Fixes#135897 by introducing a mixing thread to coreaudio, making the callback lock free. At the same time these changes unify all devices that use a mixing thread (now that's jack, pulse, pipewire and coreaudio).
Pull Request: https://projects.blender.org/blender/blender/pulls/136711
Removes the playback handle from the synchronizer API and integrates it
into the device, removing the ISynchronizer interface completely.
This has been discussed in more detail in #126047 and its main purpose is to unify the handling of synchronizer events, especially seek which needed different logic for jack vs other backends.
OpenAL now needs a silence playback handle for synchronization but all
other backends are pretty straightforward just counting the mixed
samples in the SoftwareDevice with some specializations for specific
backends that had their own synchronizers before.
Note: CoreAudio changes are untested as I don't have a Mac.
Pull Request: https://projects.blender.org/blender/blender/pulls/133191
There's a "Render -> Render Audio" operator that pretty much calls into Audaspace functionality directly. This PR adds two new options in there, that boThere's a "Render -> Render Audio" operator that pretty much calls
into Audaspace functionality directly. This PR adds two new options in
there, that both feel like an oversight:
- There's an option for AAC container, when someone wants to render
into .aac file. Previously you could render AAC audio, but only in
MKV (Matroska) container. This is a user request #131980.
- When using MKV (Matroska) or Ogg container, it did not list Opus as
an audio codec that can be used. This felt like an oversight; both
the rest of Blender and Audaspace can handle that, jus the option
was not spelled out in the RNA enums.
Pull Request: https://projects.blender.org/blender/blender/pulls/132877th feel like an oversight:
- There's an option for AAC container, when someone wants to render into `.aac` file. Previously you _could_ render AAC audio, but only in MKV (Matroska) container. This is a user request #131980.
- When using MKV (Matroska) or Ogg container, it did not list Opus as an audio codec that can be used. This felt like an oversight; both the rest of Blender and Audaspace can handle that, jus the option was not spelled out in the RNA enums. Note however that without fix in #132872 the Matroska container continues to be useless in that it produces wrong result.
Upstream Audaspace PR for the 2nd point: https://github.com/neXyon/audaspace/pull/47 (was just merged!)
Pull Request: https://projects.blender.org/blender/blender/pulls/132877
Previously in Audaspace there was choice between linear resampler (okay
for preview, but not great for final mix), or "extremely high quality
preset" for rendering the final mix; with nothing in between. I have just
landed "medium" and "low" resampler quality levels in upstream Audaspace
(see https://github.com/audaspace/audaspace/pull/18 with details and
quality spectograms, also comparison with Audacity resampler).
This PR updates Audaspace to latest upstream, and switches to use the newly
added "medium" quality resampler. There's no audible difference (nor visible
one in spectrograms), as far as I can tell.
Timings, rendering out frames 1000-3000 of Sprite Fright Edit blender
studio data set:
- Windows (Ryzen 5950X, VS2022): 92 -> 73 sec
- Mac (M1 Max, clang 15): 70 -> 62 sec
i.e. using a faster audio resampler makes the _whole render process_ be
10-20% faster (however, this from VSE where it combines already pre-rendered
image strips).
Pull Request: https://projects.blender.org/blender/blender/pulls/116059
`DeviceManager.h` uses `std::string` without explicitly including
the `<string>` header. While older MSVC implicitly included this
header somewhere, the headers for 17.4+ do not leading to a build
error.
On the blender side this commit fixes importing video files with audio
and video streams that do not share the same start time and duration.
Differential Revision: https://developer.blender.org/D12353
The duration and start time for audio strips were not correctly read in
audaspace.
Some video files have a "lead in" section of audio that plays before the
video starts playing back. Before this patch, we would play this lead in
audio at the same time as the video started and thus the audio would not
be in sync anymore.
Now the lead in audio is cut off and the duration should be correctly
calculated with this in mind.
If the audio starts after the video, the audio strip is shifted to
account for this, but it will also lead to cut off audio which might not
be wanted. However we don't have a simple way to solve this at this
point.
Differential Revision: http://developer.blender.org/D11917
The issue was that sounds were always faded from 0 volume when they
started and depending on the currently used buffer size, the fading took
longer or shorter.
The solution stores whether the sound has ever been played back and
consequently does not fade when starting to play back.
- Changing API for time values from float to double for better precision.
- Fixing minor mistakes in the documentation.
- Fixing minor unnecessary large memory allocation.
- Fixed uninitialized result used in DynamicMusic::seek().
The comment to this function says false is returned if the handle
is invalid, while in practice non-initialized value will be returned.
- Spelling typos in comment.
- Silence -Wdelete-non-abstract-non-virtual-dtor warning.
Differential Revision: https://developer.blender.org/D6896
- Silence now has an optional sample rate parameter.
- Fix: wrong length reported by modulator and superpose.
- Minor formatting, include and documentation fixes.
