Modified by: milko at real.com
Date: 9:26:06
Project: Atlas

Synopsis: Player core stability

Overview:
The submission includes a number of correction to a number of defects 
discovered whil testing more complex (multi-track) presentations.  It also 
includes changes previously presented in "Client Performance Work part 5" 
while incorporating feedback on the changes.

-> Properly handle addition and resumption of an audio stream to a playing 
audio session.  Audio player in such case will initiate E_PLAYING audio 
state change which in case of already playing audi dession will rewind, 
reset and resume the audio device.

-> Added a bunch of logging statement in method improtant for monitoring 
multi-track operation and performance.

-> Corrected audio session not to stall in case of rebuffer occured in the 
process of resuming.  Previously in such case (although very rare), the 
playback would never start or resume.

-> Corrected treatment of audio stream last write time rleative to stream 
base time and stream delay for live streams.  Live streams do not start at 
0 but at an arbitray time-stamp.  Any treatment of this value in absolute 
comparision is invalid as done previosly.  The stream base time is the 
audio stream starting time in renderer time coordinate system.  The delay, 
is the amount of time the stream is to be delayed relative to the value of 
the time-line from the moment of the introduction of the audio stream into 
the audio session.  This is accomplished by modifying the last write time 
to create the needed gap indicated by the delay.  The wrap-around is not an 
issue as it detected and accounted for at the time new audio data is written.

-> Made audio player an InterruptSafe object thus causing fake time-line 
(used when no audio is played) to fire OnTimeSync from either system or 
interrupt thread thus increasing the performance of non-audio presentations.

-> Made audio session device setup callback interrupt only since it is used 
to create the audio device on interrupt time thread for performance 
reasons.  The callback used to be InterruptSafe which meant it could fire 
on either system or interrupt time thread.  When fired on system thread, 
the devise set-up callback would reschedule until interrupt time was 
obtained.  This resulted in delays of re-starting of audio device in some 
cases.

-> Removed assert from buffer manager insisting that per stream evet lists 
be empty on its initialization.  This is no longer the case as per stream 
event lists hold data not counted towards buffering in case of file-source 
playback (equivalent to transport buffer) and they are valid to be 
prepopulated before the buffer maneger is started.

-> From the player perspective, we are buffering only if we are not 
playing.  That is, even if one of the sources is buffering and the source 
did not pause the player, this is background buffering not relevant at 
player level.  This situation may arrise when source is introduced and 
buffering ahead of its due begin time thus its buffering not affecting the 
playback.

-> On OpenTrack in the player object, scheduled an immediate player 
callback (ProcessIdle) to speed up processing of the added source.

-> Removed time check within ProcessCurrentEvents in order to dispatch all 
queued events unconditionally.  The time check was unnecessary as it was 
already done at time of insertion into the queue.

-> Made OnBegin firing tolerance (granularity) be m_ulPlayerUpdateInterval 
as that is the frequency we can expect OnBegin firing to be considered and 
we prefer to be a bit early in OnBegin call rather than late.

-> Made HXPlayer::CalculateOnBeginTime the center of establishing the begin 
time for all cases as well as for establising the difference between player 
and renderer timem cooridinate system (pRendInfo->m_ulTimeDiff) which is 
non-zero in  case of live presentations.

-> Simplified overly verbose logic for applying the pRendInfo->m_ulTimeDiff 
to various time stamps.

-> For shared wallclock, corrected time-stam comparison to be indifferent 
to magnitude of time-stamps.  Previous logic would fail in case of 
time-stamp around.

-> When resetting the source start-time (normally done due to shared 
wall-clock synchronization), hndled the case when the source needing 
starting time rest back is already playing by delaying the source by the 
difference between its current effectve start time and newly desired start 
time.  This allows for proper synchronization of wall-clock synchrnoized 
source when one such source is introduced into the playback while other is 
already playing.

-> Corrected the logic for setting of the start-time for live streams 
within the source.  The previous logic was relying on receiving at least 
one packet from each stream of a live source before starting the rendering 
of the source in order to compute the minimum start-time and use it as a 
starting point for all streams. Otherwise, it would attent to adjust the 
start-time of the streams while the source is playing (being renderered) 
which is not possible.  Since it is possible for all streams not to arrive 
by the time the source rendering is started (e.g. based on wall-clock 
preroll), any change in start time after the rendering is started is now 
ignored.

-> Corrected deadlock in base video renderer possible in any threaded build 
using OS granularity boosting for enhancing the display blt accuracy.  The 
dealock occured when a decoder object, while holding onto a decode mutex 
called a Blt method to display a due frame and thus attenoting to acquire 
the display mutex.  At the same time, the renderer being in porcess of 
displaying a frame, releasing a stale video frame and thus activating the 
decoder explicitly (due to decoder thread not being started yet) to 
backfill the released frame with another decoded frame.  The deadlock is 
resolved by renderer envoking decoder to backfill for a released frame 
outside of the display mutex.


Files Modified:
/client/audiosvc/hxaudply.cpp,v
/client/audiosvc/hxaudses.cpp,v
/client/audiosvc/hxaudstr_new.cpp,v
/client/audiosvc/pub/hxaudply.h,v
/client/audiosvc/pub/hxaudses.h,v
/client/core/buffmgr.cpp,v
/client/core/hxflsrc.cpp,v
/client/core/hxplay.cpp,v
/client/core/hxsrc.cpp,v
/client/core/srcinfo.cpp,v
/client/core/srcinfo.h,v
/client/core/pub/hxplay.h,v
/datatype/common/vidrend/vidrend.cpp,v
/datatype/common/vidrend/vidrendf.cpp,v
/datatype/common/vidrend/pub/vidrend.h,v
/datatype/common/vidrend/pub/vidrendf.h,v
/datatype/rm/audio/renderer/rarender.cpp,v
/datatype/smil/renderer/Umakefil,v
/datatype/smil/renderer/smil2/Umakefil,v
/datatype/smil/renderer/smil2/smldoc.cpp,v
/datatype/smil/renderer/smil2/smlrendr.cpp,v
/common/build/BIF/helix.bif

Image Size and Heap Use impact (Client -Only):
insignificant

Platforms and Profiles Affected:
all

Distribution Libraries Affected:
none

Distribution library impact and planned action:
n/a

Platforms and Profiles Build Verified:
win32-i386-vc7, helix-client-all-defines

Platforms and Profiles Functionality verified:
win32