The current server side channel handling of AUDIO_INPUT is currently
very constrained:
- Server implementations cannot measure the clients uplink, since the
Incoming Data PDU is currently unhandled and FreeRDPs DSP handling
delays the callback call of ReceiveSamples
- Servers currently cannot prefer a different protocol version
- Servers currently cannot change the used format
To solve these issues without running into the risk that some
simplifications constraint certain API usage, rework the current channel
handling to be very close to the documentation.
This means, that all documented API calls can be made by server
implementations and all documented PDUs, that the server side is
expected to receive are just parsed inside FreeRDP and then forwarded to
the API implementation.
The current server sided channel handling of RDPSND/AUDIO_PLAYBACK_DVC
is currently very constrained.
So, solve this. This means:
- Add the missing Training/Training Confirm PDUs
- Stop overriding the average bytes per second values, when submitting
the audio formats, as this currently makes the usage of codecs
impossible
- Add a way to send the server formats manually again, to be able to
restart the protocol after a Close PDU was sent
- Add a way to send already encoded audio data to let server
implementations to take care of the encoding process and to set
custom audio timestamps for the Video Optimized Remoting channel
- Add public attributes to let server implementations know the initial
volume and pitch values
- Add public attribute to let server implementations know the quality
mode setting
The sound and microphone redirection channels (and in part TSMF)
did not properly decouple encoding/decoding from the backends used
to play/record sound.
Encapsulating encoding/decoding in rewritten freerdp_dsp_* functions
with variable backends, simplifying alsa/oss/pulse/... audio backends.
