AudioServer+Userland: Decouple client sample rates from device rate

This change was a long time in the making ever since we obtained sample
rate awareness in the system. Now, each client has its own sample rate,
accessible via new IPC APIs, and the device sample rate is only
accessible via the management interface. AudioServer takes care of
resampling client streams into the device sample rate. Therefore, the
main improvement introduced with this commit is full responsiveness to
sample rate changes; all open audio programs will continue to play at
correct speed with the audio resampled to the new device rate.

The immediate benefits are manifold:
- Gets rid of the legacy hardware sample rate IPC message in the
  non-managing client
- Removes duplicate resampling and sample index rescaling code
  everywhere
- Avoids potential sample index scaling bugs in SoundPlayer (which have
  happened many times before) and fixes a sample index scaling bug in
  aplay
- Removes several FIXMEs
- Reduces amount of sample copying in all applications (especially
  Piano, where this is critical), improving performance
- Reduces number of resampling users, making future API changes (which
  will need to happen for correct resampling to be implemented) easier

I also threw in a simple race condition fix for Piano's audio player
loop.

Userland/Libraries/LibWeb/Platform/AudioCodecPlugin.cpp

@@ -28,7 +28,7 @@ ErrorOr<NonnullOwnPtr<AudioCodecPlugin>> AudioCodecPlugin::create(NonnullRefPtr<
     return s_creation_hook(move(loader));
 }
 
-ErrorOr<FixedArray<Audio::Sample>> AudioCodecPlugin::read_samples_from_loader(Audio::Loader& loader, size_t samples_to_load, size_t device_sample_rate)
+ErrorOr<FixedArray<Audio::Sample>> AudioCodecPlugin::read_samples_from_loader(Audio::Loader& loader, size_t samples_to_load)
 {
     auto buffer_or_error = loader.get_more_samples(samples_to_load);
     if (buffer_or_error.is_error()) {
@@ -36,30 +36,26 @@ ErrorOr<FixedArray<Audio::Sample>> AudioCodecPlugin::read_samples_from_loader(Au
         return Error::from_string_literal("Error while loading samples");
     }
 
-    Audio::ResampleHelper<Audio::Sample> resampler(loader.sample_rate(), device_sample_rate);
-    return FixedArray<Audio::Sample>::create(resampler.resample(buffer_or_error.release_value()).span());
+    return buffer_or_error.release_value();
 }
 
-Duration AudioCodecPlugin::set_loader_position(Audio::Loader& loader, double position, Duration duration, size_t device_sample_rate)
+Duration AudioCodecPlugin::set_loader_position(Audio::Loader& loader, double position, Duration duration)
 {
     if (loader.total_samples() == 0)
-        return current_loader_position(loader, device_sample_rate);
+        return current_loader_position(loader);
 
     auto duration_value = static_cast<double>(duration.to_milliseconds()) / 1000.0;
     position = position / duration_value * static_cast<double>(loader.total_samples() - 1);
 
     loader.seek(static_cast<int>(position)).release_value_but_fixme_should_propagate_errors();
-    return current_loader_position(loader, device_sample_rate);
+    return current_loader_position(loader);
 }
 
-Duration AudioCodecPlugin::current_loader_position(Audio::Loader const& loader, size_t device_sample_rate)
+Duration AudioCodecPlugin::current_loader_position(Audio::Loader const& loader)
 {
     auto samples_played = static_cast<double>(loader.loaded_samples());
     auto sample_rate = static_cast<double>(loader.sample_rate());
 
-    auto source_to_device_ratio = sample_rate / static_cast<double>(device_sample_rate);
-    samples_played *= source_to_device_ratio;
-
     return Duration::from_milliseconds(static_cast<i64>(samples_played / sample_rate * 1000.0));
 }