- use Py_ssize_t when dealing with python sequence sizes
- dont call PySequence_Size(py_b) in a loop (its slow).
- use faster sequence/float parsing in aud.Factory.filter
- define __BIG_ENDIAN__ or __LITTLE_ENDIAN__ with cmake & scons.
- ENDIAN_ORDER is now a define rather than a global short.
- replace checks like this with single ifdef: #if defined(__sgi) || defined (__sparc) || defined (__sparc__) || defined (__PPC__) || defined (__ppc__) || defined (__hppa__) || defined (__BIG_ENDIAN__)
- remove BKE_endian.h which isn't used
* Fix for high quality upsampling which was wrong.
* Fix for doppler effects which were calculated wrong for scenes.
* Improved animation evaluation at the beginning and end of a scene.
Now all sounds that are not mono but have 3D checked automatically get reduced to mono during BGE conversion time.
Also removed the now unneeded function sound_get_channels and added a missing header file to audaspace's CMakeLists.txt.
Important note: I used rsync to do the local merge, as "svn merge --reintegrate ^/branches/soc-2011-pepper" doesn't work with our svn server right now!
Final GSoC commit.
* Bugfix: Negative frames crashed
* Bugfix: JOS sample buffer size prediction error (wasted memory)
* Optimisation: for JOS upsampling (around 12 % difference measured here)
* Optimisation: Better filter for JOS resampling
* Bugfix: Error in relative 3D audio code.
* Removed Attenuation
* Bugfix: Multiple scenes in BGE lead to errors, BGE audio now all relative, to support multiple scenes.
High quality resampling on mixdown, linear for playback.
* Lots of improvements and fixes for the JOS resampler, now it works fine!
* High quality filter coefficients for the JOS resampler (sorry for the 5 MB source file).
* Fix for GE orientation bug. Note: moto uses x,y,z,w quaternion storage, while rest of blender uses w,x,y,z.
* Minor changes/fixes.
Adding a mono flag to mixdown non-mono sounds for 3D audio.
* Added mono sound loading.
* Bugfix: AUD_COMPARE_SPECS usage was wrong.
* Bugfix: JOS resampler = instead of ==.
* Bugfix: Change of a sound should apply settings in AUD_SequencerHandle.
* Bugfix: Memory leak when canceling open sound operator.
Improved waveform drawing in the sequencer.
* Drawing the waveform of a sequencer strip is now independent from whether the sound is cached or not.
* Improved drawing of the waveform in the sequencer (especially speed!).
* Making it possible to vertically zoom more in the sequencer to better see the waveform for lipsync.
* Fixed a bug which crashed blender on loading a sound file via ffmpeg.
Speaker objects fully functional!
Minor changes:
* Fixed three memory bugs found via valgrind.
* Fixed bug with jack transport crashing after file loading.
* Sound NLA Strips now start at CFRA instead of 0.
* Pepper depends on ffmpeg 0.7.1 or higher now, windows and mac build systems set to ffmpeg-0.8
* Fixed orientation retrieval in OpenAL device code.
* Added stopAll() method to AUD_IDevice (also for Python) and call it on BGE exit
* Changed BGE to use audaspace via native C++ instead over the C API.
* Made AUD_SequencerFactory and AUD_SequencerEntry thread safe.
* Changed sound caching into a flag which fixes problems on file loading, especially with undo.
* Removed unused parameter from sound_mute_scene_sound
* Fixed bug: changing FPS didn't update the sequencer sound positions.
* Fixed bug: Properties of sequencer strips weren't set correctly.
* Minor warning fixes.
Mixdown functionality.
* Mixdown possible via libsndfile and ffmpeg!
* Fixed some ffmpeg deprecation warnings
* Mixdown UI only shows working Container, Codec and Format combinations!
* Minor bugs and warnings fixed
Implementation of Julius O. Smith's resampling algorithm for high quality audio resampling.
The filter currently is a sinc filter with a 0.9 cutt-off and windowed by a kaiser window (beta = 10).
Also includes minor changes in the linear resampler.
* Minor audaspace library improvements.
* Considering location, velocity and orientation in AUD_SequencerReader and AUD_SequencerHandle.
* Bugfix: Maximum and Minimum volume weren't used before in the software device.
* Bugfix: Adding speaker objects via info space crashed.
* Listener settings now get updated in the audio system.
Implemented basic audio animation.
* AnimatableProperty: Propper cache writing and spline interpolation for reading (the solution for stair steps in audio animation)
* Animatable properties so far are: volume, pitch, panning
* Users note: Changing the pitch of a sound results in wrong seeking, due to the resulting playback length difference.
* Users note: Panning only works for mono sources, values are in the range [-2..2], this basically controls the angle of the sound, 0 is front, -1 left, 1 right and 2 and -2 are back. Typical stereo panning only supports [-1..1].
* Disabled animation of audio related ffmpeg output parameters.
* Scene Audio Panel: 3D Listener settings also for Renderer, new Volume property (animatable!), Update/Bake buttons for animation problems, moved sampling rate and channel count here
Main: Complete rewrite of the sequencer related audio code to support 3D Audio objects later and especially adressing the animation system problems (see mailing list if interested).
Note: Animation is not working yet, so with this commit volume animation doesn't work anymore, that's the next step.
Minor things:
* Changed AUD_Reference behaviour a little to be more usage safe.
* Fixed bug in AUD_I3DHandle: Missing virtual destructor
* Fixed enmus in AUD_Space.h
* Fixed a warning in rna_scene.c
* Removed an unneeded call in rna_sound.c
* Reviewed and improved the linear resampler. Now it should work pretty good also for special cases that caused errors previously.
* Fixed a crash in the GE when a sound actuator doesn't have a sound assigned.
* Corrected the OpenAL device's threading code. This is a bugfix for #27913, thanks to Juha Mäki-Kanto for helping to resolve this.
* Fixed a warning in AUD_DoubleReader.cpp
* Removed some unneeded includes
* Fixed a bug resulting in a crash when stopping a sound
* Fixed a bug where a NaN resulted in a horrible memory error
* Fixed a typo bug which caused crackling in audio playback and export
* Added memory debugging code (ifdefed)
- Sequencer dynamics: Now it's possible to change the output channels and the resampling quality also increased (previously maximum quality was 44,1 kHz)
- Changed two buffers to use ffmpeg allocation, not sure if that helps somehow.
- Implemented a nice rechanneling solution with unofficial speaker arrangement standards similar to what OpenAL soft has
- Renamend AUD_Channel in the C API to AUD_Handle
- Removed the unlogical 7.2 speaker configuration, that's a hardware only config
blender_add_lib now takes a separate include argument to suppress warnings in system includes (mostly ffmpeg & python).
also only build wm_apple.c on apple+carbon configuration.
Added central compatibility header file, which enables blender to compile
against very old ffmpeg versions as well as very new versions using the
*NEW* API. (Old API functions are simulated using macros and inline functions)
Added a whole lot of additional checks, tested against 6 different versions
down the timeline, hopefully, now finally all is well.
* removed a lot of old cruft code for ancient ffmpeg versions
* made it compile again against latest ffmpeg / libav GIT
(also shouldn't break distro ffmpegs, since those API changes
have been introduced over a year ago. If it nevertheless breaks,
please send me an email)
FFMPEG was reallocating buffers it didn't own and wasn't allowed to. This workaround should work now flawlessly.
Also fixing a bug regarding unpacking sounds, the UI stated unpacking to //audio/filename while it was unpacking to //sounds/filename
* Adding play method to the device classes to play back a reader (not used yet, preparation for a later feature).
* Using a linear resampler in case SRC is disabled.
* [#25638] 'Insufficient thread locking' for sounds
- Actually a workaround for an msvc bug, msvc STL containers are buggy
* [#25922] Sound does not play in BlenderPlayer(r34579)
- Windows OpenAL doesn't seem to have clear context error state on initialising
This fixes a rather subtle seeking issue with ffmpeg and Sony
XDCAM-footage.
Problem is: MPEG2 streams within an MP4 container can contain a start
time - at several places. There is a starttime within the video
and audio streams and one within the container.
FFMpeg commandline tool only uses the container starttime and we used
the stream starttime.
The world would be a better place, if those two timestamps always match
up, since in XDCAM-footage those two starttimes differ in 4
frames - and the container has the right one.
We now always use the container start time as ffmpeg commandline tool
does (in the hope, that there is a good explaination for this and this
is the right thing(tm) to do).
I tested this also with HDV footage, which seems to work with the new
code, too.
Additional fix: disabled seek_by_bytes again, since it will only work
correctly, if ffmpeg guessed the HDV bitrate right (which it doesn't).
If you have seeking issues with HDV and have an older version of ffmpeg
installed, please upgrade, newer versions have some fixes in them.
globbing vs explicit is discussed here.
http://www.cmake.org/pipermail/cmake/2008-December/025694.html
Practical implications are:
- developers need to keep CMakeLists.txt files up to date.
- Users wont get strange linking errors if they build after a file is added, since CMake detects CMakeLists.txt is modified and automatically reconfigure.
