AudioCore: Use nanoseconds instead of cycles for buffer time.

1 files changed, 5 insertions, 5 deletions

diff --git a/src/audio_core/stream.cpp b/src/audio_core/stream.cpp
index ca7cfb030..6d5539b6b 100644
--- a/src/audio_core/stream.cpp
+++ b/src/audio_core/stream.cpp
@@ -59,11 +59,11 @@ Stream::State Stream::GetState() const {
     return state;
 }
 
-s64 Stream::GetBufferReleaseCycles(const Buffer& buffer) const {
+s64 Stream::GetBufferReleaseNS(const Buffer& buffer) const {
     const std::size_t num_samples{buffer.GetSamples().size() / GetNumChannels()};
-    const auto us =
-        std::chrono::microseconds((static_cast<u64>(num_samples) * 1000000) / sample_rate);
-    return Core::Timing::usToCycles(us);
+    const auto ns =
+        std::chrono::nanoseconds((static_cast<u64>(num_samples) * 1000000000ULL) / sample_rate);
+    return ns.count();
 }
 
 static void VolumeAdjustSamples(std::vector<s16>& samples, float game_volume) {
@@ -105,7 +105,7 @@ void Stream::PlayNextBuffer() {
 
     sink_stream.EnqueueSamples(GetNumChannels(), active_buffer->GetSamples());
 
-    core_timing.ScheduleEvent(GetBufferReleaseCycles(*active_buffer), release_event, {});
+    core_timing.ScheduleEvent(GetBufferReleaseNS(*active_buffer), release_event, {});
 }
 
 void Stream::ReleaseActiveBuffer() {