similar to: How to make the audio header data for Speex for RTMP ?

Hi, Can you please tell me if swfdec support playing videos using rtmp? E.g. www.nytimes.com 's flash video player using rtmp. Does swfdec support it? Thank you.

New branch ''rtmp'' available with the following commits: commit 03292617084fb246a2f23f566d9b5a1afafede41 Author: Benjamin Otte <otte at gnome.org> Date: Sun Dec 14 21:43:56 2008 +0100 split in_error() from is_connected() commit 139261910ef6120f56c6c6749a3843c5dd4af4e4 Author: Benjamin Otte <otte at gnome.org> Date: Sun Dec 14 21:25:50 2008 +0100

Hi, Can you please tell me if swfdec supports RTMP? (flash streaming protocol)? Thank you.

Dear Icecast developers, we (freiheit.com and radio.de) are using Icecast as a streaming server. Our targeted client is based on Adobe flash. We investigate the options to implement RTMP streaming (an Adobe streaming format: http://osflash.org/documentation/amf) as an alternative streaming format for the Icecast server to improve the quality of audio streaming in flash clients. Background:

> Date: Tue, 30 Mar 2010 19:52:42 +0400 > From: Max Lapshin <max.lapshin at gmail.com> > Subject: Re: [Speex-dev] Speex in flash player: how to work with? > > _microphone = Microphone.getMicrophone(); > _microphone.setSilenceLevel(0); > _microphone.codec = SoundCodec.SPEEX; > _microphone.encodeQuality = 10; > _microphone.rate = 16; > _microphone.framesPerPacket

Keep in mind that Speex doesn't know about RTP, networks, files or anything like that. So the byte you're seeing is either added by the client, or else it happens that Speex produces a byte that you're misinterpreting as to mean something else. A normal Speex CBR Speex stream always has the first 5 bits set to the same value (indicating the mode), so maybe the audio is stationary

Hi, Has anybody some information on on the "Redundant audio data" header in the speex rtp payload? Is this header always present? Is its value always the same? Can it be modified through some speex_*_ctl function? Thanks, Emmanuel -- ------------------------------------------------------------- Emmanuel Wauters Tel : (+32) 11 30 13 30

Saju Palayur <saju_pa at hotmail.com> writes: > Hi > > > > We have ported the Narrow band 8 Khz Speex encoder and decoder on to TI DSP 5505 > > > > To compress 20ms of 16 bit audio sampled at 8Khz it takes 4.2ms. > > Is there any optimizations I can do to make it run faster. The first thing to do would be to profile the code (I assume you're using

I just tried to install R version 1.5.1, and it is failing ``make check'', as follows: $ make check `Makedeps' is up to date. running code in 'base-Ex.R' ...*** Error code 1 make: Fatal error: Command failed for target `base-Ex.Rout' Current working directory /home1/faculty/rolf/Traal/Rtmp/R-1.5.1/tests/Examples *** Error code 1 make: Fatal error: Command failed for target

Hi When we run the profiler these 3 functions take more than 8500 cycles. split_cb_search_shape_sign_N1() 22782 pitch_search_3tap() 19208 lpc_to_lsp() 16007 So can I configure these in a different way so it takes less cycles. I do not need very high sound quality Thanks for your time Warm Regards Saju > From: yates at ieee.org > To: saju_pa at hotmail.com > CC: speex-dev

Saju Palayur <saju_pa at hotmail.com> writes: > Hi > > > > When we run the profiler these 3 functions take more than 8500 cycles. > > > > split_cb_search_shape_sign_N1() 22782 > pitch_search_3tap() 19208 > lpc_to_lsp() 16007 > > So can I configure these in a different way so it takes less cycles. I am not familiar with the code, so I can't

I'm a software engineer. I'm terribly uncomfortable with C. I want someone to show me a simplified use of Speex decoding. FWIW I'm finally getting my head around the API but the example had me a little lost with its cryptic variable names and monolithic nature. The example I'm asking for is way simpler than the final code I will ship to production so in essence, I'm looking for

Clifton Craig <clifton.craig at gmail.com> writes: > Hi, > > I'm new to Speex and a little uncomfortable in C programming. I'm > looking for a simple example of how I'd use Speex to decode a speex > file read off of an network stream and pass the decoded samples to an > audio player. I have speex running in an iPhone project and I started > digging thru the

Don't use extra TEMPs unnecessarily in some cases. --- src/gallium/drivers/nv50/nv50_program.c | 120 +++++++++++++++--------------- 1 files changed, 60 insertions(+), 60 deletions(-) diff --git a/src/gallium/drivers/nv50/nv50_program.c b/src/gallium/drivers/nv50/nv50_program.c index d7ab28a..5594560 100644 --- a/src/gallium/drivers/nv50/nv50_program.c +++

Hi I''m trying to compile R packages on my vista (I''m very new on vista...). I installed Rtools, RExcel on a specific folder: c:\Rtools and gave full access. Unfortunately, I am still unable to compile any packages (may be still because of the security properties of vista). I checked several mailing list/forums/website but I did not found any solution. Does anybody have the

I have received your answer regarding the intention of developing audio mix algorithm based on Speex (a while ago...). You've mentioned two different scenarios. VOIP applications that can manage with high bit-rates can truly mix the audio just once at the receiving client. Today most sound cards can mix the decoded streams automatically (no need for mix software). The real problem is

The main point of this discussion has been that deleting the tempdir() a fixed number of days after its creation is a problem. It should be deleted when the process that created it is done. R attempts to do this, but there are cases in which it does not so a backup is needed. Bill Dunlap TIBCO Software wdunlap tibco.com On Wed, Apr 26, 2017 at 8:46 AM, Brian G. Peterson <brian at

Prior to having your ACM codec, I was manually compressing Speex audio and writing it into one stream of an AVI file. To be able to use your codec for playback, I simply appended your WAVEFORMATEX extra data (the extra fields in SPEEXWAVEFORMAT) to the audio format. In looking at your code trying to determine how to fill in that data, it looked like the fmtnotice was optional. Is this truly the

I am trying to record a call into a stereo mp3 in Asterisk 1.4, but I can't seem to get it to work correct. Could someone point me to what I need to do? I have attached what I believe are the relevant parts. [globals] ; script to be executed when monitoring has been finished MONITOR_EXEC=/usr/local/bin/2wav2mp3 ; uncomment this line if you are using Ogg Vorbis

Hi, Rhett, echo_diagnostic.m can produce an echo canceled file along with the possible warning messages. Can it cancel the echo properly? It may announce the "Drift estimate is ..." message if the recorded sound contains not only echo but also other sound like a near-end speech. In other cases, it may truly indicate a mismatch of the sampling frequency. Regards Kaiyu On Thu, Feb 6,