From 24c2930012baa60379da6f8c23f2cf35e38ef2d2 Mon Sep 17 00:00:00 2001
From: David Marcec <dmarcecguzman@gmail.com>
Date: Wed, 1 Jul 2020 16:52:02 +1000
Subject: [PATCH] audio: Improving audio timing for multicore/single core

Fixes the issue with needing the timestretcher for multicore.
---
 src/audio_core/stream.cpp | 23 +++++------------------
 src/audio_core/stream.h   |  4 ++--
 2 files changed, 7 insertions(+), 20 deletions(-)

diff --git a/src/audio_core/stream.cpp b/src/audio_core/stream.cpp
index dfc4805d97..307d78ecd1 100644
--- a/src/audio_core/stream.cpp
+++ b/src/audio_core/stream.cpp
@@ -38,7 +38,7 @@ Stream::Stream(Core::Timing::CoreTiming& core_timing, u32 sample_rate, Format fo
       sink_stream{sink_stream}, core_timing{core_timing}, name{std::move(name_)} {
 
     release_event = Core::Timing::CreateEvent(
-        name, [this](u64 userdata, s64 cycles_late) { ReleaseActiveBuffer(); });
+        name, [this](u64 userdata, s64 cycles_late) { ReleaseActiveBuffer(cycles_late); });
 }
 
 void Stream::Play() {
@@ -66,15 +66,6 @@ s64 Stream::GetBufferReleaseNS(const Buffer& buffer) const {
     return ns.count();
 }
 
-s64 Stream::GetBufferReleaseNSHostTiming(const Buffer& buffer) const {
-    const std::size_t num_samples{buffer.GetSamples().size() / GetNumChannels()};
-    /// DSP signals before playing the last sample, in HLE we emulate this in this way
-    s64 base_samples = std::max<s64>(static_cast<s64>(num_samples) - 1, 0);
-    const auto ns =
-        std::chrono::nanoseconds((static_cast<u64>(base_samples) * 1000000000ULL) / sample_rate);
-    return ns.count();
-}
-
 static void VolumeAdjustSamples(std::vector<s16>& samples, float game_volume) {
     const float volume{std::clamp(Settings::Volume() - (1.0f - game_volume), 0.0f, 1.0f)};
 
@@ -89,7 +80,7 @@ static void VolumeAdjustSamples(std::vector<s16>& samples, float game_volume) {
     }
 }
 
-void Stream::PlayNextBuffer() {
+void Stream::PlayNextBuffer(s64 cycles_late) {
     if (!IsPlaying()) {
         // Ensure we are in playing state before playing the next buffer
         sink_stream.Flush();
@@ -114,18 +105,14 @@ void Stream::PlayNextBuffer() {
 
     sink_stream.EnqueueSamples(GetNumChannels(), active_buffer->GetSamples());
 
-    if (core_timing.IsHostTiming()) {
-        core_timing.ScheduleEvent(GetBufferReleaseNSHostTiming(*active_buffer), release_event, {});
-    } else {
-        core_timing.ScheduleEvent(GetBufferReleaseNS(*active_buffer), release_event, {});
-    }
+    core_timing.ScheduleEvent(GetBufferReleaseNS(*active_buffer) - cycles_late, release_event, {});
 }
 
-void Stream::ReleaseActiveBuffer() {
+void Stream::ReleaseActiveBuffer(s64 cycles_late) {
     ASSERT(active_buffer);
     released_buffers.push(std::move(active_buffer));
     release_callback();
-    PlayNextBuffer();
+    PlayNextBuffer(cycles_late);
 }
 
 bool Stream::QueueBuffer(BufferPtr&& buffer) {
diff --git a/src/audio_core/stream.h b/src/audio_core/stream.h
index e309d60fe3..5243762575 100644
--- a/src/audio_core/stream.h
+++ b/src/audio_core/stream.h
@@ -90,10 +90,10 @@ public:
 
 private:
     /// Plays the next queued buffer in the audio stream, starting playback if necessary
-    void PlayNextBuffer();
+    void PlayNextBuffer(s64 cycles_late = 0);
 
     /// Releases the actively playing buffer, signalling that it has been completed
-    void ReleaseActiveBuffer();
+    void ReleaseActiveBuffer(s64 cycles_late = 0);
 
     /// Gets the number of core cycles when the specified buffer will be released
     s64 GetBufferReleaseNS(const Buffer& buffer) const;