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author | kleines Filmröllchen <malu.bertsch@gmail.com> | 2021-08-27 23:47:09 +0200 |
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committer | Andreas Kling <kling@serenityos.org> | 2021-09-12 23:38:57 +0200 |
commit | 152ec28da0d1a51ade18fe6799a34ac9b7aa07b0 (patch) | |
tree | 540e62552da0d9bb767e566f8951f081009e136a /Userland/Applications/Piano | |
parent | 2909c3a931439d0cbed3e4599d9eed47a6fdb446 (diff) | |
download | serenity-152ec28da0d1a51ade18fe6799a34ac9b7aa07b0.zip |
Audio: Change how volume works
Across the entire audio system, audio now works in 0-1 terms instead of
0-100 as before. Therefore, volume is now a double instead of an int.
The master volume of the AudioServer changes smoothly through a
FadingProperty, preventing clicks. Finally, volume computations are done
with logarithmic scaling, which is more natural for the human ear.
Note that this could be 4-5 different commits, but as they change each
other's code all the time, it makes no sense to split them up.
Diffstat (limited to 'Userland/Applications/Piano')
-rw-r--r-- | Userland/Applications/Piano/Music.h | 3 | ||||
-rw-r--r-- | Userland/Applications/Piano/Track.cpp | 7 |
2 files changed, 7 insertions, 3 deletions
diff --git a/Userland/Applications/Piano/Music.h b/Userland/Applications/Piano/Music.h index a1080155e9..ce2c814735 100644 --- a/Userland/Applications/Piano/Music.h +++ b/Userland/Applications/Piano/Music.h @@ -30,7 +30,8 @@ constexpr int buffer_size = sample_count * sizeof(Sample); constexpr double sample_rate = 44100; -constexpr double volume_factor = 1800; +// Headroom for the synth +constexpr double volume_factor = 0.1; enum Switch { Off, diff --git a/Userland/Applications/Piano/Track.cpp b/Userland/Applications/Piano/Track.cpp index f077efc994..5727a945f7 100644 --- a/Userland/Applications/Piano/Track.cpp +++ b/Userland/Applications/Piano/Track.cpp @@ -95,8 +95,8 @@ void Track::fill_sample(Sample& sample) default: VERIFY_NOT_REACHED(); } - new_sample.left += note_sample.left * m_power[note] * volume_factor * (static_cast<double>(volume()) / volume_max); - new_sample.right += note_sample.right * m_power[note] * volume_factor * (static_cast<double>(volume()) / volume_max); + new_sample.left += note_sample.left * m_power[note] * NumericLimits<i16>::max() * volume_factor * (static_cast<double>(volume()) / volume_max); + new_sample.right += note_sample.right * m_power[note] * NumericLimits<i16>::max() * volume_factor * (static_cast<double>(volume()) / volume_max); } auto new_sample_dsp = LibDSP::Signal(LibDSP::Sample { new_sample.left / NumericLimits<i16>::max(), new_sample.right / NumericLimits<i16>::max() }); @@ -105,6 +105,9 @@ void Track::fill_sample(Sample& sample) new_sample.left = delayed_sample.left * NumericLimits<i16>::max(); new_sample.right = delayed_sample.right * NumericLimits<i16>::max(); + new_sample.left = clamp(new_sample.left, NumericLimits<i16>::min(), NumericLimits<i16>::max()); + new_sample.right = clamp(new_sample.right, NumericLimits<i16>::min(), NumericLimits<i16>::max()); + sample.left += new_sample.left; sample.right += new_sample.right; } |