wip on ref audio, but it compiles

This commit is contained in:
Concedo
2026-03-12 23:46:10 +08:00
parent d5a4c17e14
commit 8f23b8d81e
8 changed files with 549 additions and 33 deletions
+1
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@@ -351,6 +351,7 @@ struct music_generation_inputs
const bool gen_codes = false;
const bool rewrite_caption = true;
const char * input_json = nullptr;
const char * music_reference_audio_data = nullptr;
};
struct music_generation_outputs
{
+5 -1
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@@ -464,7 +464,8 @@ class music_generation_inputs(ctypes.Structure):
("use_mp3", ctypes.c_bool),
("gen_codes", ctypes.c_bool),
("rewrite_caption", ctypes.c_bool),
("input_json", ctypes.c_char_p)]
("input_json", ctypes.c_char_p),
("music_reference_audio_data", ctypes.c_char_p)]
class music_generation_outputs(ctypes.Structure):
_fields_ = [("status", ctypes.c_int),
@@ -2490,6 +2491,7 @@ def music_generate_codes(genparams):
inputs.gen_codes = genparams.get('gen_codes', False)
inputs.rewrite_caption = genparams.get('rewrite_caption', True)
inputs.input_json = input_json.encode("UTF-8")
inputs.music_reference_audio_data = "".encode("UTF-8")
ret = handle.music_generate(inputs)
outstr = ""
if ret.status==1:
@@ -2507,6 +2509,8 @@ def music_generate_audio(genparams):
inputs.gen_codes = genparams.get('gen_codes', False)
inputs.rewrite_caption = genparams.get('rewrite_caption', True)
inputs.input_json = input_json.encode("UTF-8")
refaudio = genparams.get('music_reference_audio_data', None)
inputs.music_reference_audio_data = (refaudio.encode("UTF-8") if (refaudio and refaudio!="") else "".encode("UTF-8"))
ret = handle.music_generate(inputs)
outstr = ""
if ret.status==1:
+2 -2
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@@ -528,10 +528,10 @@ struct MetadataFSM {
language_name = bpe_encode(&bpe, "language:", false);
timesig_name = bpe_encode(&bpe, "timesignature:", false);
// BPM 30-300
// BPM 30-195
{
std::vector<std::string> vals;
for (int v = 30; v <= 300; v++) vals.push_back(std::to_string(v));
for (int v = 30; v <= 195; v++) vals.push_back(std::to_string(v));
build_value_tree(bpe, bpm_tree, "bpm:", vals);
}
// Duration 40-450
+125 -29
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@@ -21,6 +21,8 @@
#include "./bpe.h"
#include "./debug.h"
#include "./request.h"
#include "./vae-enc.h"
#include "otherarch/utils.h"
// Minimal WAV writer (16-bit PCM stereo)
static bool write_wav(const char * path, const float * audio, int T_audio, int sr) {
@@ -580,6 +582,7 @@ static DiTGGMLConfig music_dit_cfg;
static Timer music_dit_timer;
static bool is_turbo = false;
static VAEGGML vae = {};
static VAEEncoder vae_enc = {};
static BPETokenizer music_tok;
static Qwen3GGML music_text_enc = {};
static GGUFModel gf_te = {};
@@ -606,6 +609,7 @@ void unload_acestep_dit_others()
if(acestep_dit_others_loaded)
{
acestep_dit_others_loaded = false;
vae_enc_free(&vae_enc);
vae_ggml_free(&vae);
gf_close(&gf_te);
cond_ggml_free(&music_cond);
@@ -672,6 +676,10 @@ bool load_acestep_dit(std::string music_embd_path, std::string music_dit_path, s
vae_ggml_load(&vae, vae_gguf);
fprintf(stderr, "[Load] VAE weights: %.1f ms\n", music_dit_timer.ms());
music_dit_timer.reset();
vae_enc_load(&vae_enc, vae_gguf);
fprintf(stderr, "[Load] VAE Enc weights: %.1f ms\n", music_dit_timer.ms());
music_dit_timer.reset();
if (!load_bpe_from_gguf(&music_tok, text_enc_gguf)) {
fprintf(stderr, "FATAL: failed to load music tokenizer from %s\n", text_enc_gguf);
@@ -733,6 +741,50 @@ std::string acestep_generate_audio(const music_generation_inputs inputs)
}
}
const int FRAMES_PER_SECOND = 25;
int Oc = music_dit_cfg.out_channels; // 64
int ctx_ch = music_dit_cfg.in_channels - Oc; // 128
int batch_n = 1;
int vae_chunk = 256;
int vae_overlap = 64;
// Cover mode: load VAE encoder and encode source audio
bool have_cover = false;
std::vector<float> cover_latents; // [T_cover, 64] time-major
int T_cover = 0;
std::string custom_reference_audio_str = inputs.music_reference_audio_data;
if (custom_reference_audio_str!="") {
music_dit_timer.reset();
int T_audio = 0, wav_sr = 0;
std::vector<uint8_t> media_data_buffer = kcpp_base64_decode(custom_reference_audio_str);
std::vector<float> custom_reference_audio_pcmf32;
bool ok = kcpp_decode_audio_from_buf(media_data_buffer.data(), media_data_buffer.size(), 48000, custom_reference_audio_pcmf32);
if (!ok) {
printf("\nError: Cannot read input audio file.\n");
return "";
}
wav_sr = 48000;
T_audio = custom_reference_audio_pcmf32.size();
float * wav_data = custom_reference_audio_pcmf32.data();
fprintf(stderr, "[Cover] Source audio: %.2fs\n", (float)T_audio / (float)(wav_sr > 0 ? wav_sr : 48000));
int max_T_lat = (T_audio / 1920) + 64;
cover_latents.resize(max_T_lat * 64);
T_cover = vae_enc_encode_tiled(&vae_enc, wav_data, T_audio,
cover_latents.data(), max_T_lat,
vae_chunk, vae_overlap);
if (T_cover < 0) {
fprintf(stderr, "FATAL: VAE encode of src_audio failed\n");
return "";
}
cover_latents.resize(T_cover * 64);
fprintf(stderr, "[Cover] Encoded: T_cover=%d (%.2fs), %.1f ms\n",
T_cover, (float)T_cover * 1920.0f / 48000.0f, music_dit_timer.ms());
have_cover = true;
}
// Parse request JSON
AceRequest req;
std::string injson = inputs.input_json;
@@ -748,13 +800,6 @@ std::string acestep_generate_audio(const music_generation_inputs inputs)
req.inference_steps = (req.inference_steps>100?100:req.inference_steps); //clamp to 100
req.duration = (req.duration>420?420:req.duration); //clamp to 7 min
const int FRAMES_PER_SECOND = 25;
int Oc = music_dit_cfg.out_channels; // 64
int ctx_ch = music_dit_cfg.in_channels - Oc; // 128
int batch_n = 1;
int vae_chunk = 256;
int vae_overlap = 64;
// Extract params
const char * caption = req.caption.c_str();
const char * lyrics = req.lyrics.empty() ? "[Instrumental]" : req.lyrics.c_str();
@@ -780,8 +825,6 @@ std::string acestep_generate_audio(const music_generation_inputs inputs)
{
seed = (((uint32_t)time(NULL)) % 1000000u);
}
fprintf(stderr, "[Pipeline] seed=%lld, steps=%d, guidance=%.1f, shift=%.1f, duration=%.1fs\n",
seed, num_steps, guidance_scale, shift, duration);
// Parse audio codes from request
std::vector<int> codes_vec = parse_codes_string(req.audio_codes);
@@ -797,16 +840,24 @@ std::string acestep_generate_audio(const music_generation_inputs inputs)
}
// T = number of 25Hz latent frames for DiT
// When audio codes are present, T is determined by the codes.
// Otherwise, T is derived from the requested duration.
int T = codes_vec.empty()
? (int)(duration * FRAMES_PER_SECOND)
: (int)codes_vec.size() * 5;
// Cover: from source audio. Codes: from code count. Else: from duration.
int T;
if (have_cover) {
T = T_cover;
// duration in metas must match actual source length, not JSON default
duration = (float)T_cover / (float)FRAMES_PER_SECOND;
} else if (!codes_vec.empty()) {
T = (int)codes_vec.size() * 5;
} else {
T = (int)(duration * FRAMES_PER_SECOND);
}
T = ((T + music_dit_cfg.patch_size - 1) / music_dit_cfg.patch_size) * music_dit_cfg.patch_size;
int S = T / music_dit_cfg.patch_size;
int enc_S = 0;
fprintf(stderr, "[Pipeline] T=%d, S=%d\n", T, S);
fprintf(stderr, "[Pipeline] seed=%lld, steps=%d, guidance=%.1f, shift=%.1f, duration=%.1fs\n",
seed, num_steps, guidance_scale, shift, duration);
if (T > 15000) {
fprintf(stderr, "ERROR: T=%d exceeds silence_latent max 15000, skipping\n", T);
@@ -818,7 +869,8 @@ std::string acestep_generate_audio(const music_generation_inputs inputs)
music_dit_timer.reset();
// 2. Build formatted prompts
const char * instruction = "Generate audio semantic tokens based on the given conditions:";
// Same instruction for all modes. Cover differs only by context content (audio vs silence).
const char * instruction = "Fill the audio semantic mask based on the given conditions:";
char metas[512];
snprintf(metas, sizeof(metas),
"- bpm: %s\n- timesignature: %s\n- keyscale: %s\n- duration: %d seconds\n",
@@ -873,10 +925,10 @@ std::string acestep_generate_audio(const music_generation_inputs inputs)
// std::vector<float> silence(Oc * T);
// memcpy(silence.data(), silence_full.data(), (size_t)(Oc * T) * sizeof(float));
// Decode audio codes if provided
// Decode audio codes if provided (passthrough mode only, NOT cover)
int decoded_T = 0;
std::vector<float> decoded_latents;
if (!codes_vec.empty()) {
if (!have_cover && !codes_vec.empty()) {
int T_5Hz = (int)codes_vec.size();
int T_25Hz_codes = T_5Hz * 5;
decoded_latents.resize(T_25Hz_codes * Oc);
@@ -892,16 +944,31 @@ std::string acestep_generate_audio(const music_generation_inputs inputs)
decoded_T = T_25Hz_codes < T ? T_25Hz_codes : T;
}
// Build single context: [T, ctx_ch] = src_latents[64] + mask_ones[64]
// Build context: [T, ctx_ch] = src_latents[64] + mask_ones[64]
// Cover: VAE latents directly (matching Python: is_covers=False, raw latents as context)
// Passthrough: detokenized FSQ codes + silence padding
// Text2music: silence only
std::vector<float> context_single(T * ctx_ch);
for (int t = 0; t < T; t++) {
const float * src = (t < decoded_T)
? decoded_latents.data() + t * Oc
: silence_full.data() + (t - decoded_T) * Oc;
for (int c = 0; c < Oc; c++)
context_single[t * ctx_ch + c] = src[c];
for (int c = 0; c < Oc; c++)
context_single[t * ctx_ch + Oc + c] = 1.0f;
if (have_cover) {
for (int t = 0; t < T; t++) {
const float * src = (t < T_cover)
? cover_latents.data() + t * Oc
: silence_full.data() + t * Oc;
for (int c = 0; c < Oc; c++)
context_single[t * ctx_ch + c] = src[c];
for (int c = 0; c < Oc; c++)
context_single[t * ctx_ch + Oc + c] = 1.0f;
}
} else {
for (int t = 0; t < T; t++) {
const float * src = (t < decoded_T)
? decoded_latents.data() + t * Oc
: silence_full.data() + (t - decoded_T) * Oc;
for (int c = 0; c < Oc; c++)
context_single[t * ctx_ch + c] = src[c];
for (int c = 0; c < Oc; c++)
context_single[t * ctx_ch + Oc + c] = 1.0f;
}
}
// Replicate context for N batch samples (all identical)
@@ -911,6 +978,32 @@ std::string acestep_generate_audio(const music_generation_inputs inputs)
memcpy(context.data() + b * T * ctx_ch, context_single.data(), T * ctx_ch * sizeof(float));
}
// Cover mode: build silence context for audio_cover_strength switching
// When step >= cover_steps, DiT switches from cover context to silence context
std::vector<float> context_silence;
int cover_steps = -1;
if (have_cover) {
float cover_strength = req.audio_cover_strength;
if (cover_strength < 1.0f) {
// Build silence context: all frames use silence_latent
std::vector<float> silence_single(T * ctx_ch);
for (int t = 0; t < T; t++) {
const float * src = silence_full.data() + t * Oc;
for (int c = 0; c < Oc; c++)
silence_single[t * ctx_ch + c] = src[c];
for (int c = 0; c < Oc; c++)
silence_single[t * ctx_ch + Oc + c] = 1.0f;
}
context_silence.resize(batch_n * T * ctx_ch);
for (int b = 0; b < batch_n; b++)
memcpy(context_silence.data() + b * T * ctx_ch,
silence_single.data(), T * ctx_ch * sizeof(float));
cover_steps = (int)((float)num_steps * cover_strength);
fprintf(stderr, "[Cover] audio_cover_strength=%.2f -> switch at step %d/%d\n",
cover_strength, cover_steps, num_steps);
}
}
// Generate N noise samples
std::vector<float> noise(batch_n * Oc * T);
@@ -929,13 +1022,16 @@ std::string acestep_generate_audio(const music_generation_inputs inputs)
// DiT Generate
std::vector<float> output(batch_n * Oc * T);
fprintf(stderr, "[DiT] Starting: T=%d, S=%d, enc_S=%d, steps=%d, batch=%d\n",
T, S, enc_S, num_steps, batch_n);
fprintf(stderr, "[DiT] Starting: T=%d, S=%d, enc_S=%d, steps=%d, batch=%d%s\n",
T, S, enc_S, num_steps, batch_n,
have_cover ? " (cover)" : "");
music_dit_timer.reset();
dit_ggml_generate(&acestep_dit, noise.data(), context.data(), enc_hidden.data(),
enc_S, T, batch_n, num_steps, schedule.data(), output.data(),
guidance_scale);
guidance_scale, nullptr,
context_silence.empty() ? nullptr : context_silence.data(),
cover_steps);
fprintf(stderr, "[DiT] Total generation: %.1f ms (%.1f ms/sample)\n",
music_dit_timer.ms(), music_dit_timer.ms() / batch_n);
+16 -1
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@@ -1084,7 +1084,9 @@ static void dit_ggml_generate(
const float * schedule,
float * output,
float guidance_scale = 1.0f,
const DebugDumper * dbg = nullptr) {
const DebugDumper * dbg = nullptr,
const float * context_switch = nullptr,
int cover_steps = -1) {
DiTGGMLConfig & c = model->cfg;
int Oc = c.out_channels; // 64
@@ -1241,9 +1243,22 @@ static void dit_ggml_generate(
struct ggml_tensor * t_t = ggml_graph_get_tensor(gf, "t");
// Flow matching loop
bool switched_cover = false;
for (int step = 0; step < num_steps; step++) {
float t_curr = schedule[step];
// Cover mode: switch context from cover to non-cover at cover_steps
if (context_switch && cover_steps >= 0 && step >= cover_steps && !switched_cover) {
switched_cover = true;
for (int b = 0; b < N; b++)
for (int t = 0; t < T; t++)
memcpy(&input_buf[b * T * in_ch + t * in_ch],
&context_switch[b * T * ctx_ch + t * ctx_ch],
ctx_ch * sizeof(float));
fprintf(stderr, "[DiT] Cover: switched to non-cover context at step %d/%d\n",
step, num_steps);
}
// Set timestep (changes each step)
if (t_t) {
ggml_backend_tensor_set(t_t, &t_curr, 0, sizeof(float));
+5
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@@ -36,6 +36,7 @@ void request_init(AceRequest * r) {
r->inference_steps = 8;
r->guidance_scale = 1.0f;
r->shift = 3.0f;
r->audio_cover_strength = 0.5f;
}
// JSON string escape / unescape
@@ -250,6 +251,7 @@ bool request_parse_from_str(AceRequest * r, std::string json) {
else if (k == "inference_steps") r->inference_steps = atoi(v.c_str());
else if (k == "guidance_scale") r->guidance_scale = (float)atof(v.c_str());
else if (k == "shift") r->shift = (float)atof(v.c_str());
else if (k == "audio_cover_strength") r->audio_cover_strength = (float)atof(v.c_str());
else if (k == "codes_temperature") r->codes_temperature = (float)atof(v.c_str());
else if (k == "codes_top_p") r->codes_top_p = (float)atof(v.c_str());
@@ -297,6 +299,7 @@ bool request_write(const AceRequest * r, const char * path) {
fprintf(f, " \"inference_steps\": %d,\n", r->inference_steps);
fprintf(f, " \"guidance_scale\": %.1f,\n", r->guidance_scale);
fprintf(f, " \"shift\": %.1f,\n", r->shift);
fprintf(f, " \"audio_cover_strength\": %.2f,\n", r->audio_cover_strength);
// audio_codes last (no trailing comma)
fprintf(f, " \"audio_codes\": \"%s\"\n", json_escape(r->audio_codes).c_str());
fprintf(f, "}\n");
@@ -321,6 +324,8 @@ void request_dump(const AceRequest * r, FILE * f) {
r->lm_temperature, r->lm_top_p, r->lm_top_k);
fprintf(f, " dit: steps=%d guidance=%.1f shift=%.1f\n",
r->inference_steps, r->guidance_scale, r->shift);
if (r->audio_cover_strength != 0.5f)
fprintf(f, " cover: strength=%.2f\n", r->audio_cover_strength);
fprintf(f, " audio_codes: %s\n",
r->audio_codes.empty() ? "(none)" : "(present)");
}
+3
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@@ -47,6 +47,9 @@ struct AceRequest {
int inference_steps; // 8
float guidance_scale; // 7.0
float shift; // 1.0
// cover mode (active when --src-audio is provided on CLI)
float audio_cover_strength; // 0.5 (0-1, fraction of DiT steps using source context)
};
// Initialize all fields to defaults (matches Python GenerationParams defaults)
+392
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@@ -0,0 +1,392 @@
// vae-enc.h: AutoencoderOobleck encoder (audio -> latent) via ggml
//
// Mirror of vae.h decoder. Reuses VAEResUnit, load helpers, graph ops.
// Architecture: conv1(2->128,k=7) -> 5x(3xresunit+snake+strided_conv) -> snake+conv2(2048->128,k=3)
// Output 128ch = mean[64] + scale[64]. Deterministic encode returns mean.
// Downsample: 2x4x4x6x10 = 1920x (matches decoder upsample)
#pragma once
#include "vae.h"
// Encoder block: 3xResUnit(in_ch) -> snake(in_ch) -> strided Conv1d(in_ch -> out_ch)
// Decoder block is the mirror: snake(in_ch) -> ConvT(in_ch -> out_ch) -> 3xResUnit(out_ch)
struct VAEEncBlock {
VAEResUnit ru[3];
struct ggml_tensor * sa, * sb; // snake [1, in_ch]
struct ggml_tensor * dw, * db; // strided conv [K, in_ch, out_ch], bias [out_ch]
int in_ch, out_ch, stride, kernel, padding;
};
struct VAEEncoder {
struct ggml_tensor * c1w, * c1b; // conv1 [7, 2, 128], bias [128]
VAEEncBlock blk[5];
struct ggml_tensor * sa, * sb; // final snake [1, 2048]
struct ggml_tensor * c2w, * c2b; // conv2 [3, 2048, 128], bias [128]
ggml_backend_t backend;
ggml_backend_t cpu_backend;
ggml_backend_sched_t sched;
ggml_backend_buffer_t buf;
struct ggml_context * weight_ctx;
// graph cache (rebuilt when T_audio changes)
struct ggml_context * graph_ctx;
uint8_t * graph_buf;
struct ggml_cgraph * graph;
struct ggml_tensor * graph_input; // [T_audio, 2]
struct ggml_tensor * graph_output; // [T_latent, 128]
int graph_T; // cached T_audio (0 = no cache)
std::vector<float> scratch_in; // transposed input [2 * T_audio]
};
// Load encoder weights from the same VAE GGUF (encoder.* tensors)
static void vae_enc_load(VAEEncoder * m, const char * path) {
GGUFModel gf = {};
if (!gf_load(&gf, path)) {
fprintf(stderr, "[VAE-Enc] FATAL: cannot load %s\n", path);
exit(1);
}
// Encoder channel layout (mirror of decoder, bottom-up):
// conv1: 2 -> 128
// block: [128->128, 128->256, 256->512, 512->1024, 1024->2048]
// conv2: 2048 -> 128 (split: mean[64] + scale[64])
// ResUnits run at in_ch (before downsample), unlike decoder (at out_ch, after upsample).
static const int in_ch[] = {128, 128, 256, 512, 1024};
static const int out_ch[] = {128, 256, 512, 1024, 2048};
static const int strides[] = { 2, 4, 4, 6, 10};
static const int dilations[] = {1, 3, 9};
// Phase 1: create weight tensors
size_t ctx_size = ggml_tensor_overhead() * 256;
struct ggml_init_params p = { ctx_size, NULL, true };
m->weight_ctx = ggml_init(p);
struct ggml_context * ctx = m->weight_ctx;
m->c1w = ggml_new_tensor_3d(ctx, GGML_TYPE_F16, 7, 2, 128);
m->c1b = ggml_new_tensor_1d(ctx, GGML_TYPE_F32, 128);
for (int i = 0; i < 5; i++) {
VAEEncBlock & b = m->blk[i];
b.in_ch = in_ch[i];
b.out_ch = out_ch[i];
b.stride = strides[i];
b.kernel = strides[i] * 2;
b.padding = (strides[i] + 1) / 2; // ceil(stride / 2)
int C = in_ch[i]; // res_units + snake at in_ch
// 3 res units at in_ch
for (int r = 0; r < 3; r++) {
VAEResUnit & ru = b.ru[r];
ru.dilation = dilations[r];
ru.s1a = ggml_new_tensor_2d(ctx, GGML_TYPE_F32, 1, C);
ru.s1b = ggml_new_tensor_2d(ctx, GGML_TYPE_F32, 1, C);
ru.c1w = ggml_new_tensor_3d(ctx, GGML_TYPE_F16, 7, C, C);
ru.c1b = ggml_new_tensor_1d(ctx, GGML_TYPE_F32, C);
ru.s2a = ggml_new_tensor_2d(ctx, GGML_TYPE_F32, 1, C);
ru.s2b = ggml_new_tensor_2d(ctx, GGML_TYPE_F32, 1, C);
ru.c2w = ggml_new_tensor_3d(ctx, GGML_TYPE_F16, 1, C, C);
ru.c2b = ggml_new_tensor_1d(ctx, GGML_TYPE_F32, C);
}
// snake at in_ch (before downsample conv)
b.sa = ggml_new_tensor_2d(ctx, GGML_TYPE_F32, 1, C);
b.sb = ggml_new_tensor_2d(ctx, GGML_TYPE_F32, 1, C);
// strided conv1d: [K, in_ch, out_ch]
b.dw = ggml_new_tensor_3d(ctx, GGML_TYPE_F16, b.kernel, in_ch[i], out_ch[i]);
b.db = ggml_new_tensor_1d(ctx, GGML_TYPE_F32, out_ch[i]);
}
m->sa = ggml_new_tensor_2d(ctx, GGML_TYPE_F32, 1, 2048);
m->sb = ggml_new_tensor_2d(ctx, GGML_TYPE_F32, 1, 2048);
m->c2w = ggml_new_tensor_3d(ctx, GGML_TYPE_F16, 3, 2048, 128);
m->c2b = ggml_new_tensor_1d(ctx, GGML_TYPE_F32, 128);
// Phase 2: allocate backend buffer
BackendPair bp = backend_init("VAE-Enc");
m->backend = bp.backend;
m->cpu_backend = bp.cpu_backend;
m->sched = backend_sched_new(bp, 8192);
m->buf = ggml_backend_alloc_ctx_tensors(ctx, m->backend);
if (!m->buf) {
fprintf(stderr, "[VAE-Enc] FATAL: failed to allocate weight buffer\n");
exit(1);
}
fprintf(stderr, "[VAE-Enc] Backend: %s, Weight buffer: %.1f MB\n",
ggml_backend_name(m->backend),
(float)ggml_backend_buffer_get_size(m->buf) / (1024 * 1024));
// Phase 3: load and fuse weights
vae_fuse_wn(m->c1w, gf, "encoder.conv1");
vae_load_bias(m->c1b, gf, "encoder.conv1.bias");
for (int i = 0; i < 5; i++) {
VAEEncBlock & b = m->blk[i];
std::string blk_pfx = "encoder.block." + std::to_string(i);
// res_units first (same load pattern as decoder)
for (int r = 0; r < 3; r++) {
VAEResUnit & ru = b.ru[r];
std::string rp = blk_pfx + ".res_unit" + std::to_string(r + 1);
vae_load_snake(ru.s1a, gf, rp + ".snake1.alpha");
vae_load_snake_inv(ru.s1b, gf, rp + ".snake1.beta");
vae_fuse_wn(ru.c1w, gf, rp + ".conv1");
vae_load_bias(ru.c1b, gf, rp + ".conv1.bias");
vae_load_snake(ru.s2a, gf, rp + ".snake2.alpha");
vae_load_snake_inv(ru.s2b, gf, rp + ".snake2.beta");
vae_fuse_wn(ru.c2w, gf, rp + ".conv2");
vae_load_bias(ru.c2b, gf, rp + ".conv2.bias");
}
// snake + strided downsample conv (regular conv1d, NOT transposed)
vae_load_snake(b.sa, gf, blk_pfx + ".snake1.alpha");
vae_load_snake_inv(b.sb, gf, blk_pfx + ".snake1.beta");
vae_fuse_wn(b.dw, gf, blk_pfx + ".conv1");
vae_load_bias(b.db, gf, blk_pfx + ".conv1.bias");
}
vae_load_snake(m->sa, gf, "encoder.snake1.alpha");
vae_load_snake_inv(m->sb, gf, "encoder.snake1.beta");
vae_fuse_wn(m->c2w, gf, "encoder.conv2");
vae_load_bias(m->c2b, gf, "encoder.conv2.bias");
fprintf(stderr, "[VAE-Enc] Loaded: 5 blocks, downsample=1920x, F32 activations\n");
gf_close(&gf);
}
// Build encoder graph: audio [T_audio, 2] -> [T_latent, 128]
static struct ggml_tensor * vae_enc_build_graph(
struct ggml_context * ctx,
VAEEncoder * m,
struct ggml_tensor * audio) { // [T, 2]
// conv1: [T, 2] -> [T, 128]
struct ggml_tensor * x = vae_conv1d(ctx, m->c1w, m->c1b, audio, 1, 3, 1);
// 5 encoder blocks: resunits(in_ch) -> snake(in_ch) -> strided conv(in_ch -> out_ch)
for (int i = 0; i < 5; i++) {
VAEEncBlock & b = m->blk[i];
for (int r = 0; r < 3; r++)
x = vae_res_unit(ctx, &b.ru[r], x);
x = vae_snake(ctx, x, b.sa, b.sb);
x = vae_conv1d(ctx, b.dw, b.db, x, b.stride, b.padding, 1);
}
// Final: snake(2048) -> conv2(2048 -> 128, k=3, pad=1)
x = vae_snake(ctx, x, m->sa, m->sb);
x = vae_conv1d(ctx, m->c2w, m->c2b, x, 1, 1, 1);
return x; // [T_latent, 128]
}
// Core compute: build/cache graph, set input, run. Returns T_latent or -1.
// Output stays in m->graph_output for caller to read.
static int vae_enc_compute(
VAEEncoder * m,
const float * audio, // [T_audio, 2] time-major interleaved stereo
int T_audio) {
// Rebuild graph when T_audio changes
if (m->graph_T != T_audio) {
if (m->graph_ctx) {
ggml_backend_sched_reset(m->sched);
ggml_free(m->graph_ctx);
free(m->graph_buf);
}
size_t ctx_size = ggml_tensor_overhead() * 1024 + ggml_graph_overhead_custom(8192, false);
m->graph_buf = (uint8_t *)malloc(ctx_size);
struct ggml_init_params p = { ctx_size, m->graph_buf, true };
struct ggml_context * ctx = ggml_init(p);
m->graph_input = ggml_new_tensor_2d(ctx, GGML_TYPE_F32, T_audio, 2);
ggml_set_name(m->graph_input, "enc_input");
ggml_set_input(m->graph_input);
m->graph_output = vae_enc_build_graph(ctx, m, m->graph_input);
ggml_set_name(m->graph_output, "enc_output");
ggml_set_output(m->graph_output);
m->graph = ggml_new_graph_custom(ctx, 8192, false);
ggml_build_forward_expand(m->graph, m->graph_output);
if (!ggml_backend_sched_alloc_graph(m->sched, m->graph)) {
fprintf(stderr, "[VAE-Enc] FATAL: graph alloc failed for T=%d\n", T_audio);
ggml_free(ctx);
free(m->graph_buf);
m->graph_ctx = NULL;
m->graph_buf = NULL;
m->graph_T = 0;
return -1;
}
m->graph_ctx = ctx;
m->graph_T = T_audio;
fprintf(stderr, "[VAE-Enc] Graph: %d nodes, T_audio=%d\n",
ggml_graph_n_nodes(m->graph), T_audio);
}
// Transpose: [T, 2] time-major -> ggml [T, 2] channel-contiguous
// ggml ne[0]=T is the contiguous dim, so we write all T samples per channel
size_t in_size = (size_t)2 * T_audio;
if (m->scratch_in.size() < in_size)
m->scratch_in.resize(in_size);
for (int c = 0; c < 2; c++)
for (int t = 0; t < T_audio; t++)
m->scratch_in[c * T_audio + t] = audio[t * 2 + c];
ggml_backend_tensor_set(m->graph_input,
m->scratch_in.data(), 0, in_size * sizeof(float));
ggml_backend_sched_graph_compute(m->sched, m->graph);
return (int)m->graph_output->ne[0]; // T_latent
}
// Encode API: audio [T_audio, 2] -> latent_out [T_latent, 64] (mean only, deterministic)
// Returns T_latent (or -1 on error).
// latent_out must hold at least (T_audio / 1920) * 64 floats.
static int vae_enc_encode(
VAEEncoder * m,
const float * audio, // [T_audio, 2] interleaved stereo
int T_audio,
float * latent_out, // [T_latent, 64] output, time-major
int max_T_latent) {
int T_latent = vae_enc_compute(m, audio, T_audio);
if (T_latent < 0) return -1;
if (T_latent > max_T_latent) {
fprintf(stderr, "[VAE-Enc] T_latent %d exceeds max %d\n", T_latent, max_T_latent);
return -1;
}
// Graph output is [ne0=T_latent, ne1=128] in ggml, channel-contiguous.
// Channels 0..63 = mean, 64..127 = scale. We only read mean.
// ggml layout: data[c * T_latent + t] for channel c, time t.
// We write time-major: latent_out[t * 64 + c] = data[c * T_latent + t]
//
// Read the full 128ch output once, extract mean channels 0..63
size_t out_bytes = (size_t)128 * T_latent * sizeof(float);
std::vector<float> raw(128 * T_latent);
ggml_backend_tensor_get(m->graph_output, raw.data(), 0, out_bytes);
for (int t = 0; t < T_latent; t++)
for (int c = 0; c < 64; c++)
latent_out[t * 64 + c] = raw[c * T_latent + t];
fprintf(stderr, "[VAE-Enc] Encode: T_audio=%d -> T_latent=%d (%.2fs @ 48kHz)\n",
T_audio, T_latent, (float)T_audio / 48000.0f);
return T_latent;
}
// Tiled encode for long audio (same chunking strategy as decoder)
// chunk_size: latent frames per tile, overlap: context frames on each side
static int vae_enc_encode_tiled(
VAEEncoder * m,
const float * audio, // [T_audio, 2] interleaved stereo
int T_audio,
float * latent_out, // [T_latent, 64] output, time-major
int max_T_latent,
int chunk_size = 256,
int overlap = 64) {
// Work in audio-sample space. Each latent frame = 1920 audio samples.
int audio_chunk = chunk_size * 1920;
int audio_overlap = overlap * 1920;
// Shrink overlap until stride is positive
while (audio_chunk - 2 * audio_overlap <= 0 && audio_overlap > 0)
audio_overlap /= 2;
// Short audio: encode directly
if (T_audio <= audio_chunk)
return vae_enc_encode(m, audio, T_audio, latent_out, max_T_latent);
int audio_stride = audio_chunk - 2 * audio_overlap;
int num_steps = (T_audio + audio_stride - 1) / audio_stride;
fprintf(stderr, "[VAE-Enc] Tiled encode: %d tiles (chunk=%d, overlap=%d, stride=%d audio samples)\n",
num_steps, audio_chunk, audio_overlap, audio_stride);
float downsample_factor = 0.0f;
int latent_write_pos = 0;
for (int i = 0; i < num_steps; i++) {
// Core range in audio samples (the part we keep)
int core_start = i * audio_stride;
int core_end = core_start + audio_stride;
if (core_end > T_audio) core_end = T_audio;
// Window with overlap context
int win_start = core_start - audio_overlap;
if (win_start < 0) win_start = 0;
int win_end = core_end + audio_overlap;
if (win_end > T_audio) win_end = T_audio;
int win_len = win_end - win_start;
// Encode this window
int tile_T = vae_enc_compute(m, audio + win_start * 2, win_len);
if (tile_T < 0) {
fprintf(stderr, "[VAE-Enc] FATAL: tile %d encode failed\n", i);
return -1;
}
// Determine downsample factor from first tile
if (i == 0) {
downsample_factor = (float)tile_T / (float)win_len;
fprintf(stderr, "[VAE-Enc] Downsample factor: %.6f (expected ~1/1920)\n",
downsample_factor);
}
// Trim in latent frames (mirror of decoder trim logic)
int added_start = core_start - win_start;
int trim_start = (int)roundf((float)added_start * downsample_factor);
int added_end = win_end - core_end;
int trim_end = (int)roundf((float)added_end * downsample_factor);
int end_idx = (trim_end > 0) ? (tile_T - trim_end) : tile_T;
int core_len = end_idx - trim_start;
if (core_len <= 0) continue;
if (latent_write_pos + core_len > max_T_latent) {
fprintf(stderr, "[VAE-Enc] FATAL: tiled output exceeds max_T_latent\n");
return -1;
}
// Read tile output [ne0=tile_T, ne1=128], extract mean (ch 0..63), transpose
// Only read the first 64 channels (mean), skip scale channels 64..127
size_t out_bytes = (size_t)128 * tile_T * sizeof(float);
std::vector<float> raw(128 * tile_T);
ggml_backend_tensor_get(m->graph_output, raw.data(), 0, out_bytes);
for (int t = 0; t < core_len; t++)
for (int c = 0; c < 64; c++)
latent_out[(latent_write_pos + t) * 64 + c] =
raw[c * tile_T + (trim_start + t)];
latent_write_pos += core_len;
}
fprintf(stderr, "[VAE-Enc] Tiled encode done: %d tiles -> T_latent=%d (%.2fs @ 48kHz)\n",
num_steps, latent_write_pos, (float)T_audio / 48000.0f);
return latent_write_pos;
}
// Free all resources
static void vae_enc_free(VAEEncoder * m) {
if (m->graph_ctx) {
ggml_backend_sched_reset(m->sched);
ggml_free(m->graph_ctx);
free(m->graph_buf);
}
if (m->sched) ggml_backend_sched_free(m->sched);
if (m->buf) ggml_backend_buffer_free(m->buf);
if (m->weight_ctx) ggml_free(m->weight_ctx);
if (m->backend && m->backend != m->cpu_backend) ggml_backend_free(m->backend);
if (m->cpu_backend) ggml_backend_free(m->cpu_backend);
*m = {};
}