similar to: advantage of OPUS_SET_FORCE_CHANNELS(1)?

Phil Karn wrote: > This works well, but then I noticed the SET_FORCE_CHANNELS option. Is > there any advantage to my detecting identical left and right channels > and setting this to force mono? Or does the Opus encoder already do this > internally in the AUTO mode? No, the automatic setting should be fine. The purpose of this setting is for the case where you know the receiver will

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 openssh-1.2.1pre20 has been released at: http://violet.ibs.com.au/openssh/files/ This release integrates more of Andre Lucas' portability patch, Ben Taylor's utmpx patch and some cleanups and bugfixes of my own. The auth-passwd failures should be fixed, as should lastlog support on NetBSD. Since Andre Lucas' patch included platform

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 openssh-1.2.1pre20 has been released at: http://violet.ibs.com.au/openssh/files/ This release integrates more of Andre Lucas' portability patch, Ben Taylor's utmpx patch and some cleanups and bugfixes of my own. The auth-passwd failures should be fixed, as should lastlog support on NetBSD. Since Andre Lucas' patch included platform

There's one vulnerability that's bugged me for some time. It applies to nearly all crypto software, including ssh. That's the swapping of sensitive info (such as keys and key equivalents) onto hard drives where they could possibly be recovered later. The Linux kernel provides a system call, mlock(), that inhibits swapping of a specified region of virtual memory. It locks it into real

Scenario: Use the ssh-keygen utility in openssh-1.2pre17 to generate a host key Kill and restart sshd Remove the old host key from ~/.ssh/known_hosts Connect to the host using ssh. I get this: homer.ka9q.ampr.org$ ssh 199.106.106.3 who The authenticity of host '199.106.106.3' can't be established. Key fingerprint is 1024 a0:8d:17:f0:fa:a9:9f:6f:b5:d0:1c:d6:02:92:bd:5e. Are you sure

Hello Thomas and Amit, Thanks for your notice and the detailed decode performance report. I describe the details of my encode/decode test on STM32F407ZG. A. opus version: latest 1.2.1 (TI: opus 1.1.2) B. KEIL 5.23 (TI: ARM compiler tool chain 5.2.7) C. setup the encoder as the below (fs is the sampling frequency) enc = opus_encoder_create(fs, chans, OPUS_APPLICATION_AUDIO, &opus_err);

Hi Jean-Marc, On Jun 15, 2013, at 12:20 PMEDT, Jean-Marc Valin wrote: > > >> So I still wonder, if you set up a custom mode, but then had all the >> settings the same as a normal mode, would the codec perform worse, or >> the same? > > You'll have to try normal vs custom modes and choose. The only thing I'm > telling you is don't run a 48 kHz

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Hi, folks, I have an old CELT thing that I'm updating to Opus and I'm trying to get individual pieces upgraded.  My first task is upgrading the CELT encoder to Opus. The old CELT thing treated the L and R channels as independent mono streams, encoded them somehow with CELT to form one bitstream, sent one bitstream across, and then unpacked it manually after the decoder. Can I do that

>Yesterday I have finished writing the ambisonic pan filter for oggenc. May I ask what this "pan filter" is? I made some tentative suggestions for coupling Ambisonic B-format in a post "Vorbis Ambisonic coupling" on 4feb07 I gather from the last monthly meeting, that some of you, including Monty, had problems with the phase behaviour of B-format. Would anyone like a

FLV contains TC messages? TC message payload contains a format byte and speex frames (up to eight). In the format byte 0xb0 indicates speex. Speex is always 16 kHz, 16 bit, mono. Jozsef Message: 1 Date: Mon, 16 Nov 2009 14:40:20 -0600 From: Jeff Ramin <jeff.ramin at singlewire.com> Subject: [Speex-dev] jspeex question To: speex-dev at xiph.org Message-ID: <4B01B8B4.8020904 at

Hi guys I succesfully got my encoder and decoder working after much hassles, but when I use the same code in another project, I get these following errors: Error ---> Invalid sideband mode encountered (1st sideband): 7 Error ---> Invalid sideband mode encountered (1st sideband): 7 Error ---> Invalid sideband mode encountered (1st sideband): 6 Error ---> More than two

My apologies for the confusion, I think I have the mapping layout correct in this patch. Cheers! On Tue, Apr 25, 2017 at 10:07 AM Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: > On 25/04/17 10:12 AM, Drew Allen wrote: > > We assume that the input file is ordered first by ACN ambisonic channels > > followed by a (possible) stereo track, and we want to swap the order for

We have a customer who wants us to record anywhere from 2-4 participants on a call in stereo (as opposed to mono) quality audio. Some background.. We are using asterisk 16.6.1 We are also currently using AMI/AsyncAGI and ConfBridge to bring the parties together. I believe recording in the various file formats (based on extension), it's always recording in mono quality. My one thought is to

Hello all, I created today a fake stereo piece of music (the left and right channels were completely equal), and encoded it with the all the modes the RC2 encoder supports, and found that: * * dual * waste * waste * * mode * joint * stereo * in * in % * * * stereo * /2 * kbits * of JS *

The link is http://www.adobe.com/devnet/rtmp/. TC Message stands for TinCan message. It is 11 bytes long, first byte is message type, three bytes of payload length four bytes of timestamp and three bytes of stream ID. The first byte of the payload for audio message is the format byte and the rest of the byte is the payload. Jozsef ----- Original Message ---- From: Jeff Ramin <jeff.ramin

Looking at the Opus comparison page[1], I can't figure out whether the Opus/AAC/Vorbis/MP3 lines are meant to imply a mono or stereo encoding. Could someone please update the caption to clarify this? The single dot for G.711 is clearly mono, but for stereo music, are the codecs at the top meant to converge near 128 kbps, or 256 kbps? [1] http://opus-codec.org/comparison/

Hello, First, thanks a lot for FLAC. A few questions. Apologies if this is not the right forum or if these questions have come up before; the SourceForge archive search didn't turn up anything. 1. Would it be possible for FLAC to support more than 8 channels? 2. If (1) is not desirable because it would break compatibility or some such, then is it possible to build a local copy

Hi. To compartmentalize things a bit (e.g., to help limit the damage should one of my machines be hacked and my private RSA keys stolen) I use different RSA key pairs on my different client machines. So it occurs to me that it would be nice if ssh could log which key was used when logging in to a particular account that has more than one entry in .ssh/authorized_keys. Right now it simply says

> I agree with you here ... I wish that I could put the stream > into 30kbps/22khz stereo but the little Pen 166 MMX just > can't handle that. 30kbps/22khz stereo, which I think would only be possible with managed bitrates, would sound worse than mono. At these bitrates, stereo is luxury you cannot afford. I think most listeners would not even notice they were listening to mono