MB) Opus (--raw-rate 44100 --bitrate 56 --vbr --comp 5): (44:45?, 23.4MB): Broken Opus (--bitrate 56 --vbr --comp 5 --ignorelength - %d): (22:23, 12.1 MB, 74kbps)

Hi! Useful advice, thanks! Actually I had been using foobar2000 to recode, because it just makes it so easy to convert multiple files while keeping the metadata (I confess, I'm a "tagger"). But it's easy to miss some encoder option when being presented some default suggestions in a dialog form... Apart form that I always had the impression that Opus could be quite smart

At 09:56 PM 5/28/02 +1000, you wrote: > >I'm trying to get a bitrate scaled live ogg stream working, >but I've not found out how to get this running with the >current tools. I've downloaded the CVS icecast2, and darkice, >but the only option seems to be if the same stream is actually >encoded multiple times at different bitrates. This is not >good for my cpu.

I read this in the FAQ ( at www.vorbis.com, did everyone make shelters for the slashdot effect ? The Vorbis beta release is not posted yet, but is a matter of <some-time-unit>s :-) > Vorbis has bitrate scaling - a feature that lets you adjust the bitrate > of a Vorbis file or stream without reencoding; just chop the packets up > in the sizes you want them. So one can stream a 128

Hi again! I have some other suggestions/modifications. 1. Bitrate scaling by frequency in vorbis_encode_init: bpch=(long)((float)nominal_bitrate/(float)channels*44100.0f/(float)rate); With this modification, if we use -b128 at 22khz files, the OGG file will be 128kbit/s, not 64kbit/s ... 2. Probable quality bug: I tested some floating point constants (recalculated them (see below) and encoded

El mié, 23-08-2017 a las 14:42 +0200, Marvin Scholz escribió: > > On 23 Aug 2017, at 14:40, José Luis Artuch wrote: > > > El mié, 23-08-2017 a las 14:13 +0200, Marvin Scholz escribió: > > > > > > On 23 Aug 2017, at 14:08, José Luis Artuch wrote: > > > > > > > Hello, good morning ! > > > > Running Icecast2 and Ices2 with live

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Hi there, I'm using libvorbis in my program and need to encode to target bitrate. I know libvorbis prefer to use quality scale but I can't use it. I've found something at faq http://vorbis.com/faq/#quality *For now, quality 0 is roughly equivalent to 64kbps average, 5 is roughly 160kbps, and 10 gives about 400kbps. Most people seeking very-near-CD-quality audio encode at a quality

This patch adds a static bitrate allocation for ambisonics encoded with the multistream API. We allocate 4 bits to the mono channel (channel index 0) for every 3 bits allocated to other channels. The allocation is static because it does not depend on the input audio. This patch also forces the underlying encoders to use CELT only for ambisonics. -- Thanks, Michael Graczyk -------------- next

Hello, I am curious, what is the cause of some encoding artifacts, I guess part of them should not be present. - when encoding with a defined bitrate, some artifacts show up which I'd not suppose to be there at all. I attach a screenshot from the beginning og Harry Potter 1 at the moment before the "Privet Drive" plate slowly appears from the darkness. Note the green and red

Hello, If anyone is running an Icecast server that serves at least 3 separate streams each with it's own bitrate can I get a look at your config? Thanks. Dave.

On 2010-04-11 10:09, Randy Yates wrote: > If I specify VBR mode via > > speex_encoder_ctl(pSpeexEncoder, SPEEX_SET_VBR,&isTrue); Try: speex_encoder_ctl(pSpeexEncoder, SPEEX_SET_ABR, &desiredRate);