- Use autovideosink - xvimagesink does not work with cards with no xv ports available and cant be used if wanted to use the fluendo hardware accelerated playback codec
- Use autoaudiosink - let gstreamer choose best audio playback plugin
- Catch when autosinks add known elements so that we can manipulate properties on them
- Adjust caps of various media types to work better with gstreamer, some codecs are picky about having certain fields available
- Remove unneeded plugins such as "ffmpegcolorspace" and "videoscale" - these do not work correctly with fluendo hardware accelerated playback codec
- Name audio/video gstreamer elements better for easier debugging
- Update gstreamer pipeline and element properties to handle playback better
- Detect when valid timestamps are available for buffer from server and try to account for when they are not valid
- Start time is much more reliable then end time from server for various media formats, so use it when possible to make decisions instead of end time
- Do not rebuild gstreamer pipeline for a seek(very expensive), instead reset gstreamer time to 0 and maintain offset between real time and gstreamer time
- Change buffer filled function back to a buffer level function, so that we can use buffer level to make better choices above gstreamer decoder in tsmf
- Remove ack function from gstreamer, instead rely on ack thread to handle acks
- Rework X11 gstreamer code to handle various videosinks which implement the XOverlayInterface and to keep more detailed information on the sub-window that is used for display
- Add check to see if a decoder is available for telling the server the client various media types
- Add in support for M4S2 and WMA1 media types
- Fix flush message handling, they are for individual streams and not the entire presentation
- Delay eos response to try to allow more time for buffers to be loaded into decoder, as we anticipate acks to server and the server will issue stop as soon as we ack eos.
- Fix issue with geometry info being ignored when resent for new streams within existing presentation
- Fixed volume level initialization issue when a stream is stopped and restarted
- Attempt to sync video/audio streams...because we run two different gstreamer pipelines - they can enter pause/playing states at different times and are thus not synchronized. Attempt
to adjust video buffer timestamps based on difference between audio/video running time to account for this difference. This logic accounts for a huge improvement in audio/video sync(ie. lip sync to words)
The rdpContext gets an event which will
get set if an error occoured in a channel.
If a thread or a void callback has to report an
error it will get signaled by this system.
This patch adds some checks for the TSMF channel:
* checks malloc / calloc / strdup / _strdup return value
* checks for available bytew before reading in a stream
Change the return type of Stream_Ensure*Capacity from void to BOOL to be
able to detect realloc problems easily. Otherwise the only way to detect
this was to check if the capacity after the call was >= the required
size.
In case Stream_Ensure*Capacity fails the old memory is still available
and need to freed outside.
This commit also adds checks to most calls of Stream_Ensure*Capacity to
check if the call was successful.
* tsmf audio volume fixes
* The first audio volume message is received before everything is
initialized. Cache this data until we're ready.
* Handle the cases of gstreamer unitialized/partially ready/
fully ready.
* Default to 50% as a fallback if the above fails.
* Improved detection of the primary monitor.
* Fix an unexpected shutdown when the server asks us to resize the media
overlay to (width=0,height=0).
* Permit multiple presentations - required for repeat of audio files
and also is seen in some video cases.
* Several leak fixes.
* Fix logging related compile warnings and other usage of PRIu64.
