search for: 20ms

...Also, is this speech or music? What tool, what options? In general, it helps a lot if you post the sample (input and output). Cheers, Jean-Marc On 06/03/2016 12:48 PM, Kevin Connor wrote: > Hi Opus list, > > I'm noticing a discontinuity in the quality between use of 10ms and > 20ms framesize for mode=AUDIO at a bitrate of about 28000. > Quality drops audibly for voice signals when encoded at 20ms > framesize, versus quality at 10ms. This effect is mode=AUDIO only. > Using mode==VOIP shows no sig. difference between 10 and 20ms framing > at this bitrate. Pe...

Hi Jean-Marc, Sorry for late reply, thanks for interest. It's quality good for 10ms/audio, poorer for 20ms/audio. Quality equivalent for 10,20ms for mode=voip. PESQ was the tool that alerted me to something of interest, but I don't trust PESQ to almost any degree! It's good for hearing relative differences, of course, but not absolutes. Bitrate here was 28kbps, but I hear same thing at 32...

Hi, I have a question regarding the Asterisk Packet Time for SIP Calls. It is hardcoded at 20ms but when I do an RTP Analysis on a stream it is clear that these packets are not spaced out at 20ms. In general you see something like: Packet 50 - Delay 50ms Packet 51 - Delay 5ms Packet 52 - Delay 5ms Packet 53 - Delay 50ms Packet 54 - Delay 5ms Packet 55 - Delay 5ms Is there anyway to space...

Hi, Does anyone know a practical way to determine the number of 20ms speex frames in a given speex packet without actually looping on speex_decode* ? Cheers, Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20080208/0a3e84ac/attachment.html

Hi Opus list, I'm noticing a discontinuity in the quality between use of 10ms and 20ms framesize for mode=AUDIO at a bitrate of about 28000. Quality drops audibly for voice signals when encoded at 20ms framesize, versus quality at 10ms. This effect is mode=AUDIO only. Using mode==VOIP shows no sig. difference between 10 and 20ms framing at this bitrate. Pesq totally over...

...know, if you need more informations. opus-tools version 0.1.7 [root at studio-connect.de audio]# opusenc music_orig.wav music_orig.opus Encoding using libopus 1.0.3 (audio) ----------------------------------------------------- Input: 48kHz 2 channels Output: 2 channels (2 coupled) 20ms packets, 96kbit/sec VBR Preskip: 312 Encoding complete ----------------------------------------------------- Encoded: 1 minute and 30.82 seconds Runtime: 1 minute and 8 seconds (1.336x realtime)...

David Hogan wrote: > Does anyone know a practical way to determine the number of 20ms speex > frames in a given speex packet without actually looping on speex_decode* ? There's no ready-made function for that. You'd have to write a function that inspects the bits, determines what mode is involved, and jump the right number of bits until a terminator is found or there'...

Jean-Marc Valin wrote: > David Hogan wrote: > >> Does anyone know a practical way to determine the number of 20ms speex >> frames in a given speex packet without actually looping on speex_decode* ? >> > > There's no ready-made function for that. You'd have to write a function > that inspects the bits, determines what mode is involved, and jump the > right number of bits unt...

Hi all, I have spent some time searching, but I haven't found a way to prevent * from concatenating two frames into one IAX packet. I have a situation where I make an IAX GSM call to *, which transcodes to an iLBC SIP call. Every second voice packet the IAX client receives contains 2x 20ms frames, the other containing only one. I presume this is related to the mismatch of 20ms GSM vs 30ms iLBC. What is odd, is that this doesn't happen when I use G711 on the IAX leg - each packet contains 20ms. Anyway, the IAX client I am using isn't handling the 40 ms packets gracefully....

I am about to connect to a new provider who requires 20ms payload sizes in g729a. Is this configurable on asterisk? Is 20ms the default? Cheers, j

...from a librevox recording. It managed this at a respectable pace for complexity 10: Skipping chunk of type "LIST", length 76 Encoding using libopus 0.9.14 (audio) ----------------------------------------------------- Input: 44.1kHz 1 channel Output: 1 channel (1 uncoupled) 20ms packets, 6kbit/sec VBR Preskip: 356 Encoding complete ----------------------------------------------------- Encoded: 2 minutes and 0.02 seconds Runtime: 50.9 seconds (2.358x realtime) Wrote: 111194 bytes, 6001 packets, 123 pages Bitrate: 6.78414kbit/s (without overh...

>>> Amit Ashara <ashara.amit at gmail.com> schrieb am 12.05.2016 um 17:47 in Nachricht <CAEyg9sgjbsxQY-=VnhQrKiGeTcFSRr1wxOPUhNyCQF8Piuahow at mail.gmail.com>: > Hello Jean-Marc, > > Assuming that a 48KHz, 20ms 8-bit linear PCM data which is 960 bytes is > compressed to 64 bytes (for assumption). The with the Oggs header (4 byte) Actually what I don't understand ist this: If your whole audio is 20ms, why do you care at all to compress it? Is it that your audio is much longer, but you want to encod...

> In those cases, when you get let's say 1000 packets of 20ms from the mic > you may have only 990 packets of 20ms from RTP incoming stream. > > Thus, before sending outgoing mic/RTP stream, you would wait for 1000 > incoming packets: where last packet in fact arrive 10*20ms = 200ms > after it was supposed to. I have from my experience already...

...re that my solution is incorrect? Also, I have looked over the draft RFC of the Speex RTP payload and I was curious to know if, because I am not using a standard sample rate, my solution will work. In the draft RFC (and the Speex manual for that matter) it always refers to Speex frames containing 20ms of audio. In my case, a Speex frame actually contains roughly 14.5 seconds of audio. Am I going to need to hack the Speex encoder to make my frames be 20ms in order for the RTP to successfully play? Any advice/help anyone can give would be greatly appreciated. Thanks! Randy -----------------...

Hi Jean-Marc, A simpler question. How does opus_encode() generate packets of 20ms (SILK-only or Hybrid)? Concatenating two 10ms frames or doing it straight with just one 20ms frame?

Hi, On 2/9/08, Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > Alexander Chemeris a ?crit : > > Hi Jean-Marc, > > Is your proposition that ever this code will be written in terms > > of internal Speex bits it will be included into libspeex still in force? > > yes. I'd like to include that if someone writes it using SpeexBits. Here is first attempt.

Hi, Ok, here is cleaned up and fixed version. * Function is named speex_get_num_frames() now and return number of frames, as you suggested. * WB layers sizes are taken from wb_skip_table[], while NB frame sizes are calculated with speex_mode_query(). I've tested it with testenc in NB, WB and UWB modes with VBR enabled to test in as many cases as possible. It seems to work fine. Though,

For clarity I attached modified testenc.c I used to test my code. On 2/14/08, Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > > I also think that having rame_num.c in libspeex, code size could be > > reduced and simplified slightly by using speex_skip_wb_frame() in > > nb_decode() instead of duplicating code. > > Well, the code is a bit different (e.g.

On 2/16/08, Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > > Oh, you're right, inband handling is different, but it's a matter of one > > flag, passed to functioin. Probably not that much to keep code DRY. > > Well, let's first merge your code and then we'll see about possible > simplifications. Waiting for your decision ;) I guess you stopped

On 2/20/08, Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > Alexander Chemeris wrote: > > On 2/16/08, Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > >>> Oh, you're right, inband handling is different, but it's a matter of one > >>> flag, passed to functioin. Probably not that much to keep code DRY. > >> Well,