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authorskal <pascal.massimino@gmail.com>2026-05-20 22:44:44 +0200
committerskal <pascal.massimino@gmail.com>2026-05-20 22:44:44 +0200
commit5d20c892dedce7bc7486acbd72fbd35da69e413e (patch)
tree05e04d5e689504c2421cd5772e91a42ee69608ab /src/audio/ring_buffer.h
parent6ef8f578817ee0134fd5867ca3b80590e3eb2368 (diff)
fix: code review cleanup — bugs, dead code, factorization (-167 lines)
Bugs: - B1: fix dead tempo debug (prev_tempo captured after assignment) - B2: fix ReloadAssetsFromFile leak for disk-loaded assets; simplify DropAsset - B3: fix get_free_pool_slot leak (unregister synth + free data on reuse) - B4: volatile -> std::atomic with acquire/release in miniaudio_backend, synth - B5: fix unaligned reads in scene_loader (memcpy-based read_f32/read_u32) - B6: fix shader module + BGL + pipeline layout leaks in gpu.cc, pipeline_builder Dead code: - D1: remove unused particle_defs.h - D3: remove create_post_process_pipeline_simple (zero callers) - D4: remove empty gpu_draw() - D5: remove write-only Hybrid3D::initialized_ - D6: remove legacy pending buffer path in audio.cc Factorization: - F1: Effect::run_fullscreen_pass() replaces boilerplate in 5 effects - F2: particle_common.wgsl snippet, #include in 3 WGSL shaders - F3: gpu_create_shader_module() helper, used in 3 call sites - F5: get_world_aabb() shared between bvh.cc and physics.cc - F6: samples_to_seconds() replaces 6 inline expressions - F7: gpu_create_linear/nearest_sampler use SamplerCache; add nearest() preset 37/37 tests passing. handoff(Claude): code review batch — all items verified, no regressions.
Diffstat (limited to 'src/audio/ring_buffer.h')
-rw-r--r--src/audio/ring_buffer.h5
1 files changed, 5 insertions, 0 deletions
diff --git a/src/audio/ring_buffer.h b/src/audio/ring_buffer.h
index 1a21542..ba8cb49 100644
--- a/src/audio/ring_buffer.h
+++ b/src/audio/ring_buffer.h
@@ -19,6 +19,11 @@
RING_BUFFER_CHANNELS) / \
1000)
+// Convert sample count to seconds (accounting for stereo channels)
+inline float samples_to_seconds(int64_t samples) {
+ return (float)samples / (RING_BUFFER_SAMPLE_RATE * RING_BUFFER_CHANNELS);
+}
+
class AudioRingBuffer {
public:
AudioRingBuffer();