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-rw-r--r--src/audio/dct.h4
-rw-r--r--src/audio/fft.cc6
-rw-r--r--src/audio/mp3_sample.cc5
-rw-r--r--src/audio/mp3_sample.h2
-rw-r--r--src/audio/ola.h3
-rw-r--r--src/audio/synth.cc3
-rw-r--r--src/audio/synth.h4
-rw-r--r--src/audio/tracker.cc16
8 files changed, 24 insertions, 19 deletions
diff --git a/src/audio/dct.h b/src/audio/dct.h
index ca423c2..a44bb15 100644
--- a/src/audio/dct.h
+++ b/src/audio/dct.h
@@ -4,9 +4,9 @@
#pragma once
-#define DCT_SIZE 512
+#define DCT_SIZE 512
#define OLA_HOP_SIZE 256
-#define OLA_OVERLAP 256
+#define OLA_OVERLAP 256
// Forward declarations
void fdct_512(const float* input, float* output);
diff --git a/src/audio/fft.cc b/src/audio/fft.cc
index 6029454..ddd442e 100644
--- a/src/audio/fft.cc
+++ b/src/audio/fft.cc
@@ -143,7 +143,8 @@ void dct_fft(const float* input, float* output, size_t N) {
// IMDCT via FFT
// Produces 2N time-domain samples from N MDCT coefficients.
// Formula: x[n] = (2/N) * sum_{k=0}^{N-1} X[k] * cos(pi*(2n+1+N)*(2k+1)/(2N))
-// When windowed (Hann, length 2N) and OLA'd with hop N, gives perfect reconstruction.
+// When windowed (Hann, length 2N) and OLA'd with hop N, gives perfect
+// reconstruction.
void imdct_fft(const float* input, float* output, size_t N) {
const float PI = 3.14159265358979323846f;
const size_t M = 2 * N; // output length
@@ -172,7 +173,8 @@ void imdct_fft(const float* input, float* output, size_t N) {
imag[i] *= scale;
}
- // Post-multiply by 2N * exp(-j*pi*(2n+1)/(4N)) and take real part, scale by 2/N
+ // Post-multiply by 2N * exp(-j*pi*(2n+1)/(4N)) and take real part, scale by
+ // 2/N
const float gain = 2.0f;
for (size_t n = 0; n < M; n++) {
const float angle = -PI * (2.0f * n + 1.0f) / (4.0f * N);
diff --git a/src/audio/mp3_sample.cc b/src/audio/mp3_sample.cc
index 028fbff..9a3805b 100644
--- a/src/audio/mp3_sample.cc
+++ b/src/audio/mp3_sample.cc
@@ -45,9 +45,10 @@ int mp3_decode(Mp3Decoder* dec, int num_frames, float* out) {
}
void mp3_close(Mp3Decoder* dec) {
- if (!dec) return;
+ if (!dec)
+ return;
ma_decoder_uninit(&dec->dec);
delete dec;
}
-#endif // !STRIP_ALL
+#endif // !STRIP_ALL
diff --git a/src/audio/mp3_sample.h b/src/audio/mp3_sample.h
index e7e759e..bf3147a 100644
--- a/src/audio/mp3_sample.h
+++ b/src/audio/mp3_sample.h
@@ -31,4 +31,4 @@ int mp3_decode(Mp3Decoder* dec, int num_frames, float* out);
// Release the decoder.
void mp3_close(Mp3Decoder* dec);
-#endif // !STRIP_ALL
+#endif // !STRIP_ALL
diff --git a/src/audio/ola.h b/src/audio/ola.h
index 3dbc368..2d6267e 100644
--- a/src/audio/ola.h
+++ b/src/audio/ola.h
@@ -7,8 +7,7 @@
// Returns number of OLA frames for n_samples PCM input.
static inline int ola_num_frames(int n_samples) {
- return (n_samples > DCT_SIZE) ? (n_samples - DCT_SIZE) / OLA_HOP_SIZE + 1
- : 1;
+ return (n_samples > DCT_SIZE) ? (n_samples - DCT_SIZE) / OLA_HOP_SIZE + 1 : 1;
}
// Hann-windowed FDCT with 50% overlap (analysis).
diff --git a/src/audio/synth.cc b/src/audio/synth.cc
index 3212e0b..0866bda 100644
--- a/src/audio/synth.cc
+++ b/src/audio/synth.cc
@@ -204,7 +204,8 @@ void synth_trigger_voice(int spectrogram_id, float volume, float pan,
g_synth_data.spectrograms[spectrogram_id].num_frames;
v.ola_mode = (g_synth_data.spectrograms[spectrogram_id].version ==
SPEC_VERSION_V2_OLA);
- v.buffer_pos = v.ola_mode ? OLA_HOP_SIZE : DCT_SIZE; // Force reload on first render
+ v.buffer_pos =
+ v.ola_mode ? OLA_HOP_SIZE : DCT_SIZE; // Force reload on first render
if (v.ola_mode)
memset(v.overlap_buf, 0, sizeof(v.overlap_buf));
v.fractional_pos =
diff --git a/src/audio/synth.h b/src/audio/synth.h
index 61ecfd0..e5a8197 100644
--- a/src/audio/synth.h
+++ b/src/audio/synth.h
@@ -21,8 +21,8 @@
#define MAX_SPECTROGRAMS \
32 // Current track: 14 unique, 32 provides comfortable headroom
-#define SPEC_VERSION_V1 1
-#define SPEC_VERSION_V2_OLA 2
+#define SPEC_VERSION_V1 1
+#define SPEC_VERSION_V2_OLA 2
struct Spectrogram {
const float* spectral_data_a; // Front buffer
diff --git a/src/audio/tracker.cc b/src/audio/tracker.cc
index 7ce96fc..93c279e 100644
--- a/src/audio/tracker.cc
+++ b/src/audio/tracker.cc
@@ -12,7 +12,7 @@
#include "audio/dct.h"
#include "audio/mp3_sample.h"
#include "audio/window.h"
-#endif // !defined(STRIP_ALL)
+#endif // !defined(STRIP_ALL)
static uint32_t g_last_trigger_idx = 0;
@@ -44,7 +44,6 @@ static bool g_cache_initialized = false;
// Forward declarations
static int get_free_pool_slot();
-
#if !defined(STRIP_ALL)
// Decode an in-memory MP3 blob to a heap-allocated spectrogram (caller owns).
// Uses OLA analysis: 512-sample Hann window, OLA_HOP_SIZE advance per frame.
@@ -53,7 +52,8 @@ static float* convert_mp3_to_spectrogram(const uint8_t* data, size_t size,
int* out_num_frames) {
*out_num_frames = 0;
Mp3Decoder* dec = mp3_open(data, size);
- if (!dec) return nullptr;
+ if (!dec)
+ return nullptr;
float window[DCT_SIZE];
hann_window_512(window);
@@ -82,11 +82,13 @@ static float* convert_mp3_to_spectrogram(const uint8_t* data, size_t size,
fdct_512(pcm_chunk, dct_chunk);
spec_data.insert(spec_data.end(), dct_chunk, dct_chunk + DCT_SIZE);
- if (decoded == 0) break;
+ if (decoded == 0)
+ break;
}
mp3_close(dec);
- if (spec_data.empty()) return nullptr;
+ if (spec_data.empty())
+ return nullptr;
const int num_frames = (int)(spec_data.size() / DCT_SIZE);
float* result = new float[spec_data.size()];
@@ -94,7 +96,7 @@ static float* convert_mp3_to_spectrogram(const uint8_t* data, size_t size,
*out_num_frames = num_frames;
return result;
}
-#endif // !defined(STRIP_ALL)
+#endif // !defined(STRIP_ALL)
void tracker_init() {
g_last_trigger_idx = 0;
@@ -159,7 +161,7 @@ void tracker_init() {
FATAL_CHECK(data == nullptr || GetAssetType(aid) != AssetType::MP3,
"MP3 assets not supported in STRIP_ALL builds\n");
#endif
- if (data && size >= sizeof(SpecHeader)) {
+ if (data && size >= sizeof(SpecHeader)) {
const SpecHeader* header = (const SpecHeader*)data;
const int note_frames = header->num_frames;
const float* spectral_data =
generated by cgit v1.2.3 (git 2.51.0) at 2026-03-22 23:32:40 +0000