[Speex-dev] Re: compatibility issues.
alex at gorex.com.hk
Fri May 25 07:58:27 PDT 2007
I plan to setup linux server and in client side, XP and Linux.
Also, in order to collect the voice, I plan to use portaudio to collect, say
20ms as you suggested sample and feed to the speex encoder. For easier
implementation, I don't use OGG container, and just encode then send to the
spread network to the client.
Hope this work. Can you advise any potential problems by this approach?
----- Original Message -----
From: "Jim Crichton" <jim.crichton at comcast.net>
To: <alex at gorex.com.hk>
Cc: <greg at mytantrum.com>; <speex-dev at xiph.org>
Sent: Friday, May 25, 2007 10:31 PM
Subject: Re: [Speex-dev] Re: compatibility issues.
> For a streaming application like VOIP, you collect 20 ms of samples, feed
> this through the encoder, stick it in an RTP packet, and send if over the
> network. On the receive side you feed packets through a jitter buffer to
> the decoder, and then copy the output audio to your output device. Speex
> runs efficiently on most compilers, so the real-time requirement is not a
> big deal, as long as you know how to do this with your operating system
> (if you have one).
> You do not need Ogg containers for something like this, just search the
> archives for RTP references and you will find much more.
> Look at the SpeexClient application provided in the source tree, and at
> the testenc examples in the libspeex directory. The Speex part of things
> is pretty easy (thanks to Jean-Marc). It is the audio interface for your
> specific platform and the call setup protocals (e.g. SIP) where the real
> work is.
> If you identify the specific hardware/software platform that you plan to
> use, then someone may be able to point you to examples for the audio
> interface software.
> - Jim
> ----- Original Message -----
> From: <alex at gorex.com.hk>
> To: "Greg" <caroundw5h at yahoo.com>
> Cc: <greg at mytantrum.com>; <speex-dev at xiph.org>; "Jean-Marc Valin"
> <jean-marc.valin at usherbrooke.ca>
> Sent: Thursday, May 24, 2007 11:30 PM
> Subject: Re: [Speex-dev] Re: compatibility issues.
>> I am also working on VOIP application.
>> Now the speexenc.exe is used to encode a wav/raw file. However, in real
>> voice communication, the voice keep recording from the microphone and how
>> we employ speex to compress the voice and send over the network?
>> Quoting Greg <caroundw5h at yahoo.com>:
>>> okay that questioned is answered, thank you.
>>> I am interested in using speex in a VOIP application. do i need to put
>>> it in
>>> the ogg contianer format in order to encode/decode and send it? or will
>>> "as is"? if the latter then:
>>> "the packet is larger than the allocated buffer" message: whats your
>>> recomendaton for fixing that? i was thinking simply getting the size of
>>> and using the speex_bits_init_buffer() function to create a bigger
>>> However i
>>> want to keep latency in mind and make them as small as possible.
>>> i apologise if the questions seem simple but i've working at this for
>>> and i'm a little stump. a indication in the right direction would be
>>> thank you in advance,
>>> Jean-Marc Valin <jean-marc.valin at usherbrooke.ca> wrote: > does this have
>>> anything to do with the message on the example code:
>>> > "the packing used is not compatible with speexenc/speexdec" i know it
>>> > may sound stupid but it is my first time using speex lib. as well i
>>> > had assumed this was so and tweaked the decoder code to decode the
>>> > same file that the example encode code made, but i get a "packet is
>>> > larger than allocated buffer could not resize buffer, truncating
>>> > input" message.
>>> Well, the note says it all. It's just not compatible. speexenc/dec pack
>>> Speex data into an Ogg container, while the sampleenc/dec code just
>>> creates an incompatible ad-hoc format just to demonstrate how to use
>>> Speex (without bothering with the heavy Ogg code).
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