I compiled the speex library for symbian using carbide C++. The compile completes with zero errors/warnings. I use the compiled library in my QT application. Compiling the QT application gives the following error - warning: COMDAT symbol '.exc$T' does not match section name '.rdata' (Its really an error, not a warning. the word warnin...
There are speex audio embeded in RIFF-WAV container (The files obtained in Windows system through encoding using Speex ACM codec, WAVEFORMATEX.wFormatTag = 41225 (0xA109) is used). Now I need to write an application, that will play such wav files. I just started to learn programming under Mac OS X, so I am looking for right start points...
Im trying to create a Speex Voip client and server. I have the basics down and its working OK on the local machine over UDP. I am using JSpeex for portability. Im looking for tips on creating the client and server. What are your thoughts? The JSpeex library can only encode 320 bytes per call so the packets sent to the server are tiny (i...
Hi, I need to extract the audio from a FLV file recorded using FMS. I used the SPEEX coded in my flash application that streams the audio and video to FMS. I have tried using FFMPEG, with no luck. This is the -i outputs of my FLV: FFmpeg version SVN-rUNKNOWN, Copyright (c) 2000-2007 Fabrice Bellard, et al. configuration: --enable-gp...
How can I add SPEEX support to my FFMPEG installation? I need to extract the audio from a FLV created by FMS. I just installed it using: app-get install ffmpeg. ffmpeg -version FFmpeg version SVN-rUNKNOWN, Copyright (c) 2000-2007 Fabrice Bellard, et al. configuration: --enable-gpl --enable-pp --enable-swscaler --enable-pthreads --ena...
Hello I want to use a Microphone to capture voice and encode that with Speex and then send it over a NetStream to a server. However does only FMS (Flash Media Server) and Adobe Labs Stratus (some p2p thingy) work as server software? If that is the case can I use the sampleData event of the Microphone class to capture sound and send it o...
Hello, For some time I've been using gsm codec for network audio chat. Now I'd like to implement smth like speex or iLBC due to "voice prediction". As far as I understand i need timestamps for packets. Also I know that to make codec to predict voice I have to pass it NULL packet for example. But I cannot understand the entire concept!...