similar to: Speex - frame size & packet loss concealment

Thanks for the reply. I think I'm beginning to understand but can you clarify a bit more. 1) I am confused on "trade error-free quality" part. Basically the encoder is doing something to the encoded data by looking at the SPEEX_SET_PLC_TUNING parameter? 2) How does the receiver side determine whether the packet loss occured or not? It seems only way is assigning

Hi, I am an engineering student in IIT Guwahati, India. Currently, as a summer project I am working on 'packet loss concealment in voip scenario'. I am eager to know what PLC methods are used at receiver end of speex decoder. And if I could develop some more efficient method I could able to contribute to the speex. Actually, I searched the whole speex manual but failed to find what

Hi Does anyone know where one can get hold of a G729 codec for asterisk which effectively can do packet loss concealment using Steve Kann's wonderful new Jitter buffer. The 2 versions that I know of, (digium's and the IPP one) do not perform great at PLC, especially with 4 or 5% loss. Thanks in advance. Clive

Hi, I would like to know which method uses Speex to conceal the loss of packets. I've reviewed the code and haven't found anything. Thanks in advance. Iago Soto. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://westfish.xiph.org/pipermail/speex-dev/attachments/20040614/bfa135e0/attachment.htm

Hi, I'm a developer currently using the speex codec in a VOIP application of ours.. It sounds amazing, especially at wideband.. my question is how do I force it to do a concealment? We have a low latency application, and based on the current API, I'm guessing concealment only kicks in when a packet is lost.. However, our jitter buffer knows when a packet is missing, and I'd like to

Hi, I have read the mailing list archives and have a question regarding PLC (packet loss concealment).

Hi, I am trying to implement ogg vorbis real time decoder on SHARC ADSP 21364 processor. I am writing my own version of the decoder using the documentation vorbis_I_spec.pdf from xiph.org. However I am facing certain hurdles in floor curve decoding during audio packet decode. In the spec it says that [cval] = read from packet using codebook number [so and so] in *scalar context*.(page no 43).

The SOF ADSP vhost driver consists of two parts: a sound and a vhost part. This patch implements the vhost part of the driver. It handles QEMU communication with the vhost misc device and virtual queues to any VirtIO RPMsg guests. Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski at linux.intel.com> --- This is marked as an RFC / request for review, since it depends on other

I bought a ADSP-BF533-EZkit-LITE V2.1 to develop a speech codec application using SPEEX. I have tested SpeexEcho project from both BlackfinSDK-Rel201 and BlackfinSDK-Rel310. When disabling SPEEX processing, i.e. using PASS-THROUGH mode, the output voice is ok, which is exactly as the input voice. However, when enabling SPEEX processing, the output is bad, no matter how to set the parameters in

Hi, I am evaluating the opus codec for use in an embedded design. I have successfully ported opus on a TI C5502 (fixed poin dspt) but need to use a more powerful dsp processor for our application. With the C5502 it takes 4.5ms to encode 2 (mono) channels with a 5ms frame @80kbits per channel. I need to reduce the computing time to reduce latency and leave some time for the DSP to perform some

The ADSP device uses the RPMsg API to connect vhost and VirtIO SOF Audio DSP drivers on KVM host and guest. Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski at linux.intel.com> --- drivers/rpmsg/virtio_rpmsg_bus.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/rpmsg/virtio_rpmsg_bus.c b/drivers/rpmsg/virtio_rpmsg_bus.c index f3bd050..ebe3f19 100644 ---

Hi Mathieu, On Thu, Jun 04, 2020 at 02:01:56PM -0600, Mathieu Poirier wrote: > On Fri, May 29, 2020 at 09:37:22AM +0200, Guennadi Liakhovetski wrote: > > The ADSP device uses the RPMsg API to connect vhost and VirtIO SOF > > Audio DSP drivers on KVM host and guest. > > > > Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski at linux.intel.com> > >

> 1) May I know how Speex compared with GIPS codec? It seems that Google, > Yahoo, and Skype are licensing from GIPS. Are there any good benchmarking > or fair comparisons? I think these two emails sum up my opinion about Speex vs. iLBC: http://lists.xiph.org/pipermail/speex-dev/2005-June/003410.html http://lists.xiph.org/pipermail/speex-dev/2005-September/003652.html > 2) In

Hi Mathieu, On Mon, Jun 08, 2020 at 10:17:57AM -0600, Mathieu Poirier wrote: > On Fri, Jun 05, 2020 at 08:46:59AM +0200, Guennadi Liakhovetski wrote: > > Hi Mathieu, > > > > On Thu, Jun 04, 2020 at 02:01:56PM -0600, Mathieu Poirier wrote: > > > On Fri, May 29, 2020 at 09:37:22AM +0200, Guennadi Liakhovetski wrote: > > > > The ADSP device uses the RPMsg

Le jeudi 22 septembre 2005 ? 08:19 +0800, Joe Anny a ?crit : > Thanks, Jean for the clarification. My first name's Jean-Marc, not Jean... > I look forward to delving into your tutorial codes. What we hope is to use > Speex as the primary voice codec. We also hope to get a quick start into > the programming aspects with buffer management, denoising, etc. taken care > of (or

Hi, We are planning to use Speex as the speech codec for a VoIP application. 1) May I know how Speex compared with GIPS codec? It seems that Google, Yahoo, and Skype are licensing from GIPS. Are there any good benchmarking or fair comparisons? 2) In particular, how is the jitter buffer control for Speex in response to intermitent poor connection hiccups? Is it robust enough to smooth out

Hello I've been using VoIP over a not so reliable net: I usually get a 5% to 10% packet loss and a very high jitter. I tried several codecs and parameters, and the only thing left to test is PLC (Packet Loss Cancellement). Have the astesrisk and digium people implemented PLC?, Are they implmementing it now? and, if not, Where can i find an implementation? Thanks in advance -- Jorge

Hi Jason, On Fri, May 29, 2020 at 02:01:53PM +0800, Jason Wang wrote: > > On 2020/5/28 ??2:05, Guennadi Liakhovetski wrote: > > v3: > > - address several checkpatch warnings > > - address comments from Mathieu Poirier > > > > v2: > > - update patch #5 with a correct vhost_dev_init() prototype > > - drop patch #6 - it depends on a different patch,

Hi All I have recently downloaded Asterisk and was so impressed I thought I would setup a home server and I went out and got myself a couple of cisco 7940's. (and a sipaura 3000!). thanks to various posts on this list and the voip-info site I have managed to get chan_sccp setup and working with the 7940's but the I tried to get the messages, services and softkeys working. It seems

for the bits init I am using speex_bits_set_bit_buffer and I don't use the write to or read from because the data is already in the buffer I am reading from and I am writing to the final buffer so I don't need to move arrays around. what part is the vocoder part of the decode? Thanks for your help! -Mike >>> Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> 08/06/07