* Fixed format strings to match arguments
Reviewed and replaced all %d specifiers to match proper type
* Added proxy dynamic channel command type to log messages.
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
* Fixed remdesk settings pointer
* Fixed sign warnings in display_write_monitor_layout_pdu
* Use freerdp_abort_connect_context and freerdp_shall_disconnect_context
* Added and updates settings
* info assert/dynamic timezone
* mcs assert/log/flags
* Fixed and added assertions for wStream
* Unified stream length checks
* Added new function to check for lenght and log
* Replace all usages with this new function
* Cleaned up PER, added parser logging
* Cleaned up BER, added parser logging
* log messages
* Modified Stream_CheckAndLogRequiredLengthEx
* Allow custom format and options
* Add Stream_CheckAndLogRequiredLengthExVa for prepared va_list
* Improved Stream_CheckAndLogRequiredLength
* Now have log level adjustable
* Added function equivalents for existing logger
* Added a backtrace in case of a failure is detected
* Fixed public API input checks
RDPSND channel is special, as it has many names.
(e.g. static channel, dynamic channel and UDP one.
Use RDPSND_CHANNEL_NAME to identify the module name instad of
RDPSND_DVC_CHANNEL_NAME
* Fixed research for native PCM format
* Fixed compatibility detection for resampling
* dsp: Makes channel mix and encoding process use separated buffers
* Give FramesPerPacket to freerdp_dsp_context_reset function
* dsp: Fix DVI_ADPCM codec by respecting the required packet size
* Forbid mono channel in macos
* Fix audin command line channel option
* Disable ADPCM codec as it does not work
* Refactor valid native codec research
* Native codec research: Try different channels number
* Fix dsp channel mix
* Fix dsp resample
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.
- fixed invalid, missing or additional arguments
- removed all type casts from arguments
- added missing (void*) typecasts for %p arguments
- use inttypes defines where appropriate
