similar to: Who is using the jitter buffer?

-----Original Message----- From: speex-dev-bounces@xiph.org [mailto:speex-dev-bounces@xiph.org] On Behalf Of Thorvald Natvig Sent: Monday, March 20, 2006 12:11 PM To: speex-dev@xiph.org Subject: Re: [Speex-dev] Who is using the jitter buffer? But I must say that it really does work well at the moment, at least from the "minimizing latency" point of view :) How do you timestamp the

> That's basically my question: the timestamps at the source and > destination are not related. Just incrementing by number of samples > doesn't really convey the real time, does it? How would a jitter > buffer know that a packet is late/early? Simple, I know what packet I just played. That gives me the "time". The jitter buffer actually makes no difference (and

It seems that speex jitter buffer is tightly coupled with SPEEX codec [we have to give a speex decoder instance to JB]. It would be better if we could use it with any codec, like speex preprocessor and AEC. What are the any paper/theory/algorithms behind current ADAPTIVENESS of speex JB? Links to those algo/papers would help to understand better. -- Shantanu --- Thorvald Natvig

Thank you for a very good explanation which shed light on some of the questions that I had after reading the source code. Reading your text however, I wonder if I'm perhaps missing an important point on the proper use of the jitter buffer: ... > Now, clearly, if early_ratio is high and late_ratio is very > low, the buffer is buffering more than it needs to; it will > skip a frame

Thorvald Natvig wrote: >> I'd like know about anyone using the current jitter buffer in Speex. I'm >> planning on changing it to make it more general and I'd like some >> feedback about how to make it better. Also, let me know if you're doing >> anything serious with it and want to make sure I don't break your stuff. >> >> Basically, I want

Hello, The way you describe how the jitter buffer should be implemented makes me wonder: How does the jitter buffer works when there is no transmission? Let's say my "output" thread gets a speex frame from the jitter buffer every 20ms. What happen when there is no frame that arrived on the socket? No frames at all for a pretty long time (ie many seconds). This is my case because I

> That's basically my question: the timestamps at the > source and > destination are not related. Just incrementing by number of samples > doesn't really convey the real time, does it? How would a jitter > buffer know that a packet is late/early? > Simple, I know what packet I just played. That gives me > the "time". The jitter buffer actually makes no

Patch applied offline (I'm in the plane), remind me if I forget to push it. About the "resync on burst" issue, I was aware of it, but I'm not sure how to fix it. Any idea? Also, do you see the problem happening often? Jean-Marc Thorvald Natvig a ?crit : > > The jitter buffer would "freeze" under the following condition: > > - The sender and receiver are

Oh, I forgot to mention one thing. I currently append a few bits of custom information to each speex packet which I fetch out with things like speex_bits_unpack_unsigned(&sjJitter.current_packet, 1); It would be very usefull if the jitter buffer didn't actually decode the packet, but instead returned a pointer to it (or NULL if you should play silence and -1 if you should repeat the

Hi, I'd like know about anyone using the current jitter buffer in Speex. I'm planning on changing it to make it more general and I'd like some feedback about how to make it better. Also, let me know if you're doing anything serious with it and want to make sure I don't break your stuff. Basically, I want to make the jitter buffer easier to use with other codecs and reduce the

Jean-Marc Valin wrote: > Thorvald Natvig a ?crit : > >> I see you're changing the jitter buffer around quite a bit. Could you >> let us know when it's ready for general testing? (At the moment it >> doesn't handle missing packets at all) >> > > While I'm not completely done yet, I thought the current version was > working. Can you tell

The jitter buffer would "freeze" under the following condition: - The sender and receiver are in sync. - The sender machine freezes for a few seconds while the receiver does not. - This causes all the packets sent by the sender to have a timestamp that is too low to be considered, meaning jitter_buffer_put ditches it before it checks if lost_count > 20 to reset. - The jitter buffer

Le mardi 02 mai 2006 ? 18:15 +0200, Thorvald Natvig a ?crit : > Hi. > > After changing my code to construct a JitterBufferPacket and passing that > to speex_jitter_put, my program works with the new jitter buffer using the > wrapper functions (speex_jitter_* instead of the new jitter_buffer_*). Oops, I forgot to make sure I keep the API stable for the old buffer. Any thoughts on

pwk.linuxfan@gmx.de wrote: > On Sunday 08 July 2007 02:24, Thorvald Natvig wrote: > >> Next, and slightly unrelated question: >> Why does >> int jitter_buffer_update_delay(JitterBuffer *jitter, JitterBufferPacket >> *packet, spx_int32_t *start_offset); >> include the packet parameter? It's not used in the code, and it also >> makes the API slightly

> Hi, > > It could (and should) definitely be done, but it's somewhere > (unfortunately not on top) on my TODO list. I think the simplest way > (until I really integrate the preprocessor with the codec, which will > take a while) to do it is to add a speex_encoder_ctl() call as you > proposed. I'd call it SPEEX_SET_EXT_VAD_STATUS and I'm willing to merge > a

Is is possible to give a short hint about how the jitter buffer would "catch up" when network condition have been bad and then get better? I'm using the jitter buffer with success now, but sometimes I have a long delay that's caused by bad network conditions and then later when the conditions get better, I would think we would want the audio to gradually catch up with real-time

But that's what a jitter buffer will do, eventually - drop a packet. And it's better to drop a silence packet. Cheers, .a -----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@usherbrooke.ca] Sent: Wednesday, December 21, 2005 3:39 PM To: Alex Bakaev Cc: Shantanu Gramsci; speex-dev@xiph.org Subject: RE: [Speex-dev] Bitrate problem On Wed, 2005-12-21 at 15:29 -0800,

Hi, I see you're changing the jitter buffer around quite a bit. Could you let us know when it's ready for general testing? (At the moment it doesn't handle missing packets at all) Best wishes, Thorvald

Hi, I thought I'd repost this to the -users list for some background on the jitter buffer and its workings and remaining issue.s I'll also pu a little executive summary here at the top: Where a channel is native bridged to another iax2 channel: 1) Lag is not measured and will usually show 0ms. Any other number is an old measurement from the start of the call 2) The jitter

Thanks all for your prompt reply. I agree with Alex. I think jitter buffer will have to drop packets if its input bitrate is higher than it is configured for. I never tried speex jitter buffer. What does it do when input bitrate is higher or lower ? As an alternate solution, I was thinking to play with directsound playrate [using SetFrequency()]. Has anyone used it to solve such problem? Any