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https://issues.apache.org/jira/browse/GUACAMOLE-1413?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=17408146#comment-17408146
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Nick Couchman commented on GUACAMOLE-1413:
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{quote}
"If the RDP server uses a different value, the Guacamole audio stream will
simply be reset appropriately.", from where do you have this information?
{quote}
I'm not sure what you mean by this? This comment was written by the person who
wrote the code for this (probably Mike, but could have been one of the other
contributors). The point is that the Guacamole RDP code defaults to a certain
value (2 channels), but if the RDP server wants to change that value, it can,
and the code is written such that the default value will be changed if
requested by the server. That's what the comment is saying. You can look
through the RDP code to see where/how this happens - I think this block of code
is what handles the callback from the server with audio format information, and
sets up the number of channels that the server sends back:
https://github.com/apache/guacamole-server/blob/b2ae2fdf003a6854ac42877ce0fce8e88ceb038a/src/protocols/rdp/channels/rdpsnd/rdpsnd-messages.c#L33-L186
specifically:
https://github.com/apache/guacamole-server/blob/b2ae2fdf003a6854ac42877ce0fce8e88ceb038a/src/protocols/rdp/channels/rdpsnd/rdpsnd-messages.c#L159-L161
It is worth noting that it looks like Guacamole, in general, supports two
channels, maximum:
https://github.com/apache/guacamole-server/blob/b2ae2fdf003a6854ac42877ce0fce8e88ceb038a/src/libguac/guacamole/audio.h
If you look at the functions defined throughout that code, you'll see that
valid values for number of channels are 1 or 2 - so, essentially you can get
mono or stereo sound, but you're not going to get Dolby Digital 7.1 Surround
(or 5.1, or 3.1, or 2.1) through Guacamole.
> RDP Session Parameter for the Number of Audio Channels to be transmitted
> ------------------------------------------------------------------------
>
> Key: GUACAMOLE-1413
> URL: https://issues.apache.org/jira/browse/GUACAMOLE-1413
> Project: Guacamole
> Issue Type: New Feature
> Components: guacamole, guacd
> Reporter: Juergen Kuri
> Priority: Major
>
> h3. Context
> We want to use the Guacamole RDP proxy server with full audio support using
> # audio input redirection, rule {{enable-audio-input=true}}
> # audio output redirection, rule {{disable-audio=false}}
> In the source code, header file {{src/pulse/pulse/pulse.h,}} we see the
> following constants for the Pulse Audio library for PCM modulator
> parametrisation of the RDP server:
> {code:c}
> /**
> * Rate of audio to stream, in Hz.
> */
> #define GUAC_PULSE_AUDIO_RATE 44100
>
> /**
> * The number of channels to stream.
> */
> #define GUAC_PULSE_AUDIO_CHANNELS 2
>
> /**
> * The number of bits per sample.
> */
> #define GUAC_PULSE_AUDIO_BPS 16
> {code}
> h4. Problem
> Parameter of interest:
> * {{GUAC_PULSE_AUDIO_CHANNELS 2}}
> For remote users with an internet connection with low network bandwidth, it
> might be useful to have sessions with just *_one_* audio channel, mono
> instead of stereo expecting utilising just half of the network bandwidth for
> the part of audio blob transmission.
> Is this assumption correct, transmitting just one audio channel would half
> the network bandwidth utilisation for audio blobs?
> h3. Feature
> Add another rule parameter for an individual RDP server session or connection
> respectively
> * for database table {{guacamole_connection_parameter}}
> which allows us to parametrise the number of simultaneous audio channels to
> be transmitted *_in both redirections_*, audio in- and output.
> Possible parameter names might be:
> * {{disable-audio-stereo}}, default {{false}}
> * {{enable-audio-mono}}, default {{false}}
> * {{audio-channels}}, \{1|2}, default {{2}}
> If this is not feasible or difficult to implement for some reasons, add a
> corresponding parameter to daemon configuration file
> * {{guacd.conf}}
> * or {{guacamole.properties}}
> affecting all RDP sessions in the same way.
> h3. Also of Interest
> The parameter {{GUAC_PULSE_AUDIO_RATE}} would also help to reduce network
> bandwidth if we reduce the sampling rate of the PCM modulator. If full
> audible audio bandwidth is not needed cause we just transmit speech for
> example, the sampling rate can be reduced as we know it for telephone lines
> (audio bandwidth 300 Hz to 3400 Hz). This however is more complex, a low-pass
> or an anti-aliasing filter must be programmatically implemented avoiding
> audio signal aliasing. Topics like Nyquist–Shannon sampling theorem,
> Discrete Fourier Transformation DTFT and audio signal aliasing must be
> studied for deeper understanding before implementation.
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