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 shadow server tries to resize the client to use a compatible
resolution. If that fails, e.g. if there is another resize request after
the disconnect/reconnect sequence, abort the connection.
* Add callbacks for all messages exchanged between client and server
to allow server implementations to intercept them.
* Unify logging
* Add device tracking
When some channels are filtered, some misalignement of channel ids could happen.
This patch keeps track of the back and front channel ids to correctly identify a
channel and send packets with the correct channel id.
This big patch fixes fragmentation handling in the dynamic channel. We used to
have a single state to handle fragmentation at the main dynamic channel level, but
in fact packets can be fragmented per sub channel. So we have to maintain a fragmentation
state per sub channel, this involve treating dynamic and static channels differentely
(so the size of the patch that has to implement state tracking per dynamic channels).
This adds a User, Domain and Password parameter in the Target section of the configuration
to specify and use a fixed backend user, domain or password (overriding the one
passed by the front user).
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 rework introduce a stateful dynamic channel treatment, so that we can take early decisions
for data packet (dropping all the current packet or pass it), but also reassemble important
packets like channel creation.
* Fixed GetFileInformationByHandle initializers
* Fix#7793: Do not expose internal input API
Slow-Path input uses UINT16 for scancodes on wire, but only the
lower byte is actually used. (the extended fields are sent in
keyboardFlags field)
Hide this implementation detail and adjust the API to use UINT8
for the code instead just like the corresponding Fast-Path PDU
* Added a warning for problematic slow path keyCodes
This PR introduces per channel context so that we can speed up operations like
retrieving the channel name from its id, or knowing what shall be done for a
packet (no config ACL recomputation at each packet).
